diff --git a/media/libaaudio/src/Android.mk b/media/libaaudio/src/Android.mk
index 7131c6c..cfcf27a 100644
--- a/media/libaaudio/src/Android.mk
+++ b/media/libaaudio/src/Android.mk
@@ -32,6 +32,7 @@
     core/AudioStream.cpp \
     core/AudioStreamBuilder.cpp \
     core/AAudioAudio.cpp \
+    core/AAudioStreamParameters.cpp \
     legacy/AudioStreamLegacy.cpp \
     legacy/AudioStreamRecord.cpp \
     legacy/AudioStreamTrack.cpp \
@@ -90,6 +91,7 @@
 LOCAL_SRC_FILES = core/AudioStream.cpp \
     core/AudioStreamBuilder.cpp \
     core/AAudioAudio.cpp \
+    core/AAudioStreamParameters.cpp \
     legacy/AudioStreamLegacy.cpp \
     legacy/AudioStreamRecord.cpp \
     legacy/AudioStreamTrack.cpp \
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
index 44edb1d..e763934 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
@@ -36,17 +36,17 @@
 
 status_t AAudioStreamConfiguration::writeToParcel(Parcel* parcel) const {
     status_t status;
-    status = parcel->writeInt32(mDeviceId);
+    status = parcel->writeInt32(getDeviceId());
     if (status != NO_ERROR) goto error;
-    status = parcel->writeInt32(mSampleRate);
+    status = parcel->writeInt32(getSampleRate());
     if (status != NO_ERROR) goto error;
-    status = parcel->writeInt32(mSamplesPerFrame);
+    status = parcel->writeInt32(getSamplesPerFrame());
     if (status != NO_ERROR) goto error;
-    status = parcel->writeInt32((int32_t) mSharingMode);
+    status = parcel->writeInt32((int32_t) getSharingMode());
     if (status != NO_ERROR) goto error;
-    status = parcel->writeInt32((int32_t) mAudioFormat);
+    status = parcel->writeInt32((int32_t) getFormat());
     if (status != NO_ERROR) goto error;
-    status = parcel->writeInt32(mBufferCapacity);
+    status = parcel->writeInt32(getBufferCapacity());
     if (status != NO_ERROR) goto error;
     return NO_ERROR;
 error:
@@ -55,57 +55,27 @@
 }
 
 status_t AAudioStreamConfiguration::readFromParcel(const Parcel* parcel) {
-    status_t status = parcel->readInt32(&mDeviceId);
+    int32_t value;
+    status_t status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
-    status = parcel->readInt32(&mSampleRate);
+    setDeviceId(value);
+    status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
-    status = parcel->readInt32(&mSamplesPerFrame);
+    setSampleRate(value);
+    status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
-    status = parcel->readInt32(&mSharingMode);
+    setSamplesPerFrame(value);
+    status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
-    status = parcel->readInt32(&mAudioFormat);
+    setSharingMode(value);
+    status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
-    status = parcel->readInt32(&mBufferCapacity);
+    setFormat(value);
+    status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
+    setBufferCapacity(value);
     return NO_ERROR;
 error:
     ALOGE("AAudioStreamConfiguration.readFromParcel(): read failed = %d", status);
     return status;
-}
-
-aaudio_result_t AAudioStreamConfiguration::validate() const {
-    // Validate results of the open.
-    if (mSampleRate < 0 || mSampleRate >= 8 * 48000) { // TODO review limits
-        ALOGE("AAudioStreamConfiguration.validate(): invalid sampleRate = %d", mSampleRate);
-        return AAUDIO_ERROR_INTERNAL;
-    }
-
-    if (mSamplesPerFrame < 1 || mSamplesPerFrame >= 32) { // TODO review limits
-        ALOGE("AAudioStreamConfiguration.validate() invalid samplesPerFrame = %d", mSamplesPerFrame);
-        return AAUDIO_ERROR_INTERNAL;
-    }
-
-    switch (mAudioFormat) {
-    case AAUDIO_FORMAT_PCM_I16:
-    case AAUDIO_FORMAT_PCM_FLOAT:
-        break;
-    default:
-        ALOGE("AAudioStreamConfiguration.validate() invalid audioFormat = %d", mAudioFormat);
-        return AAUDIO_ERROR_INTERNAL;
-    }
-
-    if (mBufferCapacity < 0) {
-        ALOGE("AAudioStreamConfiguration.validate() invalid mBufferCapacity = %d", mBufferCapacity);
-        return AAUDIO_ERROR_INTERNAL;
-    }
-    return AAUDIO_OK;
-}
-
-void AAudioStreamConfiguration::dump() const {
-    ALOGD("AAudioStreamConfiguration mDeviceId        = %d", mDeviceId);
-    ALOGD("AAudioStreamConfiguration mSampleRate      = %d", mSampleRate);
-    ALOGD("AAudioStreamConfiguration mSamplesPerFrame = %d", mSamplesPerFrame);
-    ALOGD("AAudioStreamConfiguration mSharingMode     = %d", (int)mSharingMode);
-    ALOGD("AAudioStreamConfiguration mAudioFormat     = %d", (int)mAudioFormat);
-    ALOGD("AAudioStreamConfiguration mBufferCapacity  = %d", mBufferCapacity);
-}
+}
\ No newline at end of file
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.h b/media/libaaudio/src/binding/AAudioStreamConfiguration.h
index 144595a..b324896 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.h
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.h
@@ -22,6 +22,7 @@
 #include <aaudio/AAudio.h>
 #include <binder/Parcel.h>
 #include <binder/Parcelable.h>
+#include "core/AAudioStreamParameters.h"
 
 using android::status_t;
 using android::Parcel;
@@ -29,74 +30,14 @@
 
 namespace aaudio {
 
-class AAudioStreamConfiguration : public Parcelable {
+class AAudioStreamConfiguration : public AAudioStreamParameters, public Parcelable {
 public:
     AAudioStreamConfiguration();
     virtual ~AAudioStreamConfiguration();
 
-    int32_t getDeviceId() const {
-        return mDeviceId;
-    }
-
-    void setDeviceId(int32_t deviceId) {
-        mDeviceId = deviceId;
-    }
-
-    int32_t getSampleRate() const {
-        return mSampleRate;
-    }
-
-    void setSampleRate(int32_t sampleRate) {
-        mSampleRate = sampleRate;
-    }
-
-    int32_t getSamplesPerFrame() const {
-        return mSamplesPerFrame;
-    }
-
-    void setSamplesPerFrame(int32_t samplesPerFrame) {
-        mSamplesPerFrame = samplesPerFrame;
-    }
-
-    aaudio_format_t getAudioFormat() const {
-        return mAudioFormat;
-    }
-
-    void setAudioFormat(aaudio_format_t audioFormat) {
-        mAudioFormat = audioFormat;
-    }
-
-    aaudio_sharing_mode_t getSharingMode() const {
-        return mSharingMode;
-    }
-
-    void setSharingMode(aaudio_sharing_mode_t sharingMode) {
-        mSharingMode = sharingMode;
-    }
-
-    int32_t getBufferCapacity() const {
-        return mBufferCapacity;
-    }
-
-    void setBufferCapacity(int32_t frames) {
-        mBufferCapacity = frames;
-    }
-
     virtual status_t writeToParcel(Parcel* parcel) const override;
 
     virtual status_t readFromParcel(const Parcel* parcel) override;
-
-    aaudio_result_t validate() const;
-
-    void dump() const;
-
-private:
-    int32_t               mDeviceId        = AAUDIO_UNSPECIFIED;
-    int32_t               mSampleRate      = AAUDIO_UNSPECIFIED;
-    int32_t               mSamplesPerFrame = AAUDIO_UNSPECIFIED;
-    aaudio_sharing_mode_t mSharingMode     = AAUDIO_SHARING_MODE_SHARED;
-    aaudio_format_t       mAudioFormat     = AAUDIO_FORMAT_UNSPECIFIED;
-    int32_t               mBufferCapacity  = AAUDIO_UNSPECIFIED;
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/binding/IAAudioService.cpp b/media/libaaudio/src/binding/IAAudioService.cpp
index 97fbaaa..b3c4934 100644
--- a/media/libaaudio/src/binding/IAAudioService.cpp
+++ b/media/libaaudio/src/binding/IAAudioService.cpp
@@ -264,13 +264,19 @@
         case OPEN_STREAM: {
             CHECK_INTERFACE(IAAudioService, data, reply);
             request.readFromParcel(&data);
-            //ALOGD("BnAAudioService::client openStream request dump --------------------");
-            //request.dump();
-            // Override the uid and pid from the client in case they are incorrect.
-            request.setUserId(IPCThreadState::self()->getCallingUid());
-            request.setProcessId(IPCThreadState::self()->getCallingPid());
-            streamHandle = openStream(request, configuration);
-            //ALOGD("BnAAudioService::onTransact OPEN_STREAM server handle = 0x%08X", streamHandle);
+            result = request.validate();
+            if (result != AAUDIO_OK) {
+                streamHandle = result;
+            } else {
+                //ALOGD("BnAAudioService::client openStream request dump --------------------");
+                //request.dump();
+                // Override the uid and pid from the client in case they are incorrect.
+                request.setUserId(IPCThreadState::self()->getCallingUid());
+                request.setProcessId(IPCThreadState::self()->getCallingPid());
+                streamHandle = openStream(request, configuration);
+                //ALOGD("BnAAudioService::onTransact OPEN_STREAM server handle = 0x%08X",
+                //        streamHandle);
+            }
             reply->writeInt32(streamHandle);
             configuration.writeToParcel(reply);
             return NO_ERROR;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 7b01e44..8b14922 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -86,7 +86,7 @@
         setFormat(AAUDIO_FORMAT_PCM_FLOAT);
     }
     // Request FLOAT for the shared mixer.
-    request.getConfiguration().setAudioFormat(AAUDIO_FORMAT_PCM_FLOAT);
+    request.getConfiguration().setFormat(AAUDIO_FORMAT_PCM_FLOAT);
 
     // Build the request to send to the server.
     request.setUserId(getuid());
@@ -119,7 +119,7 @@
         setSharingMode(configuration.getSharingMode());
 
         // Save device format so we can do format conversion and volume scaling together.
-        mDeviceFormat = configuration.getAudioFormat();
+        mDeviceFormat = configuration.getFormat();
 
         result = mServiceInterface.getStreamDescription(mServiceStreamHandle, mEndPointParcelable);
         if (result != AAUDIO_OK) {
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
new file mode 100644
index 0000000..65c2b46
--- /dev/null
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -0,0 +1,91 @@
+/*
+ * Copyright 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#define LOG_TAG "AAudio"
+#include <utils/Log.h>
+#include <hardware/audio.h>
+
+#include "AAudioStreamParameters.h"
+
+using namespace aaudio;
+
+// TODO These defines should be moved to a central place in audio.
+#define SAMPLES_PER_FRAME_MIN        1
+// TODO Remove 8 channel limitation.
+#define SAMPLES_PER_FRAME_MAX        FCC_8
+#define SAMPLE_RATE_HZ_MIN           8000
+// HDMI supports up to 32 channels at 1536000 Hz.
+#define SAMPLE_RATE_HZ_MAX           1600000
+
+AAudioStreamParameters::AAudioStreamParameters() {}
+AAudioStreamParameters::~AAudioStreamParameters() {}
+
+aaudio_result_t AAudioStreamParameters::validate() const {
+    if (mSamplesPerFrame != AAUDIO_UNSPECIFIED
+        && (mSamplesPerFrame < SAMPLES_PER_FRAME_MIN || mSamplesPerFrame > SAMPLES_PER_FRAME_MAX)) {
+        ALOGE("AAudioStreamParameters: channelCount out of range = %d", mSamplesPerFrame);
+        return AAUDIO_ERROR_OUT_OF_RANGE;
+    }
+
+    if (mDeviceId < 0) {
+        ALOGE("AAudioStreamParameters: deviceId out of range = %d", mDeviceId);
+        return AAUDIO_ERROR_OUT_OF_RANGE;
+    }
+
+    switch (mSharingMode) {
+        case AAUDIO_SHARING_MODE_EXCLUSIVE:
+        case AAUDIO_SHARING_MODE_SHARED:
+            break;
+        default:
+            ALOGE("AAudioStreamParameters: illegal sharingMode = %d", mSharingMode);
+            return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+            // break;
+    }
+
+    switch (mAudioFormat) {
+        case AAUDIO_FORMAT_UNSPECIFIED:
+        case AAUDIO_FORMAT_PCM_I16:
+        case AAUDIO_FORMAT_PCM_FLOAT:
+            break; // valid
+        default:
+            ALOGE("AAudioStreamParameters: audioFormat not valid = %d", mAudioFormat);
+            return AAUDIO_ERROR_INVALID_FORMAT;
+            // break;
+    }
+
+    if (mSampleRate != AAUDIO_UNSPECIFIED
+        && (mSampleRate < SAMPLE_RATE_HZ_MIN || mSampleRate > SAMPLE_RATE_HZ_MAX)) {
+        ALOGE("AAudioStreamParameters: sampleRate out of range = %d", mSampleRate);
+        return AAUDIO_ERROR_INVALID_RATE;
+    }
+
+    if (mBufferCapacity < 0) {
+        ALOGE("AAudioStreamParameters: bufferCapacity out of range = %d", mBufferCapacity);
+        return AAUDIO_ERROR_OUT_OF_RANGE;
+    }
+
+    return AAUDIO_OK;
+}
+
+void AAudioStreamParameters::dump() const {
+    ALOGD("AAudioStreamParameters mDeviceId        = %d", mDeviceId);
+    ALOGD("AAudioStreamParameters mSampleRate      = %d", mSampleRate);
+    ALOGD("AAudioStreamParameters mSamplesPerFrame = %d", mSamplesPerFrame);
+    ALOGD("AAudioStreamParameters mSharingMode     = %d", (int)mSharingMode);
+    ALOGD("AAudioStreamParameters mAudioFormat     = %d", (int)mAudioFormat);
+    ALOGD("AAudioStreamParameters mBufferCapacity  = %d", mBufferCapacity);
+}
\ No newline at end of file
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.h b/media/libaaudio/src/core/AAudioStreamParameters.h
new file mode 100644
index 0000000..97379cc
--- /dev/null
+++ b/media/libaaudio/src/core/AAudioStreamParameters.h
@@ -0,0 +1,97 @@
+/*
+ * Copyright 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAUDIO_STREAM_PARAMETERS_H
+#define AAUDIO_STREAM_PARAMETERS_H
+
+#include <stdint.h>
+
+#include <aaudio/AAudio.h>
+
+namespace aaudio {
+
+class AAudioStreamParameters {
+public:
+    AAudioStreamParameters();
+    virtual ~AAudioStreamParameters();
+
+    int32_t getDeviceId() const {
+        return mDeviceId;
+    }
+
+    void setDeviceId(int32_t deviceId) {
+        mDeviceId = deviceId;
+    }
+
+    int32_t getSampleRate() const {
+        return mSampleRate;
+    }
+
+    void setSampleRate(int32_t sampleRate) {
+        mSampleRate = sampleRate;
+    }
+
+    int32_t getSamplesPerFrame() const {
+        return mSamplesPerFrame;
+    }
+
+    /**
+     * This is also known as channelCount.
+     */
+    void setSamplesPerFrame(int32_t samplesPerFrame) {
+        mSamplesPerFrame = samplesPerFrame;
+    }
+
+    aaudio_format_t getFormat() const {
+        return mAudioFormat;
+    }
+
+    void setFormat(aaudio_format_t audioFormat) {
+        mAudioFormat = audioFormat;
+    }
+
+    aaudio_sharing_mode_t getSharingMode() const {
+        return mSharingMode;
+    }
+
+    void setSharingMode(aaudio_sharing_mode_t sharingMode) {
+        mSharingMode = sharingMode;
+    }
+
+    int32_t getBufferCapacity() const {
+        return mBufferCapacity;
+    }
+
+    void setBufferCapacity(int32_t frames) {
+        mBufferCapacity = frames;
+    }
+
+    virtual aaudio_result_t validate() const;
+
+    void dump() const;
+
+private:
+    int32_t                    mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+    int32_t                    mSampleRate      = AAUDIO_UNSPECIFIED;
+    int32_t                    mDeviceId        = AAUDIO_UNSPECIFIED;
+    aaudio_sharing_mode_t      mSharingMode     = AAUDIO_SHARING_MODE_SHARED;
+    aaudio_format_t            mAudioFormat     = AAUDIO_FORMAT_UNSPECIFIED;
+    int32_t                    mBufferCapacity  = AAUDIO_UNSPECIFIED;
+};
+
+} /* namespace aaudio */
+
+#endif //AAUDIO_STREAM_PARAMETERS_H
\ No newline at end of file
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 6c4aa59..43a1ef1 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -179,36 +179,9 @@
 
     // Check for values that are ridiculously out of range to prevent math overflow exploits.
     // The service will do a better check.
-    if (mSamplesPerFrame != AAUDIO_UNSPECIFIED
-        && (mSamplesPerFrame < SAMPLES_PER_FRAME_MIN || mSamplesPerFrame > SAMPLES_PER_FRAME_MAX)) {
-        ALOGE("AudioStreamBuilder: channelCount out of range = %d", mSamplesPerFrame);
-        return AAUDIO_ERROR_OUT_OF_RANGE;
-    }
-
-    if (mDeviceId < 0) {
-        ALOGE("AudioStreamBuilder: deviceId out of range = %d", mDeviceId);
-        return AAUDIO_ERROR_OUT_OF_RANGE;
-    }
-
-    switch (mSharingMode) {
-        case AAUDIO_SHARING_MODE_EXCLUSIVE:
-        case AAUDIO_SHARING_MODE_SHARED:
-            break;
-        default:
-            ALOGE("AudioStreamBuilder: illegal sharingMode = %d", mSharingMode);
-            return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
-            // break;
-    }
-
-    switch (mFormat) {
-        case AAUDIO_FORMAT_UNSPECIFIED:
-        case AAUDIO_FORMAT_PCM_I16:
-        case AAUDIO_FORMAT_PCM_FLOAT:
-            break; // valid
-        default:
-            ALOGE("AudioStreamBuilder: audioFormat not valid = %d", mFormat);
-            return AAUDIO_ERROR_INVALID_FORMAT;
-            // break;
+    aaudio_result_t result = AAudioStreamParameters::validate();
+    if (result != AAUDIO_OK) {
+        return result;
     }
 
     switch (mDirection) {
@@ -221,17 +194,6 @@
             // break;
     }
 
-    if (mSampleRate != AAUDIO_UNSPECIFIED
-        && (mSampleRate < SAMPLE_RATE_HZ_MIN || mSampleRate > SAMPLE_RATE_HZ_MAX)) {
-        ALOGE("AudioStreamBuilder: sampleRate out of range = %d", mSampleRate);
-        return AAUDIO_ERROR_INVALID_RATE;
-    }
-
-    if (mBufferCapacity < 0) {
-        ALOGE("AudioStreamBuilder: bufferCapacity out of range = %d", mBufferCapacity);
-        return AAUDIO_ERROR_OUT_OF_RANGE;
-    }
-
     switch (mPerformanceMode) {
         case AAUDIO_PERFORMANCE_MODE_NONE:
         case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.h b/media/libaaudio/src/core/AudioStreamBuilder.h
index d757592..6e548b1 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.h
+++ b/media/libaaudio/src/core/AudioStreamBuilder.h
@@ -21,6 +21,7 @@
 
 #include <aaudio/AAudio.h>
 
+#include "AAudioStreamParameters.h"
 #include "AudioStream.h"
 
 namespace aaudio {
@@ -28,24 +29,12 @@
 /**
  * Factory class for an AudioStream.
  */
-class AudioStreamBuilder {
+class AudioStreamBuilder : public AAudioStreamParameters {
 public:
     AudioStreamBuilder();
 
     ~AudioStreamBuilder();
 
-    int getSamplesPerFrame() const {
-        return mSamplesPerFrame;
-    }
-
-    /**
-     * This is also known as channelCount.
-     */
-    AudioStreamBuilder* setSamplesPerFrame(int samplesPerFrame) {
-        mSamplesPerFrame = samplesPerFrame;
-        return this;
-    }
-
     aaudio_direction_t getDirection() const {
         return mDirection;
     }
@@ -55,33 +44,6 @@
         return this;
     }
 
-    int32_t getSampleRate() const {
-        return mSampleRate;
-    }
-
-    AudioStreamBuilder* setSampleRate(int32_t sampleRate) {
-        mSampleRate = sampleRate;
-        return this;
-    }
-
-    aaudio_format_t getFormat() const {
-        return mFormat;
-    }
-
-    AudioStreamBuilder *setFormat(aaudio_format_t format) {
-        mFormat = format;
-        return this;
-    }
-
-    aaudio_sharing_mode_t getSharingMode() const {
-        return mSharingMode;
-    }
-
-    AudioStreamBuilder* setSharingMode(aaudio_sharing_mode_t sharingMode) {
-        mSharingMode = sharingMode;
-        return this;
-    }
-
     bool isSharingModeMatchRequired() const {
         return mSharingModeMatchRequired;
     }
@@ -91,15 +53,6 @@
         return this;
     }
 
-    int32_t getBufferCapacity() const {
-        return mBufferCapacity;
-    }
-
-    AudioStreamBuilder* setBufferCapacity(int32_t frames) {
-        mBufferCapacity = frames;
-        return this;
-    }
-
     int32_t getPerformanceMode() const {
         return mPerformanceMode;
     }
@@ -109,15 +62,6 @@
         return this;
     }
 
-    int32_t getDeviceId() const {
-        return mDeviceId;
-    }
-
-    AudioStreamBuilder* setDeviceId(int32_t deviceId) {
-        mDeviceId = deviceId;
-        return this;
-    }
-
     AAudioStream_dataCallback getDataCallbackProc() const {
         return mDataCallbackProc;
     }
@@ -165,17 +109,11 @@
 
     aaudio_result_t build(AudioStream **streamPtr);
 
-    aaudio_result_t validate() const;
+    virtual aaudio_result_t validate() const override;
 
 private:
-    int32_t                    mSamplesPerFrame = AAUDIO_UNSPECIFIED;
-    int32_t                    mSampleRate = AAUDIO_UNSPECIFIED;
-    int32_t                    mDeviceId = AAUDIO_UNSPECIFIED;
-    aaudio_sharing_mode_t      mSharingMode = AAUDIO_SHARING_MODE_SHARED;
     bool                       mSharingModeMatchRequired = false; // must match sharing mode requested
-    aaudio_format_t            mFormat = AAUDIO_FORMAT_UNSPECIFIED;
     aaudio_direction_t         mDirection = AAUDIO_DIRECTION_OUTPUT;
-    int32_t                    mBufferCapacity = AAUDIO_UNSPECIFIED;
     aaudio_performance_mode_t  mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
 
     AAudioStream_dataCallback  mDataCallbackProc = nullptr;  // external callback functions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 38c9687..0df9a39 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1193,25 +1193,10 @@
         String8 value;
         if (param.get(String8(AudioParameter::keyBtNrec), value) == NO_ERROR) {
             bool btNrecIsOff = (value == AudioParameter::valueOff);
-            if (mBtNrecIsOff != btNrecIsOff) {
+            if (mBtNrecIsOff.exchange(btNrecIsOff) != btNrecIsOff) {
                 for (size_t i = 0; i < mRecordThreads.size(); i++) {
-                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
-                    audio_devices_t device = thread->inDevice();
-                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
-                    // collect all of the thread's session IDs
-                    KeyedVector<audio_session_t, bool> ids = thread->sessionIds();
-                    // suspend effects associated with those session IDs
-                    for (size_t j = 0; j < ids.size(); ++j) {
-                        audio_session_t sessionId = ids.keyAt(j);
-                        thread->setEffectSuspended(FX_IID_AEC,
-                                                   suspend,
-                                                   sessionId);
-                        thread->setEffectSuspended(FX_IID_NS,
-                                                   suspend,
-                                                   sessionId);
-                    }
+                    mRecordThreads.valueAt(i)->checkBtNrec();
                 }
-                mBtNrecIsOff = btNrecIsOff;
             }
         }
         String8 screenState;
@@ -3214,6 +3199,11 @@
 
 status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain)
 {
+    // clear possible suspended state before parking the chain so that it starts in default state
+    // when attached to a new record thread
+    chain->setEffectSuspended_l(FX_IID_AEC, false);
+    chain->setEffectSuspended_l(FX_IID_NS, false);
+
     audio_session_t session = chain->sessionId();
     ssize_t index = mOrphanEffectChains.indexOfKey(session);
     ALOGV("putOrphanEffectChain_l session %d index %zd", session, index);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 8a96c1d..9023b2d 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -350,12 +350,13 @@
                                         sync_event_callback_t callBack,
                                         const wp<RefBase>& cookie);
 
+    bool        btNrecIsOff() const { return mBtNrecIsOff.load(); }
+
+
 private:
 
                audio_mode_t getMode() const { return mMode; }
 
-                bool        btNrecIsOff() const { return mBtNrecIsOff; }
-
                             AudioFlinger() ANDROID_API;
     virtual                 ~AudioFlinger();
 
@@ -781,7 +782,7 @@
                 volatile atomic_uint_fast32_t       mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX];
 
                 audio_mode_t                        mMode;
-                bool                                mBtNrecIsOff;
+                std::atomic_bool                    mBtNrecIsOff;
 
                 // protected by mLock
                 Vector<AudioSessionRef*> mAudioSessionRefs;
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index f4428fe..f2c1c4f 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -21,10 +21,12 @@
 
 #include "Configuration.h"
 #include <utils/Log.h>
+#include <system/audio_effects/effect_aec.h>
+#include <system/audio_effects/effect_ns.h>
+#include <system/audio_effects/effect_visualizer.h>
 #include <audio_utils/primitives.h>
 #include <media/audiohal/EffectHalInterface.h>
 #include <media/audiohal/EffectsFactoryHalInterface.h>
-#include <system/audio_effects/effect_visualizer.h>
 
 #include "AudioFlinger.h"
 #include "ServiceUtilities.h"
@@ -1809,6 +1811,7 @@
                 idx_insert);
     }
     effect->configure();
+
     return NO_ERROR;
 }
 
@@ -2030,6 +2033,7 @@
             mSuspendedEffects.add(type->timeLow, desc);
             ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
         }
+
         if (desc->mRefCount++ == 0) {
             sp<EffectModule> effect = getEffectIfEnabled(type);
             if (effect != 0) {
@@ -2045,7 +2049,8 @@
         desc = mSuspendedEffects.valueAt(index);
         if (desc->mRefCount <= 0) {
             ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
-            desc->mRefCount = 1;
+            desc->mRefCount = 0;
+            return;
         }
         if (--desc->mRefCount == 0) {
             ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
@@ -2123,6 +2128,17 @@
 const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
 #endif //OPENSL_ES_H_
 
+/* static */
+bool AudioFlinger::EffectChain::isEffectEligibleForBtNrecSuspend(const effect_uuid_t *type)
+{
+    // Only NS and AEC are suspended when BtNRec is off
+    if ((memcmp(type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) ||
+        (memcmp(type, FX_IID_NS, sizeof(effect_uuid_t)) == 0)) {
+        return true;
+    }
+    return false;
+}
+
 bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
 {
     // auxiliary effects and visualizer are never suspended on output mix
@@ -2177,7 +2193,7 @@
         ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
             effect->desc().type.timeLow);
         sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
-        // if effect is requested to suspended but was not yet enabled, supend it now.
+        // if effect is requested to suspended but was not yet enabled, suspend it now.
         if (desc->mEffect == 0) {
             desc->mEffect = effect;
             effect->setEnabled(false);
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index e37529e..e29798b 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -331,7 +331,8 @@
     void setStrategy(uint32_t strategy)
             { mStrategy = strategy; }
 
-    // suspend effect of the given type
+    // suspend or restore effects of the specified type. The number of suspend requests is counted
+    // and restore occurs once all suspend requests are cancelled.
     void setEffectSuspended_l(const effect_uuid_t *type,
                               bool suspend);
     // suspend all eligible effects
@@ -372,7 +373,7 @@
     public:
         SuspendedEffectDesc() : mRefCount(0) {}
 
-        int mRefCount;
+        int mRefCount;   // > 0 when suspended
         effect_uuid_t mType;
         wp<EffectModule> mEffect;
     };
@@ -388,6 +389,8 @@
     // types or implementations from the suspend/restore mechanism.
     bool isEffectEligibleForSuspend(const effect_descriptor_t& desc);
 
+    static bool isEffectEligibleForBtNrecSuspend(const effect_uuid_t *type);
+
     void clearInputBuffer_l(const sp<ThreadBase>& thread);
 
     void setThread(const sp<ThreadBase>& thread);
@@ -414,6 +417,6 @@
              // mSuspendedEffects lists all effects currently suspended in the chain.
              // Use effect type UUID timelow field as key. There is no real risk of identical
              // timeLow fields among effect type UUIDs.
-             // Updated by updateSuspendedSessions_l() only.
+             // Updated by setEffectSuspended_l() and setEffectSuspendedAll_l() only.
              KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
 };
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d932483..a67acd6 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -991,13 +991,6 @@
     ALOGW("power manager service died !!!");
 }
 
-void AudioFlinger::ThreadBase::setEffectSuspended(
-        const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
-{
-    Mutex::Autolock _l(mLock);
-    setEffectSuspended_l(type, suspend, sessionId);
-}
-
 void AudioFlinger::ThreadBase::setEffectSuspended_l(
         const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
 {
@@ -1451,6 +1444,7 @@
     effect->setDevice(mInDevice);
     effect->setMode(mAudioFlinger->getMode());
     effect->setAudioSource(mAudioSource);
+
     return NO_ERROR;
 }
 
@@ -5947,6 +5941,7 @@
     // mPipeMemory
     // mFastCaptureNBLogWriter
     , mFastTrackAvail(false)
+    , mBtNrecSuspended(false)
 {
     snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id);
     mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
@@ -6711,12 +6706,6 @@
         }
         mTracks.add(track);
 
-        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
-        bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
-                        mAudioFlinger->btNrecIsOff();
-        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
-        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
-
         if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) {
             pid_t callingPid = IPCThreadState::self()->getCallingPid();
             // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
@@ -7079,6 +7068,26 @@
     buffer->frameCount = 0;
 }
 
+void AudioFlinger::RecordThread::checkBtNrec()
+{
+    Mutex::Autolock _l(mLock);
+    checkBtNrec_l();
+}
+
+void AudioFlinger::RecordThread::checkBtNrec_l()
+{
+    // disable AEC and NS if the device is a BT SCO headset supporting those
+    // pre processings
+    bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
+                        mAudioFlinger->btNrecIsOff();
+    if (mBtNrecSuspended.exchange(suspend) != suspend) {
+        for (size_t i = 0; i < mEffectChains.size(); i++) {
+            setEffectSuspended_l(FX_IID_AEC, suspend, mEffectChains[i]->sessionId());
+            setEffectSuspended_l(FX_IID_NS, suspend, mEffectChains[i]->sessionId());
+        }
+    }
+}
+
 
 bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
                                                         status_t& status)
@@ -7151,17 +7160,7 @@
             if (value != AUDIO_DEVICE_NONE) {
                 mPrevInDevice = value;
             }
-            // disable AEC and NS if the device is a BT SCO headset supporting those
-            // pre processings
-            if (mTracks.size() > 0) {
-                bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
-                                    mAudioFlinger->btNrecIsOff();
-                for (size_t i = 0; i < mTracks.size(); i++) {
-                    sp<RecordTrack> track = mTracks[i];
-                    setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
-                    setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
-                }
-            }
+            checkBtNrec_l();
         }
     }
     if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
@@ -7394,17 +7393,7 @@
         mEffectChains[i]->setDevice_l(mInDevice);
     }
 
-    // disable AEC and NS if the device is a BT SCO headset supporting those
-    // pre processings
-    if (mTracks.size() > 0) {
-        bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
-                            mAudioFlinger->btNrecIsOff();
-        for (size_t i = 0; i < mTracks.size(); i++) {
-            sp<RecordTrack> track = mTracks[i];
-            setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
-            setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
-        }
-    }
+    checkBtNrec_l();
 
     // store new source and send to effects
     if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 65266b0..062bad6 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -361,11 +361,6 @@
                 virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused)
                         { return 0; }
 
-                // suspend or restore effect according to the type of effect passed. a NULL
-                // type pointer means suspend all effects in the session
-                void setEffectSuspended(const effect_uuid_t *type,
-                                        bool suspend,
-                                        audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX);
                 // check if some effects must be suspended/restored when an effect is enabled
                 // or disabled
                 void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
@@ -417,10 +412,13 @@
                 void        releaseWakeLock_l();
                 void        updateWakeLockUids_l(const SortedVector<uid_t> &uids);
                 void        getPowerManager_l();
+                // suspend or restore effects of the specified type (or all if type is NULL)
+                // on a given session. The number of suspend requests is counted and restore
+                // occurs when all suspend requests are cancelled.
                 void setEffectSuspended_l(const effect_uuid_t *type,
                                           bool suspend,
                                           audio_session_t sessionId);
-                // updated mSuspendedSessions when an effect suspended or restored
+                // updated mSuspendedSessions when an effect is suspended or restored
                 void        updateSuspendedSessions_l(const effect_uuid_t *type,
                                                       bool suspend,
                                                       audio_session_t sessionId);
@@ -484,6 +482,7 @@
                 const sp<PMDeathRecipient> mDeathRecipient;
                 // list of suspended effects per session and per type. The first (outer) vector is
                 // keyed by session ID, the second (inner) by type UUID timeLow field
+                // Updated by updateSuspendedSessions_l() only.
                 KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > >
                                         mSuspendedSessions;
                 static const size_t     kLogSize = 4 * 1024;
@@ -1387,6 +1386,8 @@
                         }
     virtual bool        isOutput() const override { return false; }
 
+            void        checkBtNrec();
+
 private:
             // Enter standby if not already in standby, and set mStandby flag
             void    standbyIfNotAlreadyInStandby();
@@ -1394,6 +1395,8 @@
             // Call the HAL standby method unconditionally, and don't change mStandby flag
             void    inputStandBy();
 
+            void    checkBtNrec_l();
+
             AudioStreamIn                       *mInput;
             SortedVector < sp<RecordTrack> >    mTracks;
             // mActiveTracks has dual roles:  it indicates the current active track(s), and
@@ -1453,6 +1456,8 @@
             sp<NBLog::Writer>                   mFastCaptureNBLogWriter;
 
             bool                                mFastTrackAvail;    // true if fast track available
+            // common state to all record threads
+            std::atomic_bool                    mBtNrecSuspended;
 };
 
 class MmapThread : public ThreadBase
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index beec1f4..0807c0a 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -155,7 +155,7 @@
         return DEAD_OBJECT;
     }
 
-    mInterface = std::make_unique<HalInterface>(session, queue);
+    mInterface = new HalInterface(session, queue);
     std::string providerType;
     mVendorTagId = manager->getProviderTagIdLocked(mId.string());
 
@@ -184,7 +184,7 @@
     mTagMonitor.initialize(mVendorTagId);
 
     /** Start up request queue thread */
-    mRequestThread = new RequestThread(this, mStatusTracker, mInterface.get());
+    mRequestThread = new RequestThread(this, mStatusTracker, mInterface);
     res = mRequestThread->run(String8::format("C3Dev-%s-ReqQueue", mId.string()).string());
     if (res != OK) {
         SET_ERR_L("Unable to start request queue thread: %s (%d)",
@@ -3515,7 +3515,7 @@
 
 Camera3Device::RequestThread::RequestThread(wp<Camera3Device> parent,
         sp<StatusTracker> statusTracker,
-        HalInterface* interface) :
+        sp<HalInterface> interface) :
         Thread(/*canCallJava*/false),
         mParent(parent),
         mStatusTracker(statusTracker),
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index fb46d7e..0251d62 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -337,7 +337,7 @@
         std::vector<std::pair<int, uint64_t>> mFreedBuffers;
     };
 
-    std::unique_ptr<HalInterface> mInterface;
+    sp<HalInterface> mInterface;
 
     CameraMetadata             mDeviceInfo;
 
@@ -628,7 +628,7 @@
 
         RequestThread(wp<Camera3Device> parent,
                 sp<camera3::StatusTracker> statusTracker,
-                HalInterface* interface);
+                sp<HalInterface> interface);
         ~RequestThread();
 
         void     setNotificationListener(wp<NotificationListener> listener);
@@ -778,7 +778,7 @@
 
         wp<Camera3Device>  mParent;
         wp<camera3::StatusTracker>  mStatusTracker;
-        HalInterface*      mInterface;
+        sp<HalInterface>   mInterface;
 
         wp<NotificationListener> mListener;
 
diff --git a/services/camera/libcameraservice/device3/Camera3InputStream.cpp b/services/camera/libcameraservice/device3/Camera3InputStream.cpp
index 4eb15ad..35096eb 100644
--- a/services/camera/libcameraservice/device3/Camera3InputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3InputStream.cpp
@@ -293,8 +293,9 @@
 void Camera3InputStream::onBufferFreed(const wp<GraphicBuffer>& gb) {
     const sp<GraphicBuffer> buffer = gb.promote();
     if (buffer != nullptr) {
-        if (mBufferFreedListener != nullptr) {
-            mBufferFreedListener->onBufferFreed(mId, buffer->handle);
+        sp<Camera3StreamBufferFreedListener> callback = mBufferFreedListener.promote();
+        if (callback != nullptr) {
+            callback->onBufferFreed(mId, buffer->handle);
         }
     } else {
         ALOGE("%s: GraphicBuffer is freed before onBufferFreed callback finishes!", __FUNCTION__);
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
index e15aa43..b02cd6a 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
@@ -727,7 +727,7 @@
 
 void Camera3OutputStream::onBuffersRemovedLocked(
         const std::vector<sp<GraphicBuffer>>& removedBuffers) {
-    Camera3StreamBufferFreedListener* callback = mBufferFreedListener;
+    sp<Camera3StreamBufferFreedListener> callback = mBufferFreedListener.promote();
     if (callback != nullptr) {
         for (auto gb : removedBuffers) {
             callback->onBufferFreed(mId, gb->handle);
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.cpp b/services/camera/libcameraservice/device3/Camera3Stream.cpp
index 6789292..e77421a 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Stream.cpp
@@ -746,7 +746,7 @@
 }
 
 void Camera3Stream::setBufferFreedListener(
-        Camera3StreamBufferFreedListener* listener) {
+        wp<Camera3StreamBufferFreedListener> listener) {
     Mutex::Autolock l(mLock);
     // Only allow set listener during stream configuration because stream is guaranteed to be IDLE
     // at this state, so setBufferFreedListener won't collide with onBufferFreed callbacks
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.h b/services/camera/libcameraservice/device3/Camera3Stream.h
index b6c8396..0940d62 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.h
+++ b/services/camera/libcameraservice/device3/Camera3Stream.h
@@ -371,7 +371,7 @@
 
     // Setting listener will remove previous listener (if exists)
     virtual void     setBufferFreedListener(
-            Camera3StreamBufferFreedListener* listener) override;
+            wp<Camera3StreamBufferFreedListener> listener) override;
 
     /**
      * Return if the buffer queue of the stream is abandoned.
@@ -416,7 +416,7 @@
             android_dataspace dataSpace, camera3_stream_rotation_t rotation,
             int setId);
 
-    Camera3StreamBufferFreedListener* mBufferFreedListener;
+    wp<Camera3StreamBufferFreedListener> mBufferFreedListener;
 
     /**
      * Interface to be implemented by derived classes
diff --git a/services/camera/libcameraservice/device3/Camera3StreamBufferFreedListener.h b/services/camera/libcameraservice/device3/Camera3StreamBufferFreedListener.h
index 478a752..104cd22 100644
--- a/services/camera/libcameraservice/device3/Camera3StreamBufferFreedListener.h
+++ b/services/camera/libcameraservice/device3/Camera3StreamBufferFreedListener.h
@@ -24,7 +24,7 @@
 
 namespace camera3 {
 
-class Camera3StreamBufferFreedListener {
+class Camera3StreamBufferFreedListener : public virtual RefBase {
 public:
     // onBufferFreed is called when a buffer is no longer being managed
     // by this stream. This will not be called in events when all
diff --git a/services/camera/libcameraservice/device3/Camera3StreamInterface.h b/services/camera/libcameraservice/device3/Camera3StreamInterface.h
index c695a10..0544a1b 100644
--- a/services/camera/libcameraservice/device3/Camera3StreamInterface.h
+++ b/services/camera/libcameraservice/device3/Camera3StreamInterface.h
@@ -298,7 +298,7 @@
      * Client is responsible to keep the listener object alive throughout the lifecycle of this
      * Camera3Stream.
      */
-    virtual void setBufferFreedListener(Camera3StreamBufferFreedListener* listener) = 0;
+    virtual void setBufferFreedListener(wp<Camera3StreamBufferFreedListener> listener) = 0;
 };
 
 } // namespace camera3
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index 5895974..d726d46 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -84,7 +84,7 @@
     // Don't fall back to SHARED because that would cause recursion.
     builder.setSharingModeMatchRequired(true);
     builder.setDeviceId(mRequestedDeviceId);
-    builder.setFormat(configuration.getAudioFormat());
+    builder.setFormat(configuration.getFormat());
     builder.setSampleRate(configuration.getSampleRate());
     builder.setSamplesPerFrame(configuration.getSamplesPerFrame());
     builder.setDirection(getDirection());
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index b139119..a6fb129 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -95,7 +95,7 @@
     aaudio_direction_t direction = request.getDirection();
 
     // Fill in config
-    aaudio_format_t aaudioFormat = configurationInput.getAudioFormat();
+    aaudio_format_t aaudioFormat = configurationInput.getFormat();
     if (aaudioFormat == AAUDIO_UNSPECIFIED || aaudioFormat == AAUDIO_FORMAT_PCM_FLOAT) {
         aaudioFormat = AAUDIO_FORMAT_PCM_I16;
     }
@@ -210,7 +210,7 @@
     // Fill in AAudioStreamConfiguration
     configurationOutput.setSampleRate(mSampleRate);
     configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
-    configurationOutput.setAudioFormat(mAudioFormat);
+    configurationOutput.setFormat(mAudioFormat);
     configurationOutput.setDeviceId(deviceId);
 
     setState(AAUDIO_STREAM_STATE_OPEN);
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index fe488cb..5654113 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -112,7 +112,7 @@
     }
 
     // Is the request compatible with the shared endpoint?
-    mAudioFormat = configurationInput.getAudioFormat();
+    mAudioFormat = configurationInput.getFormat();
     if (mAudioFormat == AAUDIO_FORMAT_UNSPECIFIED) {
         mAudioFormat = AAUDIO_FORMAT_PCM_FLOAT;
     } else if (mAudioFormat != AAUDIO_FORMAT_PCM_FLOAT) {
@@ -169,7 +169,7 @@
     // Fill in configuration for client.
     configurationOutput.setSampleRate(mSampleRate);
     configurationOutput.setSamplesPerFrame(mSamplesPerFrame);
-    configurationOutput.setAudioFormat(mAudioFormat);
+    configurationOutput.setFormat(mAudioFormat);
     configurationOutput.setDeviceId(mServiceEndpoint->getDeviceId());
 
     result = mServiceEndpoint->registerStream(keep);