aaudio: indicate client UID and PID to audio flinger

Implement correct indication of client UID and PID to audio flinger
for AAudio MMAP streams in both exclusive mode and shared mode.
- Add start/stop client methods on MMAP streams used only when the MMAP
stream is in AAudio service and carries a mix of shared streams.
- Add "In Service'" indication from "client" side to AAudioServiceStreamMMAP
so that the behavior can be adapted accordingly.
- Modify logic on audio flinger side with regard to mmap tracks and
audio HAL stream activity:
  - use same audio session for all clients on a same stream to match
  audio policy logic to share same direct output stream for clients on same
  session. This is also more consistent with current volume and effect
  handling as all MMAP  clients sharing the same output stream have the
  same volume and use case.
  - start/stop the HAL when the stream is started/stopped with the initial client
  handle (returned when the stream is opened) but do not create a track.
  AAudioService implementation will always send an additional start command before
  first client starts and a stop command after last client stops,
  in both shared and exclusive mode.
  - start/stop a track only if the start/stop stream command is received
  with a handle different from the initial handle.
- Allow more than one active client from the same UID on a MMAP input in audio policy.

Bug: 62950008
Test: verify playback and capture in mmap mode
Change-Id: I86151bbb637ff172d2fd5f813056eab13a7bcd3c
diff --git a/include/media/MmapStreamInterface.h b/include/media/MmapStreamInterface.h
index 7dbc19e..d689e25 100644
--- a/include/media/MmapStreamInterface.h
+++ b/include/media/MmapStreamInterface.h
@@ -18,6 +18,7 @@
 #define ANDROID_AUDIO_MMAP_STREAM_INTERFACE_H
 
 #include <system/audio.h>
+#include <media/AudioClient.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
 
@@ -37,12 +38,6 @@
         DIRECTION_INPUT,       /**< open a capture mmap stream */
     } stream_direction_t;
 
-    class Client {
-     public:
-        uid_t clientUid;
-        pid_t clientPid;
-        String16 packageName;
-    };
     /**
      * Open a playback or capture stream in MMAP mode at the audio HAL.
      *
@@ -53,13 +48,14 @@
      * \param[in,out] config audio parameters (sampling rate, format ...) for the stream.
      *                       Requested parameters as input,
      *                       Actual parameters as output
-     * \param[in] client a Client struct describing the first client using this stream.
+     * \param[in] client a AudioClient struct describing the first client using this stream.
      * \param[in,out] deviceId audio device the stream should preferably be routed to/from
      *                       Requested as input,
      *                       Actual as output
      * \param[in] callback the MmapStreamCallback interface used by AudioFlinger to notify
      *                     condition changes affecting the stream operation
      * \param[out] interface the MmapStreamInterface interface controlling the created stream
+     * \param[out] same unique handle as the one used for the first client stream started.
      * \return OK if the stream was successfully created.
      *         NO_INIT if AudioFlinger is not properly initialized
      *         BAD_VALUE if the stream cannot be opened because of invalid arguments
@@ -68,10 +64,11 @@
     static status_t openMmapStream(stream_direction_t direction,
                                            const audio_attributes_t *attr,
                                            audio_config_base_t *config,
-                                           const Client& client,
+                                           const AudioClient& client,
                                            audio_port_handle_t *deviceId,
                                            const sp<MmapStreamCallback>& callback,
-                                           sp<MmapStreamInterface>& interface);
+                                           sp<MmapStreamInterface>& interface,
+                                           audio_port_handle_t *handle);
 
     /**
      * Retrieve information on the mmap buffer used for audio samples transfer.
@@ -105,13 +102,13 @@
      * Start a stream operating in mmap mode.
      * createMmapBuffer() must be called before calling start()
      *
-     * \param[in] client a Client struct describing the client starting on this stream.
+     * \param[in] client a AudioClient struct describing the client starting on this stream.
      * \param[out] handle unique handle for this instance. Used with stop().
      * \return OK in case of success.
      *         NO_INIT in case of initialization error
      *         INVALID_OPERATION if called out of sequence
      */
-    virtual status_t start(const Client& client, audio_port_handle_t *handle) = 0;
+    virtual status_t start(const AudioClient& client, audio_port_handle_t *handle) = 0;
 
     /**
      * Stop a stream operating in mmap mode.
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.h b/media/libaaudio/src/binding/AAudioBinderClient.h
index 469f0a8..89ae85c 100644
--- a/media/libaaudio/src/binding/AAudioBinderClient.h
+++ b/media/libaaudio/src/binding/AAudioBinderClient.h
@@ -97,6 +97,17 @@
     aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
                                                   pid_t clientThreadId) override;
 
+    aaudio_result_t startClient(aaudio_handle_t streamHandle __unused,
+                                      const android::AudioClient& client __unused,
+                                      audio_port_handle_t *clientHandle) override {
+        return AAUDIO_ERROR_UNAVAILABLE;
+    }
+
+    aaudio_result_t stopClient(aaudio_handle_t streamHandle __unused,
+                               audio_port_handle_t clientHandle __unused)  override {
+        return AAUDIO_ERROR_UNAVAILABLE;
+    }
+
     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 7368062..a64405b 100644
--- a/media/libaaudio/src/binding/AAudioServiceInterface.h
+++ b/media/libaaudio/src/binding/AAudioServiceInterface.h
@@ -18,6 +18,7 @@
 #define ANDROID_AAUDIO_BINDING_AAUDIO_SERVICE_INTERFACE_H
 
 #include <utils/StrongPointer.h>
+#include <media/AudioClient.h>
 
 #include "binding/AAudioServiceDefinitions.h"
 #include "binding/AAudioStreamRequest.h"
@@ -86,6 +87,13 @@
 
     virtual aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
                                                   pid_t clientThreadId) = 0;
+
+    virtual aaudio_result_t startClient(aaudio_handle_t streamHandle,
+                                      const android::AudioClient& client,
+                                      audio_port_handle_t *clientHandle) = 0;
+
+    virtual aaudio_result_t stopClient(aaudio_handle_t streamHandle,
+                                       audio_port_handle_t clientHandle) = 0;
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/binding/AAudioStreamRequest.cpp b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
index 8a765ad..abdcf5b 100644
--- a/media/libaaudio/src/binding/AAudioStreamRequest.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
@@ -52,8 +52,12 @@
     status = parcel->writeBool(mSharingModeMatchRequired);
     if (status != NO_ERROR) goto error;
 
+    status = parcel->writeBool(mInService);
+    if (status != NO_ERROR) goto error;
+
     status = mConfiguration.writeToParcel(parcel);
     if (status != NO_ERROR) goto error;
+
     return NO_ERROR;
 
 error:
@@ -74,8 +78,12 @@
     status = parcel->readBool(&mSharingModeMatchRequired);
     if (status != NO_ERROR) goto error;
 
+    status = parcel->readBool(&mInService);
+    if (status != NO_ERROR) goto error;
+
     status = mConfiguration.readFromParcel(parcel);
     if (status != NO_ERROR) goto error;
+
     return NO_ERROR;
 
 error:
@@ -91,5 +99,7 @@
     ALOGD("AAudioStreamRequest mUserId    = %d", mUserId);
     ALOGD("AAudioStreamRequest mProcessId = %d", mProcessId);
     ALOGD("AAudioStreamRequest mDirection = %d", mDirection);
+    ALOGD("AAudioStreamRequest mSharingModeMatchRequired = %d", mSharingModeMatchRequired);
+    ALOGD("AAudioStreamRequest mInService = %d", mInService);
     mConfiguration.dump();
 }
diff --git a/media/libaaudio/src/binding/AAudioStreamRequest.h b/media/libaaudio/src/binding/AAudioStreamRequest.h
index 462246b..b0fa96a 100644
--- a/media/libaaudio/src/binding/AAudioStreamRequest.h
+++ b/media/libaaudio/src/binding/AAudioStreamRequest.h
@@ -76,6 +76,14 @@
         return mConfiguration;
     }
 
+    bool isInService() const {
+        return mInService;
+    }
+
+    void setInService(bool inService) {
+        mInService = inService;
+    }
+
     virtual status_t writeToParcel(Parcel* parcel) const override;
 
     virtual status_t readFromParcel(const Parcel* parcel) override;
@@ -90,6 +98,7 @@
     pid_t                      mProcessId;
     aaudio_direction_t         mDirection;
     bool                       mSharingModeMatchRequired = false;
+    bool                       mInService = false; // Stream opened by AAudioservice
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index ff13fc2..7b01e44 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -16,7 +16,7 @@
 
 // This file is used in both client and server processes.
 // This is needed to make sense of the logs more easily.
-#define LOG_TAG (mInService ? "AAudioService" : "AAudio")
+#define LOG_TAG (mInService ? "AudioStreamInternal_Service" : "AudioStreamInternal_Client")
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -93,6 +93,7 @@
     request.setProcessId(getpid());
     request.setDirection(getDirection());
     request.setSharingModeMatchRequired(isSharingModeMatchRequired());
+    request.setInService(mInService);
 
     request.getConfiguration().setDeviceId(getDeviceId());
     request.getConfiguration().setSampleRate(getSampleRate());
@@ -326,6 +327,21 @@
     return mServiceInterface.unregisterAudioThread(mServiceStreamHandle, gettid());
 }
 
+aaudio_result_t AudioStreamInternal::startClient(const android::AudioClient& client,
+                                                 audio_port_handle_t *clientHandle) {
+    if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    return mServiceInterface.startClient(mServiceStreamHandle, client, clientHandle);
+}
+
+aaudio_result_t AudioStreamInternal::stopClient(audio_port_handle_t clientHandle) {
+    if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    return mServiceInterface.stopClient(mServiceStreamHandle, clientHandle);
+}
+
 aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId,
                            int64_t *framePosition,
                            int64_t *timeNanoseconds) {
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 257a702..109e425 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -87,6 +87,11 @@
     //PlayerBase virtuals
     virtual void destroy();
 
+    aaudio_result_t startClient(const android::AudioClient& client,
+                                audio_port_handle_t *clientHandle);
+
+    aaudio_result_t stopClient(audio_port_handle_t clientHandle);
+
 protected:
 
     aaudio_result_t processData(void *buffer,
@@ -170,7 +175,6 @@
 
     AudioEndpointParcelable  mEndPointParcelable; // description of the buffers filled by service
     EndpointDescriptor       mEndpointDescriptor; // buffer description with resolved addresses
-
 };
 
 } /* namespace aaudio */
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index d320320..ceba211 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -323,6 +323,7 @@
             return BAD_VALUE;
         }
         data.write(attr, sizeof(audio_attributes_t));
+        data.writeInt32(*input);
         data.writeInt32(session);
         data.writeInt32(pid);
         data.writeInt32(uid);
@@ -1024,6 +1025,7 @@
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             audio_attributes_t attr;
             data.read(&attr, sizeof(audio_attributes_t));
+            audio_io_handle_t input = (audio_io_handle_t)data.readInt32();
             audio_session_t session = (audio_session_t)data.readInt32();
             pid_t pid = (pid_t)data.readInt32();
             uid_t uid = (uid_t)data.readInt32();
@@ -1033,7 +1035,6 @@
             audio_input_flags_t flags = (audio_input_flags_t) data.readInt32();
             audio_port_handle_t selectedDeviceId = (audio_port_handle_t) data.readInt32();
             audio_port_handle_t portId = (audio_port_handle_t)data.readInt32();
-            audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
             status_t status = getInputForAttr(&attr, &input, session, pid, uid,
                                               &config,
                                               flags, &selectedDeviceId, &portId);
diff --git a/media/libaudioclient/include/media/AudioClient.h b/media/libaudioclient/include/media/AudioClient.h
new file mode 100644
index 0000000..9efd76d
--- /dev/null
+++ b/media/libaudioclient/include/media/AudioClient.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 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 ANDROID_AUDIO_CLIENT_H
+#define ANDROID_AUDIO_CLIENT_H
+
+#include <system/audio.h>
+#include <utils/String16.h>
+
+namespace android {
+
+class AudioClient {
+ public:
+    AudioClient() :
+        clientUid(-1), clientPid(-1), packageName("") {}
+
+    uid_t clientUid;
+    pid_t clientPid;
+    String16 packageName;
+};
+
+}; // namespace android
+
+#endif  // ANDROID_AUDIO_CLIENT_H
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d850aa9..0d6d5ee 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -260,10 +260,11 @@
 status_t MmapStreamInterface::openMmapStream(MmapStreamInterface::stream_direction_t direction,
                                              const audio_attributes_t *attr,
                                              audio_config_base_t *config,
-                                             const MmapStreamInterface::Client& client,
+                                             const AudioClient& client,
                                              audio_port_handle_t *deviceId,
                                              const sp<MmapStreamCallback>& callback,
-                                             sp<MmapStreamInterface>& interface)
+                                             sp<MmapStreamInterface>& interface,
+                                             audio_port_handle_t *handle)
 {
     sp<AudioFlinger> af;
     {
@@ -273,7 +274,7 @@
     status_t ret = NO_INIT;
     if (af != 0) {
         ret = af->openMmapStream(
-                direction, attr, config, client, deviceId, callback, interface);
+                direction, attr, config, client, deviceId, callback, interface, handle);
     }
     return ret;
 }
@@ -281,10 +282,11 @@
 status_t AudioFlinger::openMmapStream(MmapStreamInterface::stream_direction_t direction,
                                       const audio_attributes_t *attr,
                                       audio_config_base_t *config,
-                                      const MmapStreamInterface::Client& client,
+                                      const AudioClient& client,
                                       audio_port_handle_t *deviceId,
                                       const sp<MmapStreamCallback>& callback,
-                                      sp<MmapStreamInterface>& interface)
+                                      sp<MmapStreamInterface>& interface,
+                                      audio_port_handle_t *handle)
 {
     status_t ret = initCheck();
     if (ret != NO_ERROR) {
@@ -293,7 +295,7 @@
 
     audio_session_t sessionId = (audio_session_t) newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
     audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT;
-    audio_io_handle_t io;
+    audio_io_handle_t io = AUDIO_IO_HANDLE_NONE;
     audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
     if (direction == MmapStreamInterface::DIRECTION_OUTPUT) {
         audio_config_t fullConfig = AUDIO_CONFIG_INITIALIZER;
@@ -325,6 +327,7 @@
     if (thread != 0) {
         interface = new MmapThreadHandle(thread);
         thread->configure(attr, streamType, sessionId, callback, portId);
+        *handle = portId;
     } else {
         ret = NO_INIT;
     }
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 2e0bc66..8a96c1d 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -294,10 +294,11 @@
     status_t openMmapStream(MmapStreamInterface::stream_direction_t direction,
                             const audio_attributes_t *attr,
                             audio_config_base_t *config,
-                            const MmapStreamInterface::Client& client,
+                            const AudioClient& client,
                             audio_port_handle_t *deviceId,
                             const sp<MmapStreamCallback>& callback,
-                            sp<MmapStreamInterface>& interface);
+                            sp<MmapStreamInterface>& interface,
+                            audio_port_handle_t *handle);
 private:
     // FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed.
     static const size_t kLogMemorySize = 400 * 1024;
@@ -596,7 +597,7 @@
         virtual status_t createMmapBuffer(int32_t minSizeFrames,
                                           struct audio_mmap_buffer_info *info);
         virtual status_t getMmapPosition(struct audio_mmap_position *position);
-        virtual status_t start(const MmapStreamInterface::Client& client,
+        virtual status_t start(const AudioClient& client,
                                          audio_port_handle_t *handle);
         virtual status_t stop(audio_port_handle_t handle);
         virtual status_t standby();
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 6329f20..7871ae6 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -7544,7 +7544,7 @@
     return mThread->getMmapPosition(position);
 }
 
-status_t AudioFlinger::MmapThreadHandle::start(const MmapStreamInterface::Client& client,
+status_t AudioFlinger::MmapThreadHandle::start(const AudioClient& client,
         audio_port_handle_t *handle)
 
 {
@@ -7638,77 +7638,75 @@
     return mHalStream->getMmapPosition(position);
 }
 
-status_t AudioFlinger::MmapThread::start(const MmapStreamInterface::Client& client,
+status_t AudioFlinger::MmapThread::start(const AudioClient& client,
                                          audio_port_handle_t *handle)
 {
-    ALOGV("%s clientUid %d mStandby %d", __FUNCTION__, client.clientUid, mStandby);
+    ALOGV("%s clientUid %d mStandby %d mPortId %d *handle %d", __FUNCTION__,
+          client.clientUid, mStandby, mPortId, *handle);
     if (mHalStream == 0) {
         return NO_INIT;
     }
 
     status_t ret;
-    audio_session_t sessionId;
-    audio_port_handle_t portId;
 
-    if (mActiveTracks.size() == 0) {
+    if (*handle == mPortId) {
         // for the first track, reuse portId and session allocated when the stream was opened
         ret = mHalStream->start();
         if (ret != NO_ERROR) {
             ALOGE("%s: error mHalStream->start() = %d for first track", __FUNCTION__, ret);
             return ret;
         }
-        portId = mPortId;
-        sessionId = mSessionId;
         mStandby = false;
+        return NO_ERROR;
+    }
+
+    audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+
+    audio_io_handle_t io = mId;
+    if (isOutput()) {
+        audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+        config.sample_rate = mSampleRate;
+        config.channel_mask = mChannelMask;
+        config.format = mFormat;
+        audio_stream_type_t stream = streamType();
+        audio_output_flags_t flags =
+                (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT);
+        audio_port_handle_t deviceId = AUDIO_PORT_HANDLE_NONE;
+        ret = AudioSystem::getOutputForAttr(&mAttr, &io,
+                                            mSessionId,
+                                            &stream,
+                                            client.clientUid,
+                                            &config,
+                                            flags,
+                                            &deviceId,
+                                            &portId);
     } else {
-        // for other tracks than first one, get a new port ID from APM.
-        sessionId = (audio_session_t)mAudioFlinger->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
-        audio_io_handle_t io;
-        if (isOutput()) {
-            audio_config_t config = AUDIO_CONFIG_INITIALIZER;
-            config.sample_rate = mSampleRate;
-            config.channel_mask = mChannelMask;
-            config.format = mFormat;
-            audio_stream_type_t stream = streamType();
-            audio_output_flags_t flags =
-                    (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT);
-            audio_port_handle_t deviceId = AUDIO_PORT_HANDLE_NONE;
-            ret = AudioSystem::getOutputForAttr(&mAttr, &io,
-                                                sessionId,
-                                                &stream,
-                                                client.clientUid,
-                                                &config,
-                                                flags,
-                                                &deviceId,
-                                                &portId);
-        } else {
-            audio_config_base_t config;
-            config.sample_rate = mSampleRate;
-            config.channel_mask = mChannelMask;
-            config.format = mFormat;
-            audio_port_handle_t deviceId = AUDIO_PORT_HANDLE_NONE;
-            ret = AudioSystem::getInputForAttr(&mAttr, &io,
-                                                  sessionId,
-                                                  client.clientPid,
-                                                  client.clientUid,
-                                                  &config,
-                                                  AUDIO_INPUT_FLAG_MMAP_NOIRQ,
-                                                  &deviceId,
-                                                  &portId);
-        }
-        // APM should not chose a different input or output stream for the same set of attributes
-        // and audo configuration
-        if (ret != NO_ERROR || io != mId) {
-            ALOGE("%s: error getting output or input from APM (error %d, io %d expected io %d)",
-                  __FUNCTION__, ret, io, mId);
-            return BAD_VALUE;
-        }
+        audio_config_base_t config;
+        config.sample_rate = mSampleRate;
+        config.channel_mask = mChannelMask;
+        config.format = mFormat;
+        audio_port_handle_t deviceId = AUDIO_PORT_HANDLE_NONE;
+        ret = AudioSystem::getInputForAttr(&mAttr, &io,
+                                              mSessionId,
+                                              client.clientPid,
+                                              client.clientUid,
+                                              &config,
+                                              AUDIO_INPUT_FLAG_MMAP_NOIRQ,
+                                              &deviceId,
+                                              &portId);
+    }
+    // APM should not chose a different input or output stream for the same set of attributes
+    // and audo configuration
+    if (ret != NO_ERROR || io != mId) {
+        ALOGE("%s: error getting output or input from APM (error %d, io %d expected io %d)",
+              __FUNCTION__, ret, io, mId);
+        return BAD_VALUE;
     }
 
     if (isOutput()) {
-        ret = AudioSystem::startOutput(mId, streamType(), sessionId);
+        ret = AudioSystem::startOutput(mId, streamType(), mSessionId);
     } else {
-        ret = AudioSystem::startInput(mId, sessionId);
+        ret = AudioSystem::startInput(mId, mSessionId);
     }
 
     // abort if start is rejected by audio policy manager
@@ -7716,9 +7714,9 @@
         ALOGE("%s: error start rejected by AudioPolicyManager = %d", __FUNCTION__, ret);
         if (mActiveTracks.size() != 0) {
             if (isOutput()) {
-                AudioSystem::releaseOutput(mId, streamType(), sessionId);
+                AudioSystem::releaseOutput(mId, streamType(), mSessionId);
             } else {
-                AudioSystem::releaseInput(mId, sessionId);
+                AudioSystem::releaseInput(mId, mSessionId);
             }
         } else {
             mHalStream->stop();
@@ -7726,12 +7724,11 @@
         return PERMISSION_DENIED;
     }
 
-    sp<MmapTrack> track = new MmapTrack(this, mSampleRate, mFormat, mChannelMask, sessionId,
-                                        client.clientUid, client.clientPid,
-                                        portId);
+    sp<MmapTrack> track = new MmapTrack(this, mSampleRate, mFormat, mChannelMask, mSessionId,
+                                        client.clientUid, client.clientPid, portId);
 
     mActiveTracks.add(track);
-    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    sp<EffectChain> chain = getEffectChain_l(mSessionId);
     if (chain != 0) {
         chain->setStrategy(AudioSystem::getStrategyForStream(streamType()));
         chain->incTrackCnt();
@@ -7739,10 +7736,9 @@
     }
 
     *handle = portId;
-
     broadcast_l();
 
-    ALOGV("%s DONE handle %d stream %p", __FUNCTION__, portId, mHalStream.get());
+    ALOGV("%s DONE handle %d stream %p", __FUNCTION__, *handle, mHalStream.get());
 
     return NO_ERROR;
 }
@@ -7755,6 +7751,11 @@
         return NO_INIT;
     }
 
+    if (handle == mPortId) {
+        mHalStream->stop();
+        return NO_ERROR;
+    }
+
     sp<MmapTrack> track;
     for (const sp<MmapTrack> &t : mActiveTracks) {
         if (handle == t->portId()) {
@@ -7770,14 +7771,10 @@
 
     if (isOutput()) {
         AudioSystem::stopOutput(mId, streamType(), track->sessionId());
-        if (mActiveTracks.size() != 0) {
-            AudioSystem::releaseOutput(mId, streamType(), track->sessionId());
-        }
+        AudioSystem::releaseOutput(mId, streamType(), track->sessionId());
     } else {
         AudioSystem::stopInput(mId, track->sessionId());
-        if (mActiveTracks.size() != 0) {
-            AudioSystem::releaseInput(mId, track->sessionId());
-        }
+        AudioSystem::releaseInput(mId, track->sessionId());
     }
 
     sp<EffectChain> chain = getEffectChain_l(track->sessionId());
@@ -7788,9 +7785,6 @@
 
     broadcast_l();
 
-    if (mActiveTracks.size() == 0) {
-        mHalStream->stop();
-    }
     return NO_ERROR;
 }
 
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index cc11ddb..22d78eb 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1482,7 +1482,7 @@
     status_t createMmapBuffer(int32_t minSizeFrames,
                                       struct audio_mmap_buffer_info *info);
     status_t getMmapPosition(struct audio_mmap_position *position);
-    status_t start(const MmapStreamInterface::Client& client, audio_port_handle_t *handle);
+    status_t start(const AudioClient& client, audio_port_handle_t *handle);
     status_t stop(audio_port_handle_t handle);
     status_t standby();
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioSession.h b/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
index 2e653e2..cedf22d 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
@@ -52,6 +52,7 @@
     audio_channel_mask_t channelMask() const { return mConfig.channel_mask; }
     audio_input_flags_t flags() const { return mFlags; }
     uid_t uid() const { return mRecordClientInfo.uid; }
+    void setUid(uid_t uid) { mRecordClientInfo.uid = uid; }
     bool matches(const sp<AudioSession> &other) const;
     bool isSoundTrigger() const { return mIsSoundTrigger; }
     uint32_t openCount() const { return mOpenCount; } ;
@@ -65,7 +66,7 @@
     virtual void onSessionInfoUpdate() const;
 
 private:
-    const record_client_info_t mRecordClientInfo;
+    record_client_info_t mRecordClientInfo;
     const struct audio_config_base mConfig;
     const audio_input_flags_t mFlags;
     bool  mIsSoundTrigger;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 3dcb22d..55c364f 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1495,6 +1495,43 @@
             "session %d, flags %#x",
           attr->source, config->sample_rate, config->format, config->channel_mask, session, flags);
 
+    // special case for mmap capture: if an input IO handle is specified, we reuse this input if
+    // possible
+    if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) == AUDIO_INPUT_FLAG_MMAP_NOIRQ &&
+            *input != AUDIO_IO_HANDLE_NONE) {
+        ssize_t index = mInputs.indexOfKey(*input);
+        if (index < 0) {
+            ALOGW("getInputForAttr() unknown MMAP input %d", *input);
+            return BAD_VALUE;
+        }
+        sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+        sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
+        if (audioSession == 0) {
+            ALOGW("getInputForAttr() unknown session %d on input %d", session, *input);
+            return BAD_VALUE;
+        }
+        // For MMAP mode, the first call to getInputForAttr() is made on behalf of audioflinger.
+        // The second call is for the first active client and sets the UID. Any further call
+        // corresponds to a new client and is only permitted from the same UId.
+        if (audioSession->openCount() == 1) {
+            audioSession->setUid(uid);
+        } else if (audioSession->uid() != uid) {
+            ALOGW("getInputForAttr() bad uid %d for session %d uid %d",
+                  uid, session, audioSession->uid());
+            return INVALID_OPERATION;
+        }
+        audioSession->changeOpenCount(1);
+        *inputType = API_INPUT_LEGACY;
+        if (*portId == AUDIO_PORT_HANDLE_NONE) {
+            *portId = AudioPort::getNextUniqueId();
+        }
+        DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(inputDesc->mDevice);
+        *selectedDeviceId = inputDevices.size() > 0 ? inputDevices.itemAt(0)->getId()
+                : AUDIO_PORT_HANDLE_NONE;
+        ALOGI("%s reusing MMAP input %d for session %d", __FUNCTION__, *input, session);
+        return NO_ERROR;
+    }
+
     *input = AUDIO_IO_HANDLE_NONE;
     *inputType = API_INPUT_INVALID;
 
@@ -1898,6 +1935,11 @@
                 continue;
             }
 
+            if ((audioSession->flags() & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0 &&
+                    activeDesc->getId() == inputDesc->getId()) {
+                continue;
+            }
+
             audio_source_t activeSource = activeDesc->inputSource(true);
             if (audioSession->inputSource() == AUDIO_SOURCE_HOTWORD) {
                 if (activeSource == AUDIO_SOURCE_HOTWORD) {
diff --git a/services/oboeservice/AAudioEndpointManager.cpp b/services/oboeservice/AAudioEndpointManager.cpp
index 02d4a19..ec2f5b9 100644
--- a/services/oboeservice/AAudioEndpointManager.cpp
+++ b/services/oboeservice/AAudioEndpointManager.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudioService"
+#define LOG_TAG "AAudioEndpointManager"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -104,6 +104,8 @@
 
     // If we can't find an existing one then open a new one.
     if (endpoint == nullptr) {
+        // we must call openStream with audioserver identity
+        int64_t token = IPCThreadState::self()->clearCallingIdentity();
         switch(direction) {
             case AAUDIO_DIRECTION_INPUT:
                 capture = new AAudioServiceEndpointCapture(audioService);
@@ -138,6 +140,7 @@
         }
         ALOGD("AAudioEndpointManager::openEndpoint(), created %p for device = %d, dir = %d",
               endpoint, configuration.getDeviceId(), (int)direction);
+        IPCThreadState::self()->restoreCallingIdentity(token);
     }
 
     if (endpoint != nullptr) {
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index 669bb54..3992719 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -50,8 +50,9 @@
 
 android::AAudioService::AAudioService()
     : BnAAudioService() {
-    mCachedProcessId = getpid();
-    mCachedUserId = getuid();   // TODO consider using geteuid()
+    mAudioClient.clientUid = getuid();   // TODO consider using geteuid()
+    mAudioClient.clientPid = getpid();
+    mAudioClient.packageName = String16("");
     AAudioClientTracker::getInstance().setAAudioService(this);
 }
 
@@ -92,7 +93,7 @@
 
     // Enforce limit on client processes.
     pid_t pid = request.getProcessId();
-    if (pid != mCachedProcessId) {
+    if (pid != mAudioClient.clientPid) {
         int32_t count = AAudioClientTracker::getInstance().getStreamCount(pid);
         if (count >= MAX_STREAMS_PER_PROCESS) {
             ALOGE("AAudioService::openStream(): exceeded max streams per process %d >= %d",
@@ -107,7 +108,13 @@
     }
 
     if (sharingMode == AAUDIO_SHARING_MODE_EXCLUSIVE) {
-        serviceStream = new AAudioServiceStreamMMAP(mCachedUserId);
+        // only trust audioserver for in service indication
+        bool inService = false;
+        if (mAudioClient.clientPid == IPCThreadState::self()->getCallingPid() &&
+                mAudioClient.clientUid == IPCThreadState::self()->getCallingUid()) {
+            inService = request.isInService();
+        }
+        serviceStream = new AAudioServiceStreamMMAP(mAudioClient, inService);
         result = serviceStream->open(request, configurationOutput);
         if (result != AAUDIO_OK) {
             // fall back to using a shared stream
@@ -132,8 +139,6 @@
               result, AAudio_convertResultToText(result));
         return result;
     } else {
-        const uid_t ownerUserId = request.getUserId(); // only set by service, not by client
-        serviceStream->setOwnerUserId(ownerUserId);
         aaudio_handle_t handle = mHandleTracker.put(AAUDIO_HANDLE_TYPE_STREAM, serviceStream.get());
         if (handle < 0) {
             ALOGE("AAudioService::openStream(): handle table full");
@@ -143,7 +148,6 @@
             ALOGD("AAudioService::openStream(): handle = 0x%08X", handle);
             serviceStream->setHandle(handle);
             pid_t pid = request.getProcessId();
-            serviceStream->setOwnerProcessId(pid);
             AAudioClientTracker::getInstance().registerClientStream(pid, serviceStream);
         }
         return handle;
@@ -181,8 +185,8 @@
         const uid_t callingUserId = IPCThreadState::self()->getCallingUid();
         const uid_t ownerUserId = serviceStream->getOwnerUserId();
         bool callerOwnsIt = callingUserId == ownerUserId;
-        bool serverCalling = callingUserId == mCachedUserId;
-        bool serverOwnsIt = ownerUserId == mCachedUserId;
+        bool serverCalling = callingUserId == mAudioClient.clientUid;
+        bool serverOwnsIt = ownerUserId == mAudioClient.clientUid;
         bool allowed = callerOwnsIt || serverCalling || serverOwnsIt;
         if (!allowed) {
             ALOGE("AAudioService: calling uid %d cannot access stream 0x%08X owned by %d",
@@ -212,6 +216,7 @@
         ALOGE("AAudioService::startStream(), illegal stream handle = 0x%0x", streamHandle);
         return AAUDIO_ERROR_INVALID_HANDLE;
     }
+
     aaudio_result_t result = serviceStream->start();
     return result;
 }
@@ -286,3 +291,26 @@
     serviceStream->setRegisteredThread(0);
     return AAUDIO_OK;
 }
+
+aaudio_result_t AAudioService::startClient(aaudio_handle_t streamHandle,
+                                  const android::AudioClient& client,
+                                  audio_port_handle_t *clientHandle) {
+    AAudioServiceStreamBase *serviceStream = convertHandleToServiceStream(streamHandle);
+    if (serviceStream == nullptr) {
+        ALOGE("AAudioService::startClient(), illegal stream handle = 0x%0x",
+              streamHandle);
+        return AAUDIO_ERROR_INVALID_HANDLE;
+    }
+    return serviceStream->startClient(client, clientHandle);
+}
+
+aaudio_result_t AAudioService::stopClient(aaudio_handle_t streamHandle,
+                                          audio_port_handle_t clientHandle) {
+    AAudioServiceStreamBase *serviceStream = convertHandleToServiceStream(streamHandle);
+    if (serviceStream == nullptr) {
+        ALOGE("AAudioService::stopClient(), illegal stream handle = 0x%0x",
+              streamHandle);
+        return AAUDIO_ERROR_INVALID_HANDLE;
+    }
+    return serviceStream->stopClient(clientHandle);
+}
diff --git a/services/oboeservice/AAudioService.h b/services/oboeservice/AAudioService.h
index f84ac4c..8421efc 100644
--- a/services/oboeservice/AAudioService.h
+++ b/services/oboeservice/AAudioService.h
@@ -21,6 +21,7 @@
 #include <pthread.h>
 
 #include <binder/BinderService.h>
+#include <media/AudioClient.h>
 
 #include <aaudio/AAudio.h>
 #include "utility/HandleTracker.h"
@@ -72,14 +73,20 @@
     virtual aaudio_result_t unregisterAudioThread(aaudio_handle_t streamHandle,
                                                   pid_t tid);
 
+    virtual aaudio_result_t startClient(aaudio_handle_t streamHandle,
+                                      const android::AudioClient& client,
+                                      audio_port_handle_t *clientHandle);
+
+    virtual aaudio_result_t stopClient(aaudio_handle_t streamHandle,
+                                       audio_port_handle_t clientHandle);
+
 private:
 
     aaudio::AAudioServiceStreamBase *convertHandleToServiceStream(aaudio_handle_t streamHandle) const;
 
     HandleTracker mHandleTracker;
 
-    uid_t   mCachedUserId = -1;
-    pid_t   mCachedProcessId = -1;
+    android::AudioClient mAudioClient;
 
     enum constants {
         DEFAULT_AUDIO_PRIORITY = 2
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index b519829..5895974 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudioService"
+#define LOG_TAG "AAudioServiceEndpoint"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -94,7 +94,7 @@
 }
 
 aaudio_result_t AAudioServiceEndpoint::close() {
-    return getStreamInternal()->close();
+     return getStreamInternal()->close();
 }
 
 // TODO, maybe use an interface to reduce exposure
@@ -113,26 +113,25 @@
 
 aaudio_result_t AAudioServiceEndpoint::startStream(sp<AAudioServiceStreamShared> sharedStream) {
     // TODO use real-time technique to avoid mutex, eg. atomic command FIFO
+    aaudio_result_t result = AAUDIO_OK;
     std::lock_guard<std::mutex> lock(mLockStreams);
-    mRunningStreams.push_back(sharedStream);
-    if (mRunningStreams.size() == 1) {
+    if (++mRunningStreams == 1) {
+        result = getStreamInternal()->requestStart();
         startSharingThread_l();
     }
-    return AAUDIO_OK;
+    return result;
 }
 
 aaudio_result_t AAudioServiceEndpoint::stopStream(sp<AAudioServiceStreamShared> sharedStream) {
     int numRunningStreams = 0;
     {
         std::lock_guard<std::mutex> lock(mLockStreams);
-        mRunningStreams.erase(
-                std::remove(mRunningStreams.begin(), mRunningStreams.end(), sharedStream),
-                mRunningStreams.end());
-        numRunningStreams = mRunningStreams.size();
+        numRunningStreams = --mRunningStreams;
     }
     if (numRunningStreams == 0) {
         // Don't call this under a lock because the callbackLoop also uses the lock.
         stopSharingThread();
+        getStreamInternal()->requestStop();
     }
     return AAUDIO_OK;
 }
@@ -163,11 +162,8 @@
 
 void AAudioServiceEndpoint::disconnectRegisteredStreams() {
     std::lock_guard<std::mutex> lock(mLockStreams);
-    for(auto baseStream : mRunningStreams) {
-        baseStream->onStop();
-    }
-    mRunningStreams.clear();
     for(auto sharedStream : mRegisteredStreams) {
+        sharedStream->stop();
         sharedStream->disconnect();
     }
     mRegisteredStreams.clear();
@@ -189,3 +185,4 @@
 
     return true;
 }
+
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index a78d3fa..ed995e5 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -79,7 +79,7 @@
 
     std::vector<android::sp<AAudioServiceStreamShared>> mRegisteredStreams;
 
-    std::vector<android::sp<AAudioServiceStreamShared>> mRunningStreams;
+    size_t                   mRunningStreams = 0;
 
 private:
     aaudio_result_t startSharingThread_l();
diff --git a/services/oboeservice/AAudioServiceEndpointCapture.cpp b/services/oboeservice/AAudioServiceEndpointCapture.cpp
index a144c54..6a37330 100644
--- a/services/oboeservice/AAudioServiceEndpointCapture.cpp
+++ b/services/oboeservice/AAudioServiceEndpointCapture.cpp
@@ -57,9 +57,7 @@
 void *AAudioServiceEndpointCapture::callbackLoop() {
     ALOGD("AAudioServiceEndpointCapture(): callbackLoop() entering");
     int32_t underflowCount = 0;
-
-    aaudio_result_t result = getStreamInternal()->requestStart();
-
+    aaudio_result_t result = AAUDIO_OK;
     int64_t timeoutNanos = getStreamInternal()->calculateReasonableTimeout();
 
     // result might be a frame count
@@ -78,21 +76,21 @@
         // Distribute data to each active stream.
         { // use lock guard
             std::lock_guard <std::mutex> lock(mLockStreams);
-            for (sp<AAudioServiceStreamShared> sharedStream : mRunningStreams) {
-                FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
-                if (fifo->getFifoControllerBase()->getEmptyFramesAvailable() <
-                    getFramesPerBurst()) {
-                    underflowCount++;
-                } else {
-                    fifo->write(mDistributionBuffer, getFramesPerBurst());
+            for (sp<AAudioServiceStreamShared> sharedStream : mRegisteredStreams) {
+                if (sharedStream->isRunning()) {
+                    FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
+                    if (fifo->getFifoControllerBase()->getEmptyFramesAvailable() <
+                        getFramesPerBurst()) {
+                        underflowCount++;
+                    } else {
+                        fifo->write(mDistributionBuffer, getFramesPerBurst());
+                    }
+                    sharedStream->markTransferTime(AudioClock::getNanoseconds());
                 }
-                sharedStream->markTransferTime(AudioClock::getNanoseconds());
             }
         }
     }
 
-    result = getStreamInternal()->requestStop();
-
     ALOGD("AAudioServiceEndpointCapture(): callbackLoop() exiting, %d underflows", underflowCount);
     return NULL; // TODO review
 }
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.cpp b/services/oboeservice/AAudioServiceEndpointPlay.cpp
index 1afcc1e..86ccde0 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.cpp
+++ b/services/oboeservice/AAudioServiceEndpointPlay.cpp
@@ -68,9 +68,7 @@
 void *AAudioServiceEndpointPlay::callbackLoop() {
     ALOGD("AAudioServiceEndpointPlay(): callbackLoop() entering");
     int32_t underflowCount = 0;
-
-    aaudio_result_t result = getStreamInternal()->requestStart();
-
+    aaudio_result_t result = AAUDIO_OK;
     int64_t timeoutNanos = getStreamInternal()->calculateReasonableTimeout();
 
     // result might be a frame count
@@ -79,13 +77,15 @@
         mMixer.clear();
         { // use lock guard
             std::lock_guard <std::mutex> lock(mLockStreams);
-            for (sp<AAudioServiceStreamShared> sharedStream : mRunningStreams) {
-                FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
-                float volume = 1.0; // to match legacy volume
-                bool underflowed = mMixer.mix(fifo, volume);
-                underflowCount += underflowed ? 1 : 0;
-                // TODO log underflows in each stream
-                sharedStream->markTransferTime(AudioClock::getNanoseconds());
+            for (sp<AAudioServiceStreamShared> sharedStream : mRegisteredStreams) {
+                if (sharedStream->isRunning()) {
+                    FifoBuffer *fifo = sharedStream->getDataFifoBuffer();
+                    float volume = 1.0; // to match legacy volume
+                    bool underflowed = mMixer.mix(fifo, volume);
+                    underflowCount += underflowed ? 1 : 0;
+                    // TODO log underflows in each stream
+                    sharedStream->markTransferTime(AudioClock::getNanoseconds());
+                }
             }
         }
 
@@ -102,8 +102,6 @@
         }
     }
 
-    result = getStreamInternal()->requestStop();
-
     ALOGD("AAudioServiceEndpointPlay(): callbackLoop() exiting, %d underflows", underflowCount);
     return NULL; // TODO review
 }
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 52b1801..e0f6ad4 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudioService"
+#define LOG_TAG "AAudioServiceStreamBase"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -38,6 +38,9 @@
 AAudioServiceStreamBase::AAudioServiceStreamBase()
         : mUpMessageQueue(nullptr)
         , mAAudioThread() {
+    mMmapClient.clientUid = -1;
+    mMmapClient.clientPid = -1;
+    mMmapClient.packageName = String16("");
 }
 
 AAudioServiceStreamBase::~AAudioServiceStreamBase() {
@@ -45,7 +48,8 @@
     // If the stream is deleted when OPEN or in use then audio resources will leak.
     // This would indicate an internal error. So we want to find this ASAP.
     LOG_ALWAYS_FATAL_IF(!(mState == AAUDIO_STREAM_STATE_CLOSED
-                        || mState == AAUDIO_STREAM_STATE_UNINITIALIZED),
+                        || mState == AAUDIO_STREAM_STATE_UNINITIALIZED
+                        || mState == AAUDIO_STREAM_STATE_DISCONNECTED),
                         "service stream still open, state = %d", mState);
 }
 
@@ -58,13 +62,18 @@
     result << "      framesPerBurst = " << mFramesPerBurst << "\n";
     result << "      channelCount   = " << mSamplesPerFrame << "\n";
     result << "      capacityFrames = " << mCapacityInFrames << "\n";
-    result << "      owner uid      = " << mOwnerUserId << "\n";
+    result << "      owner uid      = " << mMmapClient.clientUid << "\n";
 
     return result.str();
 }
 
 aaudio_result_t AAudioServiceStreamBase::open(const aaudio::AAudioStreamRequest &request,
                      aaudio::AAudioStreamConfiguration &configurationOutput) {
+
+    mMmapClient.clientUid = request.getUserId();
+    mMmapClient.clientPid = request.getProcessId();
+    mMmapClient.packageName.setTo(String16("")); // FIXME what should we do here?
+
     std::lock_guard<std::mutex> lock(mLockUpMessageQueue);
     if (mUpMessageQueue != nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
@@ -86,6 +95,9 @@
 }
 
 aaudio_result_t AAudioServiceStreamBase::start() {
+    if (isRunning()) {
+        return AAUDIO_OK;
+    }
     sendServiceEvent(AAUDIO_SERVICE_EVENT_STARTED);
     mState = AAUDIO_STREAM_STATE_STARTED;
     mThreadEnabled.store(true);
@@ -94,32 +106,34 @@
 
 aaudio_result_t AAudioServiceStreamBase::pause() {
     aaudio_result_t result = AAUDIO_OK;
-    if (isRunning()) {
-        sendCurrentTimestamp();
-        mThreadEnabled.store(false);
-        result = mAAudioThread.stop();
-        if (result != AAUDIO_OK) {
-            disconnect();
-            return result;
-        }
-        sendServiceEvent(AAUDIO_SERVICE_EVENT_PAUSED);
+    if (!isRunning()) {
+        return result;
     }
+    sendCurrentTimestamp();
+    mThreadEnabled.store(false);
+    result = mAAudioThread.stop();
+    if (result != AAUDIO_OK) {
+        disconnect();
+        return result;
+    }
+    sendServiceEvent(AAUDIO_SERVICE_EVENT_PAUSED);
     mState = AAUDIO_STREAM_STATE_PAUSED;
     return result;
 }
 
 aaudio_result_t AAudioServiceStreamBase::stop() {
     aaudio_result_t result = AAUDIO_OK;
-    if (isRunning()) {
-        // TODO wait for data to be played out
-        sendCurrentTimestamp(); // warning - this calls a virtual function
-        result = stopTimestampThread();
-        if (result != AAUDIO_OK) {
-            disconnect();
-            return result;
-        }
-        sendServiceEvent(AAUDIO_SERVICE_EVENT_STOPPED);
+    if (!isRunning()) {
+        return result;
     }
+    // TODO wait for data to be played out
+    sendCurrentTimestamp(); // warning - this calls a virtual function
+    result = stopTimestampThread();
+    if (result != AAUDIO_OK) {
+        disconnect();
+        return result;
+    }
+    sendServiceEvent(AAUDIO_SERVICE_EVENT_STOPPED);
     mState = AAUDIO_STREAM_STATE_STOPPED;
     return result;
 }
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index c7df6f3..93a522e 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -27,6 +27,7 @@
 #include "binding/AudioEndpointParcelable.h"
 #include "binding/AAudioServiceMessage.h"
 #include "utility/AAudioUtilities.h"
+#include <media/AudioClient.h>
 
 #include "SharedRingBuffer.h"
 #include "AAudioThread.h"
@@ -85,9 +86,19 @@
      */
     virtual aaudio_result_t flush();
 
+    virtual aaudio_result_t startClient(const android::AudioClient& client __unused,
+                                        audio_port_handle_t *clientHandle __unused) {
+        return AAUDIO_ERROR_UNAVAILABLE;
+    }
+
+    virtual aaudio_result_t stopClient(audio_port_handle_t clientHandle __unused) {
+        return AAUDIO_ERROR_UNAVAILABLE;
+    }
+
     bool isRunning() const {
         return mState == AAUDIO_STREAM_STATE_STARTED;
     }
+
     // -------------------------------------------------------------------
 
     /**
@@ -124,17 +135,11 @@
     void disconnect();
 
     uid_t getOwnerUserId() const {
-        return mOwnerUserId;
-    }
-    void setOwnerUserId(uid_t uid) {
-        mOwnerUserId = uid;
+        return mMmapClient.clientUid;
     }
 
     pid_t getOwnerProcessId() const {
-        return mOwnerProcessId;
-    }
-    void setOwnerProcessId(pid_t pid) {
-        mOwnerProcessId = pid;
+        return mMmapClient.clientPid;
     }
 
     aaudio_handle_t getHandle() const {
@@ -164,24 +169,25 @@
 
     aaudio_stream_state_t   mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
 
-    pid_t              mRegisteredClientThread = ILLEGAL_THREAD_ID;
+    pid_t                   mRegisteredClientThread = ILLEGAL_THREAD_ID;
 
-    SharedRingBuffer*  mUpMessageQueue;
-    std::mutex         mLockUpMessageQueue;
+    SharedRingBuffer*       mUpMessageQueue;
+    std::mutex              mLockUpMessageQueue;
 
-    AAudioThread       mAAudioThread;
+    AAudioThread            mAAudioThread;
     // This is used by one thread to tell another thread to exit. So it must be atomic.
-    std::atomic<bool>  mThreadEnabled;
+    std::atomic<bool>       mThreadEnabled;
 
-    aaudio_format_t    mAudioFormat = AAUDIO_FORMAT_UNSPECIFIED;
-    int32_t            mFramesPerBurst = 0;
-    int32_t            mSamplesPerFrame = AAUDIO_UNSPECIFIED;
-    int32_t            mSampleRate = AAUDIO_UNSPECIFIED;
-    int32_t            mCapacityInFrames = AAUDIO_UNSPECIFIED;
-    uid_t              mOwnerUserId = -1;
-    pid_t              mOwnerProcessId = -1;
+    aaudio_format_t         mAudioFormat = AAUDIO_FORMAT_UNSPECIFIED;
+    int32_t                 mFramesPerBurst = 0;
+    int32_t                 mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+    int32_t                 mSampleRate = AAUDIO_UNSPECIFIED;
+    int32_t                 mCapacityInFrames = AAUDIO_UNSPECIFIED;
+    android::AudioClient    mMmapClient;
+    audio_port_handle_t     mClientHandle = AUDIO_PORT_HANDLE_NONE;
+
 private:
-    aaudio_handle_t    mHandle = -1;
+    aaudio_handle_t         mHandle = -1;
 };
 
 } /* namespace aaudio */
diff --git a/services/oboeservice/AAudioServiceStreamExclusive.h b/services/oboeservice/AAudioServiceStreamExclusive.h
deleted file mode 100644
index db382a3..0000000
--- a/services/oboeservice/AAudioServiceStreamExclusive.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (C) 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_AAUDIO_SERVICE_STREAM_EXCLUSIVE_H
-#define AAUDIO_AAUDIO_SERVICE_STREAM_EXCLUSIVE_H
-
-#include "AAudioServiceStreamMMAP.h"
-
-namespace aaudio {
-
-/**
- * Exclusive mode stream in the AAudio service.
- *
- * This is currently a stub.
- * We may move code from AAudioServiceStreamMMAP into this class.
- * If not, then it will be removed.
- */
-class AAudioServiceStreamExclusive : public AAudioServiceStreamMMAP {
-
-public:
-    AAudioServiceStreamExclusive() {};
-    virtual ~AAudioServiceStreamExclusive() = default;
-};
-
-} /* namespace aaudio */
-
-#endif //AAUDIO_AAUDIO_SERVICE_STREAM_EXCLUSIVE_H
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index 1b80486..68be3c3 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudioService"
+#define LOG_TAG "AAudioServiceStreamMMAP"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -41,19 +41,21 @@
  * Service Stream that uses an MMAP buffer.
  */
 
-AAudioServiceStreamMMAP::AAudioServiceStreamMMAP(uid_t serviceUid)
+AAudioServiceStreamMMAP::AAudioServiceStreamMMAP(const android::AudioClient& serviceClient,
+                                                 bool inService)
         : AAudioServiceStreamBase()
         , mMmapStreamCallback(new MyMmapStreamCallback(*this))
         , mPreviousFrameCounter(0)
         , mMmapStream(nullptr)
-        , mCachedUserId(serviceUid) {
+        , mServiceClient(serviceClient)
+        , mInService(inService) {
 }
 
 aaudio_result_t AAudioServiceStreamMMAP::close() {
     if (mState == AAUDIO_STREAM_STATE_CLOSED) {
         return AAUDIO_OK;
     }
-
+    stop();
     if (mMmapStream != 0) {
         mMmapStream.clear(); // TODO review. Is that all we have to do?
         // Apparently the above close is asynchronous. An attempt to open a new device
@@ -90,9 +92,6 @@
 
     const AAudioStreamConfiguration &configurationInput = request.getConstantConfiguration();
     audio_port_handle_t deviceId = configurationInput.getDeviceId();
-
-    mMmapClient.clientUid = request.getUserId();
-    mMmapClient.clientPid = request.getProcessId();
     aaudio_direction_t direction = request.getDirection();
 
     // Fill in config
@@ -123,8 +122,6 @@
         return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
     }
 
-    mMmapClient.packageName.setTo(String16("aaudio_service")); // FIXME what should we do here?
-
     MmapStreamInterface::stream_direction_t streamDirection = (direction == AAUDIO_DIRECTION_OUTPUT)
         ? MmapStreamInterface::DIRECTION_OUTPUT : MmapStreamInterface::DIRECTION_INPUT;
 
@@ -135,7 +132,8 @@
                                                           mMmapClient,
                                                           &deviceId,
                                                           mMmapStreamCallback,
-                                                          mMmapStream);
+                                                          mMmapStream,
+                                                          &mPortHandle);
     if (status != OK) {
         ALOGE("openMmapStream returned status %d", status);
         return AAUDIO_ERROR_UNAVAILABLE;
@@ -172,7 +170,7 @@
         mCapacityInFrames = -mCapacityInFrames;
     } else {
         // exclusive mode is only possible if the final fd destination is inside audioserver
-        if ((mMmapClient.clientUid != mCachedUserId) &&
+        if ((mMmapClient.clientUid != mServiceClient.clientUid) &&
                 configurationInput.getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE) {
             // Fallback is handled by caller but indicate what is possible in case
             // this is used in the future
@@ -223,15 +221,21 @@
  * Start the flow of data.
  */
 aaudio_result_t AAudioServiceStreamMMAP::start() {
+    if (isRunning()) {
+        return AAUDIO_OK;
+    }
     if (mMmapStream == nullptr) return AAUDIO_ERROR_NULL;
     aaudio_result_t result;
-    status_t status = mMmapStream->start(mMmapClient, &mPortHandle);
+    status_t status = mMmapStream->start(mServiceClient, &mPortHandle);
     if (status != OK) {
         ALOGE("AAudioServiceStreamMMAP::start() mMmapStream->start() returned %d", status);
         disconnect();
         result = AAudioConvert_androidToAAudioResult(status);
     } else {
         result = AAudioServiceStreamBase::start();
+        if (!mInService && result == AAUDIO_OK) {
+            startClient(mMmapClient, &mClientHandle);
+        }
     }
     return result;
 }
@@ -240,18 +244,28 @@
  * Stop the flow of data such that start() can resume with loss of data.
  */
 aaudio_result_t AAudioServiceStreamMMAP::pause() {
+    if (!isRunning()) {
+        return AAUDIO_OK;
+    }
     if (mMmapStream == nullptr) return AAUDIO_ERROR_NULL;
-
     aaudio_result_t result1 = AAudioServiceStreamBase::pause();
+    if (!mInService) {
+        stopClient(mClientHandle);
+    }
     status_t status = mMmapStream->stop(mPortHandle);
     mFramesRead.reset32();
     return (result1 != AAUDIO_OK) ? result1 : AAudioConvert_androidToAAudioResult(status);
 }
 
 aaudio_result_t AAudioServiceStreamMMAP::stop() {
+    if (!isRunning()) {
+        return AAUDIO_OK;
+    }
     if (mMmapStream == nullptr) return AAUDIO_ERROR_NULL;
-
     aaudio_result_t result1 = AAudioServiceStreamBase::stop();
+    if (!mInService) {
+        stopClient(mClientHandle);
+    }
     aaudio_result_t status = mMmapStream->stop(mPortHandle);
     mFramesRead.reset32();
     return (result1 != AAUDIO_OK) ? result1 :  AAudioConvert_androidToAAudioResult(status);
@@ -266,6 +280,15 @@
     return AAudioServiceStreamBase::flush();;
 }
 
+aaudio_result_t AAudioServiceStreamMMAP::startClient(const android::AudioClient& client,
+                                                     audio_port_handle_t *clientHandle) {
+    return AAudioConvert_androidToAAudioResult(mMmapStream->start(client, clientHandle));
+}
+
+aaudio_result_t AAudioServiceStreamMMAP::stopClient(audio_port_handle_t clientHandle) {
+    return AAudioConvert_androidToAAudioResult(mMmapStream->stop(clientHandle));
+}
+
 aaudio_result_t AAudioServiceStreamMMAP::getFreeRunningPosition(int64_t *positionFrames,
                                                                 int64_t *timeNanos) {
     struct audio_mmap_position position;
@@ -301,11 +324,11 @@
 
 void AAudioServiceStreamMMAP::onRoutingChanged(audio_port_handle_t deviceId) {
     ALOGD("AAudioServiceStreamMMAP::onRoutingChanged() called with %d, old = %d",
-          deviceId, mPortHandle);
-    if (mPortHandle > 0 && mPortHandle != deviceId) {
+          deviceId, mDeviceId);
+    if (mDeviceId != AUDIO_PORT_HANDLE_NONE  && mDeviceId != deviceId) {
         disconnect();
     }
-    mPortHandle = deviceId;
+    mDeviceId = deviceId;
 };
 
 /**
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.h b/services/oboeservice/AAudioServiceStreamMMAP.h
index 257bea9..533e5a8 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.h
+++ b/services/oboeservice/AAudioServiceStreamMMAP.h
@@ -43,7 +43,7 @@
     , public android::MmapStreamCallback {
 
 public:
-    AAudioServiceStreamMMAP(uid_t serviceUid);
+    AAudioServiceStreamMMAP(const android::AudioClient& serviceClient, bool inService);
     virtual ~AAudioServiceStreamMMAP() = default;
 
     aaudio_result_t open(const aaudio::AAudioStreamRequest &request,
@@ -77,6 +77,11 @@
 
     aaudio_result_t close() override;
 
+    virtual aaudio_result_t startClient(const android::AudioClient& client,
+                                        audio_port_handle_t *clientHandle);
+
+    virtual aaudio_result_t stopClient(audio_port_handle_t clientHandle);
+
     /**
      * Send a MMAP/NOIRQ buffer timestamp to the client.
      */
@@ -131,9 +136,10 @@
     // Interface to the AudioFlinger MMAP support.
     android::sp<android::MmapStreamInterface> mMmapStream;
     struct audio_mmap_buffer_info             mMmapBufferinfo;
-    android::MmapStreamInterface::Client      mMmapClient;
-    audio_port_handle_t                       mPortHandle = -1; // TODO review best default
-    uid_t                                     mCachedUserId = -1;
+    audio_port_handle_t                       mPortHandle = AUDIO_PORT_HANDLE_NONE;
+    audio_port_handle_t                       mDeviceId = AUDIO_PORT_HANDLE_NONE;
+    android::AudioClient                      mServiceClient;
+    bool                                      mInService = false;
 };
 
 } // namespace aaudio
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index 8bb34d1..fe488cb 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudioService"
+#define LOG_TAG "AAudioServiceStreamShared"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -191,6 +191,9 @@
  * An AAUDIO_SERVICE_EVENT_STARTED will be sent to the client when complete.
  */
 aaudio_result_t AAudioServiceStreamShared::start()  {
+    if (isRunning()) {
+        return AAUDIO_OK;
+    }
     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
@@ -201,7 +204,10 @@
         ALOGE("AAudioServiceStreamShared::start() mServiceEndpoint returned %d", result);
         disconnect();
     } else {
-        result = AAudioServiceStreamBase::start();
+        result = endpoint->getStreamInternal()->startClient(mMmapClient, &mClientHandle);
+        if (result == AAUDIO_OK) {
+            result = AAudioServiceStreamBase::start();
+        }
     }
     return result;
 }
@@ -212,10 +218,14 @@
  * An AAUDIO_SERVICE_EVENT_PAUSED will be sent to the client when complete.
 */
 aaudio_result_t AAudioServiceStreamShared::pause()  {
+    if (!isRunning()) {
+        return AAUDIO_OK;
+    }
     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
+    endpoint->getStreamInternal()->stopClient(mClientHandle);
     aaudio_result_t result = endpoint->stopStream(this);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::pause() mServiceEndpoint returned %d", result);
@@ -225,10 +235,14 @@
 }
 
 aaudio_result_t AAudioServiceStreamShared::stop()  {
+    if (!isRunning()) {
+        return AAUDIO_OK;
+    }
     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
+    endpoint->getStreamInternal()->stopClient(mClientHandle);
     aaudio_result_t result = endpoint->stopStream(this);
     if (result != AAUDIO_OK) {
         ALOGE("AAudioServiceStreamShared::stop() mServiceEndpoint returned %d", result);
@@ -248,7 +262,7 @@
         return AAUDIO_ERROR_INVALID_STATE;
     }
     if (mState != AAUDIO_STREAM_STATE_PAUSED) {
-        ALOGE("AAudioServiceStreamShared::flush() stream not paused, state = %s",
+         ALOGE("AAudioServiceStreamShared::flush() stream not paused, state = %s",
             AAudio_convertStreamStateToText(mState));
         return AAUDIO_ERROR_INVALID_STATE;
     }
@@ -261,11 +275,12 @@
         return AAUDIO_OK;
     }
 
+    stop();
+
     AAudioServiceEndpoint *endpoint = mServiceEndpoint;
     if (endpoint == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
-    endpoint->stopStream(this);
 
     endpoint->unregisterStream(this);
 
@@ -277,7 +292,6 @@
         delete mAudioDataQueue;
         mAudioDataQueue = nullptr;
     }
-
     return AAudioServiceStreamBase::close();
 }
 
@@ -293,9 +307,6 @@
     return AAUDIO_OK;
 }
 
-void AAudioServiceStreamShared::onStop() {
-}
-
 void AAudioServiceStreamShared::markTransferTime(int64_t nanoseconds) {
     mMarkedPosition = mAudioDataQueue->getFifoBuffer()->getReadCounter();
     mMarkedTime = nanoseconds;
diff --git a/services/oboeservice/AAudioServiceStreamShared.h b/services/oboeservice/AAudioServiceStreamShared.h
index 742c5af..6b67337 100644
--- a/services/oboeservice/AAudioServiceStreamShared.h
+++ b/services/oboeservice/AAudioServiceStreamShared.h
@@ -87,8 +87,6 @@
      */
     void markTransferTime(int64_t nanoseconds);
 
-    void onStop();
-
 protected:
 
     aaudio_result_t getDownDataDescription(AudioEndpointParcelable &parcelable) override;