diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 8d0a705..f7b6f64 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -29,6 +29,7 @@
     Tracks.cpp                  \
     Effects.cpp                 \
     AudioMixer.cpp.arm          \
+    PatchPanel.cpp
 
 LOCAL_SRC_FILES += StateQueue.cpp
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index e174fc9..457ac3d 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -177,6 +177,7 @@
     if (doLog) {
         mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY);
     }
+
 #ifdef TEE_SINK
     (void) property_get("ro.debuggable", value, "0");
     int debuggable = atoi(value);
@@ -218,6 +219,8 @@
         }
     }
 
+    mPatchPanel = new PatchPanel(this);
+
     mMode = AUDIO_MODE_NORMAL;
 }
 
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 3d7a562..29dc6b2 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -225,41 +225,24 @@
 
     /* List available audio ports and their attributes */
     virtual status_t listAudioPorts(unsigned int *num_ports,
-                                    struct audio_port *ports)
-    {
-        return INVALID_OPERATION;
-    }
+                                    struct audio_port *ports);
 
     /* Get attributes for a given audio port */
-    virtual status_t getAudioPort(struct audio_port *port)
-    {
-        return INVALID_OPERATION;
-    }
+    virtual status_t getAudioPort(struct audio_port *port);
 
     /* Create an audio patch between several source and sink ports */
     virtual status_t createAudioPatch(const struct audio_patch *patch,
-                                       audio_patch_handle_t *handle)
-    {
-        return INVALID_OPERATION;
-    }
+                                       audio_patch_handle_t *handle);
 
     /* Release an audio patch */
-    virtual status_t releaseAudioPatch(audio_patch_handle_t handle)
-    {
-        return INVALID_OPERATION;
-    }
+    virtual status_t releaseAudioPatch(audio_patch_handle_t handle);
 
     /* List existing audio patches */
     virtual status_t listAudioPatches(unsigned int *num_patches,
-                                      struct audio_patch *patches)
-    {
-        return INVALID_OPERATION;
-    }
+                                      struct audio_patch *patches);
+
     /* Set audio port configuration */
-    virtual status_t setAudioPortConfig(const struct audio_port_config *config)
-    {
-        return INVALID_OPERATION;
-    }
+    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
 
     virtual     status_t    onTransact(
                                 uint32_t code,
@@ -435,6 +418,8 @@
 
 #include "Effects.h"
 
+#include "PatchPanel.h"
+
     // server side of the client's IAudioTrack
     class TrackHandle : public android::BnAudioTrack {
     public:
@@ -542,6 +527,8 @@
 
         const char *moduleName() const { return mModuleName; }
         audio_hw_device_t *hwDevice() const { return mHwDevice; }
+        uint32_t version() const { return mHwDevice->common.version; }
+
     private:
         const char * const mModuleName;
         audio_hw_device_t * const mHwDevice;
@@ -702,6 +689,8 @@
     bool    mIsLowRamDevice;
     bool    mIsDeviceTypeKnown;
     nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
+
+    sp<PatchPanel> mPatchPanel;
 };
 
 #undef INCLUDING_FROM_AUDIOFLINGER_H
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
new file mode 100644
index 0000000..eee74b3
--- /dev/null
+++ b/services/audioflinger/PatchPanel.cpp
@@ -0,0 +1,409 @@
+/*
+**
+** Copyright 2014, 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 "AudioFlinger::PatchPanel"
+//#define LOG_NDEBUG 0
+
+#include "Configuration.h"
+#include <utils/Log.h>
+#include <audio_utils/primitives.h>
+
+#include "AudioFlinger.h"
+#include "ServiceUtilities.h"
+#include <media/AudioParameter.h>
+
+// ----------------------------------------------------------------------------
+
+// Note: the following macro is used for extremely verbose logging message.  In
+// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
+// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
+// are so verbose that we want to suppress them even when we have ALOG_ASSERT
+// turned on.  Do not uncomment the #def below unless you really know what you
+// are doing and want to see all of the extremely verbose messages.
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+namespace android {
+
+/* List connected audio ports and their attributes */
+status_t AudioFlinger::listAudioPorts(unsigned int *num_ports,
+                                struct audio_port *ports)
+{
+    Mutex::Autolock _l(mLock);
+    if (mPatchPanel != 0) {
+        return mPatchPanel->listAudioPorts(num_ports, ports);
+    }
+    return NO_INIT;
+}
+
+/* Get supported attributes for a given audio port */
+status_t AudioFlinger::getAudioPort(struct audio_port *port)
+{
+    Mutex::Autolock _l(mLock);
+    if (mPatchPanel != 0) {
+        return mPatchPanel->getAudioPort(port);
+    }
+    return NO_INIT;
+}
+
+
+/* Connect a patch between several source and sink ports */
+status_t AudioFlinger::createAudioPatch(const struct audio_patch *patch,
+                                   audio_patch_handle_t *handle)
+{
+    Mutex::Autolock _l(mLock);
+    if (mPatchPanel != 0) {
+        return mPatchPanel->createAudioPatch(patch, handle);
+    }
+    return NO_INIT;
+}
+
+/* Disconnect a patch */
+status_t AudioFlinger::releaseAudioPatch(audio_patch_handle_t handle)
+{
+    Mutex::Autolock _l(mLock);
+    if (mPatchPanel != 0) {
+        return mPatchPanel->releaseAudioPatch(handle);
+    }
+    return NO_INIT;
+}
+
+
+/* List connected audio ports and they attributes */
+status_t AudioFlinger::listAudioPatches(unsigned int *num_patches,
+                                  struct audio_patch *patches)
+{
+    Mutex::Autolock _l(mLock);
+    if (mPatchPanel != 0) {
+        return mPatchPanel->listAudioPatches(num_patches, patches);
+    }
+    return NO_INIT;
+}
+
+/* Set audio port configuration */
+status_t AudioFlinger::setAudioPortConfig(const struct audio_port_config *config)
+{
+    Mutex::Autolock _l(mLock);
+    if (mPatchPanel != 0) {
+        return mPatchPanel->setAudioPortConfig(config);
+    }
+    return NO_INIT;
+}
+
+
+AudioFlinger::PatchPanel::PatchPanel(const sp<AudioFlinger>& audioFlinger)
+                                   : mAudioFlinger(audioFlinger)
+{
+}
+
+AudioFlinger::PatchPanel::~PatchPanel()
+{
+}
+
+/* List connected audio ports and their attributes */
+status_t AudioFlinger::PatchPanel::listAudioPorts(unsigned int *num_ports __unused,
+                                struct audio_port *ports __unused)
+{
+    ALOGV("listAudioPorts");
+    return NO_ERROR;
+}
+
+/* Get supported attributes for a given audio port */
+status_t AudioFlinger::PatchPanel::getAudioPort(struct audio_port *port __unused)
+{
+    ALOGV("getAudioPort");
+    return NO_ERROR;
+}
+
+
+/* Connect a patch between several source and sink ports */
+status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *patch,
+                                   audio_patch_handle_t *handle)
+{
+    ALOGV("createAudioPatch() num_sources %d num_sinks %d handle %d",
+          patch->num_sources, patch->num_sinks, *handle);
+    status_t status = NO_ERROR;
+
+    audio_patch_handle_t halHandle = AUDIO_PATCH_HANDLE_NONE;
+
+    sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
+    if (audioflinger == 0) {
+        return NO_INIT;
+    }
+    if (handle == NULL || patch == NULL) {
+        return BAD_VALUE;
+    }
+    // limit number of sources to 1 for now
+    if (patch->num_sources == 0 || patch->num_sources > 1 ||
+            patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
+        return BAD_VALUE;
+    }
+
+    for (size_t index = 0; *handle != 0 && index < mPatches.size(); index++) {
+        if (*handle == mPatches[index]->mHandle) {
+            ALOGV("createAudioPatch() removing patch handle %d", *handle);
+            halHandle = mPatches[index]->mHalHandle;
+            mPatches.removeAt(index);
+            break;
+        }
+    }
+
+    switch (patch->sources[0].type) {
+        case AUDIO_PORT_TYPE_DEVICE: {
+            // limit number of sinks to 1 for now
+            if (patch->num_sinks > 1) {
+                return BAD_VALUE;
+            }
+            audio_module_handle_t src_module = patch->sources[0].ext.device.hw_module;
+            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(src_module);
+            if (index < 0) {
+                ALOGW("createAudioPatch() bad src hw module %d", src_module);
+                return BAD_VALUE;
+            }
+            for (unsigned int i = 0; i < patch->num_sinks; i++) {
+                // limit to connections between devices and output streams
+                if (patch->sinks[i].type != AUDIO_PORT_TYPE_MIX) {
+                    ALOGW("createAudioPatch() invalid sink type %d for device source",
+                          patch->sinks[i].type);
+                    return BAD_VALUE;
+                }
+                // limit to connections between sinks and sources on same HW module
+                if (patch->sinks[i].ext.mix.hw_module != src_module) {
+                    ALOGW("createAudioPatch() cannot connect source on module %d to"
+                            "sink on module %d", src_module, patch->sinks[i].ext.mix.hw_module);
+                    return BAD_VALUE;
+                }
+            }
+
+            AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
+            if (audioHwDevice->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+                if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+                    sp<ThreadBase> thread = audioflinger->checkRecordThread_l(
+                                                                    patch->sinks[0].ext.mix.handle);
+                    if (thread == 0) {
+                        ALOGW("createAudioPatch() bad capture I/O handle %d",
+                                                                  patch->sinks[0].ext.mix.handle);
+                        return BAD_VALUE;
+                    }
+                    status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
+                } else {
+                    audio_hw_device_t *hwDevice = audioHwDevice->hwDevice();
+                    status = hwDevice->create_audio_patch(hwDevice,
+                                                           patch->num_sources,
+                                                           patch->sources,
+                                                           patch->num_sinks,
+                                                           patch->sinks,
+                                                           &halHandle);
+                }
+            } else {
+                sp<ThreadBase> thread = audioflinger->checkRecordThread_l(
+                                                                    patch->sinks[0].ext.mix.handle);
+                if (thread == 0) {
+                    ALOGW("createAudioPatch() bad capture I/O handle %d",
+                                                                  patch->sinks[0].ext.mix.handle);
+                    return BAD_VALUE;
+                }
+                AudioParameter param;
+                param.addInt(String8(AudioParameter::keyRouting),
+                             (int)patch->sources[0].ext.device.type);
+                param.addInt(String8(AudioParameter::keyInputSource),
+                                                     (int)patch->sinks[0].ext.mix.usecase.source);
+
+                ALOGW("createAudioPatch() AUDIO_PORT_TYPE_DEVICE setParameters %s",
+                                                                      param.toString().string());
+                status = thread->setParameters(param.toString());
+            }
+        } break;
+        case AUDIO_PORT_TYPE_MIX: {
+            audio_module_handle_t src_module =  patch->sources[0].ext.mix.hw_module;
+            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(src_module);
+            if (index < 0) {
+                ALOGW("createAudioPatch() bad src hw module %d", src_module);
+                return BAD_VALUE;
+            }
+            // limit to connections between devices and output streams
+            for (unsigned int i = 0; i < patch->num_sinks; i++) {
+                if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
+                    ALOGW("createAudioPatch() invalid sink type %d for bus source",
+                          patch->sinks[i].type);
+                    return BAD_VALUE;
+                }
+                // limit to connections between sinks and sources on same HW module
+                if (patch->sinks[i].ext.device.hw_module != src_module) {
+                    return BAD_VALUE;
+                }
+            }
+            AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
+            sp<ThreadBase> thread =
+                            audioflinger->checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
+            if (thread == 0) {
+                ALOGW("createAudioPatch() bad playback I/O handle %d",
+                          patch->sources[0].ext.mix.handle);
+                return BAD_VALUE;
+            }
+            if (audioHwDevice->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+                status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
+            } else {
+                audio_devices_t type = AUDIO_DEVICE_NONE;
+                for (unsigned int i = 0; i < patch->num_sinks; i++) {
+                    type |= patch->sinks[i].ext.device.type;
+                }
+                AudioParameter param;
+                param.addInt(String8(AudioParameter::keyRouting), (int)type);
+                status = thread->setParameters(param.toString());
+            }
+
+        } break;
+        default:
+            return BAD_VALUE;
+    }
+    ALOGV("createAudioPatch() status %d", status);
+    if (status == NO_ERROR) {
+        *handle = audioflinger->nextUniqueId();
+        Patch *newPatch = new Patch(patch);
+        newPatch->mHandle = *handle;
+        newPatch->mHalHandle = halHandle;
+        mPatches.add(newPatch);
+        ALOGV("createAudioPatch() added new patch handle %d halHandle %d", *handle, halHandle);
+    }
+    return status;
+}
+
+/* Disconnect a patch */
+status_t AudioFlinger::PatchPanel::releaseAudioPatch(audio_patch_handle_t handle)
+{
+    ALOGV("releaseAudioPatch handle %d", handle);
+    status_t status = NO_ERROR;
+    size_t index;
+
+    sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
+    if (audioflinger == 0) {
+        return NO_INIT;
+    }
+
+    for (index = 0; index < mPatches.size(); index++) {
+        if (handle == mPatches[index]->mHandle) {
+            break;
+        }
+    }
+    if (index == mPatches.size()) {
+        return BAD_VALUE;
+    }
+
+    struct audio_patch *patch = &mPatches[index]->mAudioPatch;
+
+    switch (patch->sources[0].type) {
+        case AUDIO_PORT_TYPE_DEVICE: {
+            audio_module_handle_t src_module = patch->sources[0].ext.device.hw_module;
+            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(src_module);
+            if (index < 0) {
+                ALOGW("releaseAudioPatch() bad src hw module %d", src_module);
+                status = BAD_VALUE;
+                break;
+            }
+            AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
+            if (audioHwDevice->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+                if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+                    sp<ThreadBase> thread = audioflinger->checkRecordThread_l(
+                                                                    patch->sinks[0].ext.mix.handle);
+                    if (thread == 0) {
+                        ALOGW("createAudioPatch() bad capture I/O handle %d",
+                                                                  patch->sinks[0].ext.mix.handle);
+                        status = BAD_VALUE;
+                        break;
+                    }
+                    status = thread->sendReleaseAudioPatchConfigEvent(mPatches[index]->mHalHandle);
+                } else {
+                    audio_hw_device_t *hwDevice = audioHwDevice->hwDevice();
+                    status = hwDevice->release_audio_patch(hwDevice, mPatches[index]->mHalHandle);
+                }
+            } else {
+                sp<ThreadBase> thread = audioflinger->checkRecordThread_l(
+                                                                    patch->sinks[0].ext.mix.handle);
+                if (thread == 0) {
+                    ALOGW("releaseAudioPatch() bad capture I/O handle %d",
+                                                                  patch->sinks[0].ext.mix.handle);
+                    status = BAD_VALUE;
+                    break;
+                }
+                AudioParameter param;
+                param.addInt(String8(AudioParameter::keyRouting), 0);
+                ALOGW("releaseAudioPatch() AUDIO_PORT_TYPE_DEVICE setParameters %s",
+                                                                      param.toString().string());
+                status = thread->setParameters(param.toString());
+            }
+        } break;
+        case AUDIO_PORT_TYPE_MIX: {
+            audio_module_handle_t src_module =  patch->sources[0].ext.mix.hw_module;
+            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(src_module);
+            if (index < 0) {
+                ALOGW("releaseAudioPatch() bad src hw module %d", src_module);
+                status = BAD_VALUE;
+                break;
+            }
+            sp<ThreadBase> thread =
+                            audioflinger->checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
+            if (thread == 0) {
+                ALOGW("releaseAudioPatch() bad playback I/O handle %d",
+                                                              patch->sources[0].ext.mix.handle);
+                status = BAD_VALUE;
+                break;
+            }
+            AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
+            if (audioHwDevice->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+                status = thread->sendReleaseAudioPatchConfigEvent(mPatches[index]->mHalHandle);
+            } else {
+                AudioParameter param;
+                param.addInt(String8(AudioParameter::keyRouting), (int)0);
+                status = thread->setParameters(param.toString());
+            }
+        } break;
+        default:
+            status = BAD_VALUE;
+            break;
+    }
+
+    delete (mPatches[index]);
+    mPatches.removeAt(index);
+    return status;
+}
+
+
+/* List connected audio ports and they attributes */
+status_t AudioFlinger::PatchPanel::listAudioPatches(unsigned int *num_patches __unused,
+                                  struct audio_patch *patches __unused)
+{
+    ALOGV("listAudioPatches");
+    return NO_ERROR;
+}
+
+/* Set audio port configuration */
+status_t AudioFlinger::PatchPanel::setAudioPortConfig(
+        const struct audio_port_config *config __unused)
+{
+    ALOGV("setAudioPortConfig");
+    return NO_ERROR;
+}
+
+
+
+}; // namespace android
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
new file mode 100644
index 0000000..7f78621
--- /dev/null
+++ b/services/audioflinger/PatchPanel.h
@@ -0,0 +1,60 @@
+/*
+**
+** Copyright 2014, 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 INCLUDING_FROM_AUDIOFLINGER_H
+    #error This header file should only be included from AudioFlinger.h
+#endif
+
+class PatchPanel : public RefBase {
+public:
+    PatchPanel(const sp<AudioFlinger>& audioFlinger);
+    virtual ~PatchPanel();
+
+    /* List connected audio ports and their attributes */
+    status_t listAudioPorts(unsigned int *num_ports,
+                                    struct audio_port *ports);
+
+    /* Get supported attributes for a given audio port */
+    status_t getAudioPort(struct audio_port *port);
+
+    /* Create a patch between several source and sink ports */
+    status_t createAudioPatch(const struct audio_patch *patch,
+                                       audio_patch_handle_t *handle);
+
+    /* Release a patch */
+    status_t releaseAudioPatch(audio_patch_handle_t handle);
+
+    /* List connected audio devices and they attributes */
+    status_t listAudioPatches(unsigned int *num_patches,
+                                      struct audio_patch *patches);
+
+    /* Set audio port configuration */
+    status_t setAudioPortConfig(const struct audio_port_config *config);
+
+    class Patch {
+    public:
+        Patch(const struct audio_patch *patch) :
+            mAudioPatch(*patch), mHandle(0), mHalHandle(0) {}
+
+        struct audio_patch mAudioPatch;
+        audio_patch_handle_t mHandle;
+        audio_patch_handle_t mHalHandle;
+    };
+private:
+    const wp<AudioFlinger>  mAudioFlinger;
+    SortedVector <Patch *> mPatches;
+};
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index cdc9a19..4972c7a 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -401,6 +401,30 @@
     return sendConfigEvent_l(configEvent);
 }
 
+status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent(
+                                                        const struct audio_patch *patch,
+                                                        audio_patch_handle_t *handle)
+{
+    Mutex::Autolock _l(mLock);
+    sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle);
+    status_t status = sendConfigEvent_l(configEvent);
+    if (status == NO_ERROR) {
+        CreateAudioPatchConfigEventData *data =
+                                        (CreateAudioPatchConfigEventData *)configEvent->mData.get();
+        *handle = data->mHandle;
+    }
+    return status;
+}
+
+status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent(
+                                                                const audio_patch_handle_t handle)
+{
+    Mutex::Autolock _l(mLock);
+    sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle);
+    return sendConfigEvent_l(configEvent);
+}
+
+
 // post condition: mConfigEvents.isEmpty()
 void AudioFlinger::ThreadBase::processConfigEvents_l()
 {
@@ -431,6 +455,16 @@
                 configChanged = true;
             }
         } break;
+        case CFG_EVENT_CREATE_AUDIO_PATCH: {
+            CreateAudioPatchConfigEventData *data =
+                                            (CreateAudioPatchConfigEventData *)event->mData.get();
+            event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
+        } break;
+        case CFG_EVENT_RELEASE_AUDIO_PATCH: {
+            ReleaseAudioPatchConfigEventData *data =
+                                            (ReleaseAudioPatchConfigEventData *)event->mData.get();
+            event->mStatus = releaseAudioPatch_l(data->mHandle);
+        } break;
         default:
             ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
             break;
@@ -2594,6 +2628,47 @@
     }
     return INVALID_OPERATION;
 }
+
+status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
+                                                          audio_patch_handle_t *handle)
+{
+    status_t status = NO_ERROR;
+    if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+        // store new device and send to effects
+        audio_devices_t type = AUDIO_DEVICE_NONE;
+        for (unsigned int i = 0; i < patch->num_sinks; i++) {
+            type |= patch->sinks[i].ext.device.type;
+        }
+        mOutDevice = type;
+        for (size_t i = 0; i < mEffectChains.size(); i++) {
+            mEffectChains[i]->setDevice_l(mOutDevice);
+        }
+
+        audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
+        status = hwDevice->create_audio_patch(hwDevice,
+                                               patch->num_sources,
+                                               patch->sources,
+                                               patch->num_sinks,
+                                               patch->sinks,
+                                               handle);
+    } else {
+        ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL");
+    }
+    return status;
+}
+
+status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+{
+    status_t status = NO_ERROR;
+    if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+        audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
+        status = hwDevice->release_audio_patch(hwDevice, handle);
+    } else {
+        ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL");
+    }
+    return status;
+}
+
 // ----------------------------------------------------------------------------
 
 AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
@@ -5743,4 +5818,61 @@
     return 0;
 }
 
+status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch,
+                                                          audio_patch_handle_t *handle)
+{
+    status_t status = NO_ERROR;
+    if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+        // store new device and send to effects
+        mInDevice = patch->sources[0].ext.device.type;
+        for (size_t i = 0; i < mEffectChains.size(); i++) {
+            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());
+            }
+        }
+
+        // store new source and send to effects
+        if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
+            mAudioSource = patch->sinks[0].ext.mix.usecase.source;
+            for (size_t i = 0; i < mEffectChains.size(); i++) {
+                mEffectChains[i]->setAudioSource_l(mAudioSource);
+            }
+        }
+
+        audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice();
+        status = hwDevice->create_audio_patch(hwDevice,
+                                               patch->num_sources,
+                                               patch->sources,
+                                               patch->num_sinks,
+                                               patch->sinks,
+                                               handle);
+    } else {
+        ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL");
+    }
+    return status;
+}
+
+status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+{
+    status_t status = NO_ERROR;
+    if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+        audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice();
+        status = hwDevice->release_audio_patch(hwDevice, handle);
+    } else {
+        ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL");
+    }
+    return status;
+}
+
+
 }; // namespace android
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index cc2b246..f8037c6 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -48,6 +48,8 @@
         CFG_EVENT_IO,
         CFG_EVENT_PRIO,
         CFG_EVENT_SET_PARAMETER,
+        CFG_EVENT_CREATE_AUDIO_PATCH,
+        CFG_EVENT_RELEASE_AUDIO_PATCH,
     };
 
     class ConfigEventData: public RefBase {
@@ -161,6 +163,52 @@
         virtual ~SetParameterConfigEvent() {}
     };
 
+    class CreateAudioPatchConfigEventData : public ConfigEventData {
+    public:
+        CreateAudioPatchConfigEventData(const struct audio_patch patch,
+                                        audio_patch_handle_t handle) :
+            mPatch(patch), mHandle(handle) {}
+
+        virtual  void dump(char *buffer, size_t size) {
+            snprintf(buffer, size, "Patch handle: %u\n", mHandle);
+        }
+
+        const struct audio_patch mPatch;
+        audio_patch_handle_t mHandle;
+    };
+
+    class CreateAudioPatchConfigEvent : public ConfigEvent {
+    public:
+        CreateAudioPatchConfigEvent(const struct audio_patch patch,
+                                    audio_patch_handle_t handle) :
+            ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) {
+            mData = new CreateAudioPatchConfigEventData(patch, handle);
+            mWaitStatus = true;
+        }
+        virtual ~CreateAudioPatchConfigEvent() {}
+    };
+
+    class ReleaseAudioPatchConfigEventData : public ConfigEventData {
+    public:
+        ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) :
+            mHandle(handle) {}
+
+        virtual  void dump(char *buffer, size_t size) {
+            snprintf(buffer, size, "Patch handle: %u\n", mHandle);
+        }
+
+        audio_patch_handle_t mHandle;
+    };
+
+    class ReleaseAudioPatchConfigEvent : public ConfigEvent {
+    public:
+        ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) :
+            ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) {
+            mData = new ReleaseAudioPatchConfigEventData(handle);
+            mWaitStatus = true;
+        }
+        virtual ~ReleaseAudioPatchConfigEvent() {}
+    };
 
     class PMDeathRecipient : public IBinder::DeathRecipient {
     public:
@@ -209,8 +257,15 @@
                 void        sendIoConfigEvent_l(int event, int param = 0);
                 void        sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
                 status_t    sendSetParameterConfigEvent_l(const String8& keyValuePair);
+                status_t    sendCreateAudioPatchConfigEvent(const struct audio_patch *patch,
+                                                            audio_patch_handle_t *handle);
+                status_t    sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle);
                 void        processConfigEvents_l();
     virtual     void        cacheParameters_l() = 0;
+    virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
+                                               audio_patch_handle_t *handle) = 0;
+    virtual     status_t    releaseAudioPatch_l(const audio_patch_handle_t handle) = 0;
+
 
                 // see note at declaration of mStandby, mOutDevice and mInDevice
                 bool        standby() const { return mStandby; }
@@ -641,6 +696,10 @@
 
     virtual     uint32_t    correctLatency_l(uint32_t latency) const;
 
+    virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
+                                   audio_patch_handle_t *handle);
+    virtual     status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);
+
 private:
 
     friend class AudioFlinger;      // for numerous
@@ -1030,6 +1089,9 @@
     virtual void        cacheParameters_l() {}
     virtual String8     getParameters(const String8& keys);
     virtual void        audioConfigChanged(int event, int param = 0);
+    virtual status_t    createAudioPatch_l(const struct audio_patch *patch,
+                                           audio_patch_handle_t *handle);
+    virtual status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);
             void        readInputParameters_l();
     virtual uint32_t    getInputFramesLost();
 
diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/AudioPolicyClientImpl.cpp
index 44c47c3..8225e36 100644
--- a/services/audiopolicy/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/AudioPolicyClientImpl.cpp
@@ -182,6 +182,17 @@
     return af->moveEffects(session, src_output, dst_output);
 }
 
+status_t AudioPolicyService::AudioPolicyClient::createAudioPatch(const struct audio_patch *patch,
+                                                                  audio_patch_handle_t *handle,
+                                                                  int delayMs)
+{
+    return mAudioPolicyService->clientCreateAudioPatch(patch, handle, delayMs);
+}
 
+status_t AudioPolicyService::AudioPolicyClient::releaseAudioPatch(audio_patch_handle_t handle,
+                                                                  int delayMs)
+{
+    return mAudioPolicyService->clientReleaseAudioPatch(handle, delayMs);
+}
 
 }; // namespace android
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 66260e3..bb2deb6 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -246,6 +246,15 @@
                                      audio_io_handle_t srcOutput,
                                      audio_io_handle_t dstOutput) = 0;
 
+    /* Create a patch between several source and sink ports */
+    virtual status_t createAudioPatch(const struct audio_patch *patch,
+                                       audio_patch_handle_t *handle,
+                                       int delayMs) = 0;
+
+    /* Release a patch */
+    virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
+                                       int delayMs) = 0;
+
 };
 
 extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface);
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index bd9b15a..b047e1d 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -267,7 +267,7 @@
             // also force a device 0 for the two outputs it is duplicated to which may override
             // a valid device selection on those outputs.
             setOutputDevice(mOutputs.keyAt(i),
-                            getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
+                            getNewOutputDevice(mOutputs.keyAt(i), true /*fromCache*/),
                             !mOutputs.valueAt(i)->isDuplicated(),
                             0);
         }
@@ -419,7 +419,7 @@
     }
 
     // check for device and output changes triggered by new phone state
-    newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+    newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
     checkA2dpSuspend();
     checkOutputForAllStrategies();
     updateDevicesAndOutputs();
@@ -544,7 +544,7 @@
     updateDevicesAndOutputs();
     for (size_t i = 0; i < mOutputs.size(); i++) {
         audio_io_handle_t output = mOutputs.keyAt(i);
-        audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
+        audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/);
         setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
         if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
             applyStreamVolumes(output, newDevice, 0, true);
@@ -553,16 +553,7 @@
 
     audio_io_handle_t activeInput = getActiveInput();
     if (activeInput != 0) {
-        AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
-        audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
-        if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
-            ALOGV("setForceUse() changing device from %x to %x for input %d",
-                    inputDesc->mDevice, newDevice, activeInput);
-            inputDesc->mDevice = newDevice;
-            AudioParameter param = AudioParameter();
-            param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
-            mpClientInterface->setParameters(activeInput, param.toString());
-        }
+        setInputDevice(activeInput, getNewInputDevice(activeInput));
     }
 
 }
@@ -579,7 +570,7 @@
 
 // Find a direct output profile compatible with the parameters passed, even if the input flags do
 // not explicitly request a direct output
-AudioPolicyManager::IOProfile *AudioPolicyManager::getProfileForDirectOutput(
+sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getProfileForDirectOutput(
                                                                audio_devices_t device,
                                                                uint32_t samplingRate,
                                                                audio_format_t format,
@@ -591,7 +582,7 @@
             continue;
         }
         for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
-            IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+            sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
             bool found = false;
             if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
                 if (profile->isCompatibleProfile(device, samplingRate, format,
@@ -676,7 +667,7 @@
     // FIXME: We should check the audio session here but we do not have it in this context.
     // This may prevent offloading in rare situations where effects are left active by apps
     // in the background.
-    IOProfile *profile = NULL;
+    sp<IOProfile> profile;
     if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
             !isNonOffloadableEffectEnabled()) {
         profile = getProfileForDirectOutput(device,
@@ -686,7 +677,7 @@
                                            (audio_output_flags_t)flags);
     }
 
-    if (profile != NULL) {
+    if (profile != 0) {
         AudioOutputDescriptor *outputDesc = NULL;
 
         for (size_t i = 0; i < mOutputs.size(); i++) {
@@ -705,7 +696,7 @@
         }
         // close direct output if currently open and configured with different parameters
         if (outputDesc != NULL) {
-            closeOutput(outputDesc->mId);
+            closeOutput(outputDesc->mIoHandle);
         }
         outputDesc = new AudioOutputDescriptor(profile);
         outputDesc->mDevice = device;
@@ -837,7 +828,7 @@
     outputDesc->changeRefCount(stream, 1);
 
     if (outputDesc->mRefCount[stream] == 1) {
-        audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+        audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
         routing_strategy strategy = getStrategy(stream);
         bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
                             (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
@@ -910,7 +901,7 @@
         // store time at which the stream was stopped - see isStreamActive()
         if (outputDesc->mRefCount[stream] == 0) {
             outputDesc->mStopTime[stream] = systemTime();
-            audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+            audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
             // delay the device switch by twice the latency because stopOutput() is executed when
             // the track stop() command is received and at that time the audio track buffer can
             // still contain data that needs to be drained. The latency only covers the audio HAL
@@ -928,7 +919,7 @@
                         outputDesc->sharesHwModuleWith(desc) &&
                         (newDevice != desc->device())) {
                     setOutputDevice(curOutput,
-                                    getNewDevice(curOutput, false /*fromCache*/),
+                                    getNewOutputDevice(curOutput, false /*fromCache*/),
                                     true,
                                     outputDesc->mLatency*2);
                 }
@@ -1018,11 +1009,11 @@
         break;
     }
 
-    IOProfile *profile = getInputProfile(device,
+    sp<IOProfile> profile = getInputProfile(device,
                                          samplingRate,
                                          format,
                                          channelMask);
-    if (profile == NULL) {
+    if (profile == 0) {
         ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d, "
                 "channelMask %04x",
                 device, samplingRate, format, channelMask);
@@ -1095,10 +1086,7 @@
         }
     }
 
-    audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
-    if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
-        inputDesc->mDevice = newDevice;
-    }
+    setInputDevice(input, getNewInputDevice(input), true /* force */);
 
     // automatically enable the remote submix output when input is started
     if (audio_is_remote_submix_device(inputDesc->mDevice)) {
@@ -1106,17 +1094,8 @@
                 AUDIO_POLICY_DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
     }
 
-    AudioParameter param = AudioParameter();
-    param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
-
-    int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ?
-                                        AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource;
-
-    param.addInt(String8(AudioParameter::keyInputSource), aliasSource);
     ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
 
-    mpClientInterface->setParameters(input, param.toString());
-
     inputDesc->mRefCount = 1;
     return NO_ERROR;
 }
@@ -1141,9 +1120,7 @@
                     AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
         }
 
-        AudioParameter param = AudioParameter();
-        param.addInt(String8(AudioParameter::keyRouting), 0);
-        mpClientInterface->setParameters(input, param.toString());
+        resetInputDevice(input);
         inputDesc->mRefCount = 0;
         return NO_ERROR;
     }
@@ -1608,13 +1585,13 @@
 
     // See if there is a profile to support this.
     // AUDIO_DEVICE_NONE
-    IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+    sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
                                             offloadInfo.sample_rate,
                                             offloadInfo.format,
                                             offloadInfo.channel_mask,
                                             AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
-    ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
-    return (profile != NULL);
+    ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
+    return (profile != 0);
 }
 
 // ----------------------------------------------------------------------------
@@ -1671,7 +1648,7 @@
         // This also validates mAvailableOutputDevices list
         for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
         {
-            const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j];
+            const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
 
             if (outProfile->mSupportedDevices.isEmpty()) {
                 ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName);
@@ -1683,7 +1660,7 @@
                     ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
                 AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
 
-                outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice->mType & profileTypes);
+                outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice->mDeviceType & profileTypes);
                 audio_io_handle_t output = mpClientInterface->openOutput(
                                                 outProfile->mModule->mHandle,
                                                 &outputDesc->mDevice,
@@ -1699,7 +1676,7 @@
                     delete outputDesc;
                 } else {
                     for (size_t k = 0; k  < outProfile->mSupportedDevices.size(); k++) {
-                        audio_devices_t type = outProfile->mSupportedDevices[k]->mType;
+                        audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType;
                         ssize_t index =
                                 mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
                         // give a valid ID to an attached device once confirmed it is reachable
@@ -1722,7 +1699,7 @@
         // mAvailableInputDevices list
         for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
         {
-            const IOProfile *inProfile = mHwModules[i]->mInputProfiles[j];
+            const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
 
             if (inProfile->mSupportedDevices.isEmpty()) {
                 ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName);
@@ -1734,7 +1711,7 @@
                 AudioInputDescriptor *inputDesc = new AudioInputDescriptor(inProfile);
 
                 inputDesc->mInputSource = AUDIO_SOURCE_MIC;
-                inputDesc->mDevice = inProfile->mSupportedDevices[0]->mType;
+                inputDesc->mDevice = inProfile->mSupportedDevices[0]->mDeviceType;
                 audio_io_handle_t input = mpClientInterface->openInput(
                                                     inProfile->mModule->mHandle,
                                                     &inputDesc->mDevice,
@@ -1744,7 +1721,7 @@
 
                 if (input != 0) {
                     for (size_t k = 0; k  < inProfile->mSupportedDevices.size(); k++) {
-                        audio_devices_t type = inProfile->mSupportedDevices[k]->mType;
+                        audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType;
                         ssize_t index =
                                 mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
                         // give a valid ID to an attached device once confirmed it is reachable
@@ -1765,7 +1742,7 @@
     // make sure all attached devices have been allocated a unique ID
     for (size_t i = 0; i  < mAvailableOutputDevices.size();) {
         if (mAvailableOutputDevices[i]->mId == 0) {
-            ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mType);
+            ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
             mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
             continue;
         }
@@ -1773,7 +1750,7 @@
     }
     for (size_t i = 0; i  < mAvailableInputDevices.size();) {
         if (mAvailableInputDevices[i]->mId == 0) {
-            ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mType);
+            ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
             mAvailableInputDevices.remove(mAvailableInputDevices[i]);
             continue;
         }
@@ -1781,7 +1758,7 @@
     }
     // make sure default device is reachable
     if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
-        ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mType);
+        ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType);
     }
 
     ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
@@ -1990,16 +1967,18 @@
 
 // ---
 
-void AudioPolicyManager::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
+void AudioPolicyManager::addOutput(audio_io_handle_t output, AudioOutputDescriptor *outputDesc)
 {
-    outputDesc->mId = id;
-    mOutputs.add(id, outputDesc);
+    outputDesc->mIoHandle = output;
+    outputDesc->mId = nextUniqueId();
+    mOutputs.add(output, outputDesc);
 }
 
-void AudioPolicyManager::addInput(audio_io_handle_t id, AudioInputDescriptor *inputDesc)
+void AudioPolicyManager::addInput(audio_io_handle_t input, AudioInputDescriptor *inputDesc)
 {
-    inputDesc->mId = id;
-    mInputs.add(id, inputDesc);
+    inputDesc->mIoHandle = input;
+    inputDesc->mId = nextUniqueId();
+    mInputs.add(input, inputDesc);
 }
 
 String8 AudioPolicyManager::addressToParameter(audio_devices_t device, const String8 address)
@@ -2027,7 +2006,7 @@
             }
         }
         // then look for output profiles that can be routed to this device
-        SortedVector<IOProfile *> profiles;
+        SortedVector< sp<IOProfile> > profiles;
         for (size_t i = 0; i < mHwModules.size(); i++)
         {
             if (mHwModules[i]->mHandle == 0) {
@@ -2050,7 +2029,7 @@
         // open outputs for matching profiles if needed. Direct outputs are also opened to
         // query for dynamic parameters and will be closed later by setDeviceConnectionState()
         for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
-            IOProfile *profile = profiles[profile_index];
+            sp<IOProfile> profile = profiles[profile_index];
 
             // nothing to do if one output is already opened for this profile
             size_t j;
@@ -2096,7 +2075,7 @@
                               reply.string());
                     value = strpbrk((char *)reply.string(), "=");
                     if (value != NULL) {
-                        loadSamplingRates(value + 1, profile);
+                        profile->loadSamplingRates(value + 1);
                     }
                 }
                 if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
@@ -2106,7 +2085,7 @@
                               reply.string());
                     value = strpbrk((char *)reply.string(), "=");
                     if (value != NULL) {
-                        loadFormats(value + 1, profile);
+                        profile->loadFormats(value + 1);
                     }
                 }
                 if (profile->mChannelMasks[0] == 0) {
@@ -2116,7 +2095,7 @@
                               reply.string());
                     value = strpbrk((char *)reply.string(), "=");
                     if (value != NULL) {
-                        loadOutChannels(value + 1, profile);
+                        profile->loadOutChannels(value + 1);
                     }
                 }
                 if (((profile->mSamplingRates[0] == 0) &&
@@ -2211,7 +2190,7 @@
             }
             for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
             {
-                IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+                sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
                 if (profile->mSupportedDevices.types() & device) {
                     ALOGV("checkOutputsForDevice(): "
                             "clearing direct output profile %zu on module %zu", j, i);
@@ -2251,7 +2230,7 @@
         }
 
         // then look for input profiles that can be routed to this device
-        SortedVector<IOProfile *> profiles;
+        SortedVector< sp<IOProfile> > profiles;
         for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
         {
             if (mHwModules[module_idx]->mHandle == 0) {
@@ -2279,7 +2258,7 @@
         // query for dynamic parameters and will be closed later by setDeviceConnectionState()
         for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
 
-            IOProfile *profile = profiles[profile_index];
+            sp<IOProfile> profile = profiles[profile_index];
             // nothing to do if one input is already opened for this profile
             size_t input_index;
             for (input_index = 0; input_index < mInputs.size(); input_index++) {
@@ -2317,7 +2296,7 @@
                               reply.string());
                     value = strpbrk((char *)reply.string(), "=");
                     if (value != NULL) {
-                        loadSamplingRates(value + 1, profile);
+                        profile->loadSamplingRates(value + 1);
                     }
                 }
                 if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
@@ -2326,7 +2305,7 @@
                     ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
                     value = strpbrk((char *)reply.string(), "=");
                     if (value != NULL) {
-                        loadFormats(value + 1, profile);
+                        profile->loadFormats(value + 1);
                     }
                 }
                 if (profile->mChannelMasks[0] == 0) {
@@ -2336,7 +2315,7 @@
                               reply.string());
                     value = strpbrk((char *)reply.string(), "=");
                     if (value != NULL) {
-                        loadInChannels(value + 1, profile);
+                        profile->loadInChannels(value + 1);
                     }
                 }
                 if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
@@ -2386,7 +2365,7 @@
             for (size_t profile_index = 0;
                  profile_index < mHwModules[module_index]->mInputProfiles.size();
                  profile_index++) {
-                IOProfile *profile = mHwModules[module_index]->mInputProfiles[profile_index];
+                sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
                 if (profile->mSupportedDevices.types() & device) {
                     ALOGV("checkInputsForDevice(): clearing direct input profile %d on module %d",
                           profile_index, module_index);
@@ -2605,7 +2584,7 @@
     }
 }
 
-audio_devices_t AudioPolicyManager::getNewDevice(audio_io_handle_t output, bool fromCache)
+audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache)
 {
     audio_devices_t device = AUDIO_DEVICE_NONE;
 
@@ -2638,7 +2617,16 @@
         device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
     }
 
-    ALOGV("getNewDevice() selected device %x", device);
+    ALOGV("getNewOutputDevice() selected device %x", device);
+    return device;
+}
+
+audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
+{
+    AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
+    audio_devices_t device = getDeviceForInputSource(inputDesc->mInputSource);
+
+    ALOGV("getNewInputDevice() selected device %x", device);
     return device;
 }
 
@@ -2784,7 +2772,7 @@
             }
             device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE;
             if (device) break;
-            device = mDefaultOutputDevice->mType;
+            device = mDefaultOutputDevice->mDeviceType;
             if (device == AUDIO_DEVICE_NONE) {
                 ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
             }
@@ -2813,7 +2801,7 @@
             }
             device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
             if (device) break;
-            device = mDefaultOutputDevice->mType;
+            device = mDefaultOutputDevice->mDeviceType;
             if (device == AUDIO_DEVICE_NONE) {
                 ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
             }
@@ -2895,7 +2883,7 @@
         // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
         device |= device2;
         if (device) break;
-        device = mDefaultOutputDevice->mType;
+        device = mDefaultOutputDevice->mDeviceType;
         if (device == AUDIO_DEVICE_NONE) {
             ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
         }
@@ -2981,9 +2969,9 @@
         }
         for (size_t i = 0; i < NUM_STRATEGIES; i++) {
             if (outputDesc->isStrategyActive((routing_strategy)i)) {
-                setStrategyMute((routing_strategy)i, true, outputDesc->mId);
+                setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle);
                 // do tempMute unmute after twice the mute wait time
-                setStrategyMute((routing_strategy)i, false, outputDesc->mId,
+                setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle,
                                 muteWaitMs *2, device);
             }
         }
@@ -3009,8 +2997,8 @@
     uint32_t muteWaitMs;
 
     if (outputDesc->isDuplicated()) {
-        muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
-        muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
+        muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs);
+        muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs);
         return muteWaitMs;
     }
     // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
@@ -3042,9 +3030,34 @@
     }
 
     ALOGV("setOutputDevice() changing device");
+
     // do the routing
-    param.addInt(String8(AudioParameter::keyRouting), (int)device);
-    mpClientInterface->setParameters(output, param.toString(), delayMs);
+    if (device == AUDIO_DEVICE_NONE) {
+        resetOutputDevice(output, delayMs);
+    } else {
+        DeviceVector deviceList = mAvailableOutputDevices.getDevicesFromType(device);
+        if (!deviceList.isEmpty()) {
+            struct audio_patch patch;
+            outputDesc->toAudioPortConfig(&patch.sources[0]);
+            patch.num_sources = 1;
+            patch.num_sinks = 0;
+            for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
+                deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
+                patch.sinks[i].ext.device.hw_module = patch.sources[0].ext.mix.hw_module;
+                patch.num_sinks++;
+            }
+            audio_patch_handle_t patchHandle = outputDesc->mPatchHandle;
+            status_t status = mpClientInterface->createAudioPatch(&patch,
+                                                                  &patchHandle,
+                                                                  delayMs);
+            ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
+                    "num_sources %d num_sinks %d",
+                                       status, patchHandle, patch.num_sources, patch.num_sinks);
+            if (status == NO_ERROR) {
+                outputDesc->mPatchHandle = patchHandle;
+            }
+        }
+    }
 
     // update stream volumes according to new device
     applyStreamVolumes(output, device, delayMs);
@@ -3052,7 +3065,65 @@
     return muteWaitMs;
 }
 
-AudioPolicyManager::IOProfile *AudioPolicyManager::getInputProfile(audio_devices_t device,
+status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output,
+                                               int delayMs)
+{
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    if (outputDesc->mPatchHandle == 0) {
+        return INVALID_OPERATION;
+    }
+    status_t status = mpClientInterface->releaseAudioPatch(outputDesc->mPatchHandle, delayMs);
+    ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
+    outputDesc->mPatchHandle = 0;
+    return status;
+}
+
+status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
+                                            audio_devices_t device,
+                                            bool force)
+{
+    status_t status = NO_ERROR;
+
+    AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
+    if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
+        inputDesc->mDevice = device;
+
+        DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
+        if (!deviceList.isEmpty()) {
+            struct audio_patch patch;
+            inputDesc->toAudioPortConfig(&patch.sinks[0]);
+            patch.num_sinks = 1;
+            //only one input device for now
+            deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
+            patch.sources[0].ext.device.hw_module = patch.sinks[0].ext.mix.hw_module;
+            patch.num_sources = 1;
+            audio_patch_handle_t patchHandle = inputDesc->mPatchHandle;
+            status_t status = mpClientInterface->createAudioPatch(&patch,
+                                                                  &patchHandle,
+                                                                  0);
+            ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
+                                                                          status, patchHandle);
+            if (status == NO_ERROR) {
+                inputDesc->mPatchHandle = patchHandle;
+            }
+        }
+    }
+    return status;
+}
+
+status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input)
+{
+    AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
+    if (inputDesc->mPatchHandle == 0) {
+        return INVALID_OPERATION;
+    }
+    status_t status = mpClientInterface->releaseAudioPatch(inputDesc->mPatchHandle, 0);
+    ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
+    inputDesc->mPatchHandle = 0;
+    return status;
+}
+
+sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
                                                    uint32_t samplingRate,
                                                    audio_format_t format,
                                                    audio_channel_mask_t channelMask)
@@ -3067,7 +3138,7 @@
         }
         for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
         {
-            IOProfile *profile = mHwModules[i]->mInputProfiles[j];
+            sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
             // profile->log();
             if (profile->isCompatibleProfile(device, samplingRate, format,
                                              channelMask, AUDIO_OUTPUT_FLAG_NONE)) {
@@ -3648,10 +3719,10 @@
 // --- AudioOutputDescriptor class implementation
 
 AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
-        const IOProfile *profile)
-    : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
+        const sp<IOProfile>& profile)
+    : mId(0), mIoHandle(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
       mChannelMask(0), mLatency(0),
-    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE),
+    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0),
     mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
 {
     // clear usage count for all stream types
@@ -3770,6 +3841,32 @@
     return false;
 }
 
+void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig(
+                                                        struct audio_port_config *config) const
+{
+    config->id = mId;
+    config->role = AUDIO_PORT_ROLE_SOURCE;
+    config->type = AUDIO_PORT_TYPE_MIX;
+    config->sample_rate = mSamplingRate;
+    config->channel_mask = mChannelMask;
+    config->format = mFormat;
+    config->gain.index = -1;
+    config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+                            AUDIO_PORT_CONFIG_FORMAT;
+    config->ext.mix.hw_module = mProfile->mModule->mHandle;
+    config->ext.mix.handle = mIoHandle;
+    config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
+}
+
+void AudioPolicyManager::AudioOutputDescriptor::toAudioPort(
+                                                    struct audio_port *port) const
+{
+    mProfile->toAudioPort(port);
+    port->id = mId;
+    port->ext.mix.handle = mIoHandle;
+    port->ext.mix.latency_class =
+            mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
+}
 
 status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
 {
@@ -3803,9 +3900,10 @@
 
 // --- AudioInputDescriptor class implementation
 
-AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const IOProfile *profile)
-    : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
-      mDevice(AUDIO_DEVICE_NONE), mRefCount(0),
+AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
+    : mId(0), mIoHandle(0), mSamplingRate(0),
+      mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
+      mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0), mRefCount(0),
       mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile)
 {
     if (profile != NULL) {
@@ -3815,6 +3913,33 @@
     }
 }
 
+void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig(
+                                                        struct audio_port_config *config) const
+{
+    config->id = mId;
+    config->role = AUDIO_PORT_ROLE_SINK;
+    config->type = AUDIO_PORT_TYPE_MIX;
+    config->sample_rate = mSamplingRate;
+    config->channel_mask = mChannelMask;
+    config->format = mFormat;
+    config->gain.index = -1;
+    config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+                          AUDIO_PORT_CONFIG_FORMAT;
+    config->ext.mix.hw_module = mProfile->mModule->mHandle;
+    config->ext.mix.handle = mIoHandle;
+    config->ext.mix.usecase.source = (mInputSource == AUDIO_SOURCE_HOTWORD) ?
+                                                    AUDIO_SOURCE_VOICE_RECOGNITION : mInputSource;
+}
+
+void AudioPolicyManager::AudioInputDescriptor::toAudioPort(
+                                                    struct audio_port *port) const
+{
+    mProfile->toAudioPort(port);
+    port->id = mId;
+    port->ext.mix.handle = mIoHandle;
+    port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
+}
+
 status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
 {
     const size_t SIZE = 256;
@@ -3897,7 +4022,7 @@
     return NO_ERROR;
 }
 
-// --- IOProfile class implementation
+// --- HwModule class implementation
 
 AudioPolicyManager::HwModule::HwModule(const char *name)
     : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0)
@@ -3908,11 +4033,9 @@
 {
     for (size_t i = 0; i < mOutputProfiles.size(); i++) {
         mOutputProfiles[i]->mSupportedDevices.clear();
-        delete mOutputProfiles[i];
     }
     for (size_t i = 0; i < mInputProfiles.size(); i++) {
         mInputProfiles[i]->mSupportedDevices.clear();
-        delete mInputProfiles[i];
     }
     free((void *)mName);
 }
@@ -3946,8 +4069,129 @@
     }
 }
 
-AudioPolicyManager::IOProfile::IOProfile(HwModule *module)
-    : mFlags((audio_output_flags_t)0), mModule(module)
+// --- AudioPort class implementation
+
+void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const
+{
+    port->role = mRole;
+    port->type = mType;
+    unsigned int i;
+    for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
+        port->sample_rates[i] = mSamplingRates[i];
+    }
+    port->num_sample_rates = i;
+    for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
+        port->channel_masks[i] = mChannelMasks[i];
+    }
+    port->num_channel_masks = i;
+    for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
+        port->formats[i] = mFormats[i];
+    }
+    port->num_formats = i;
+    port->num_gains = 0;
+}
+
+
+void AudioPolicyManager::AudioPort::loadSamplingRates(char *name)
+{
+    char *str = strtok(name, "|");
+
+    // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+    // rates should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        mSamplingRates.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        uint32_t rate = atoi(str);
+        if (rate != 0) {
+            ALOGV("loadSamplingRates() adding rate %d", rate);
+            mSamplingRates.add(rate);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManager::AudioPort::loadFormats(char *name)
+{
+    char *str = strtok(name, "|");
+
+    // by convention, "0' in the first entry in mFormats indicates the supported formats
+    // should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        mFormats.add(AUDIO_FORMAT_DEFAULT);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
+                                                             ARRAY_SIZE(sFormatNameToEnumTable),
+                                                             str);
+        if (format != AUDIO_FORMAT_DEFAULT) {
+            mFormats.add(format);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManager::AudioPort::loadInChannels(char *name)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadInChannels() %s", name);
+
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        mChannelMasks.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_channel_mask_t channelMask =
+                (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sInChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+            mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+void AudioPolicyManager::AudioPort::loadOutChannels(char *name)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadOutChannels() %s", name);
+
+    // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+    // masks should be read from the output stream after it is opened for the first time
+    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+        mChannelMasks.add(0);
+        return;
+    }
+
+    while (str != NULL) {
+        audio_channel_mask_t channelMask =
+                (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sOutChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+// --- IOProfile class implementation
+
+AudioPolicyManager::IOProfile::IOProfile(audio_port_role_t role, HwModule *module)
+    : AudioPort(AUDIO_PORT_TYPE_MIX, role, module), mFlags((audio_output_flags_t)0)
 {
 }
 
@@ -4083,7 +4327,7 @@
     // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
     // - have the same address or one device does not specify the address
     // - have the same channel mask or one device does not specify the channel mask
-    return (mType == other->mType) &&
+    return (mDeviceType == other->mDeviceType) &&
            (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
            (mChannelMask == 0 || other->mChannelMask == 0 ||
                 mChannelMask == other->mChannelMask);
@@ -4091,11 +4335,11 @@
 
 void AudioPolicyManager::DeviceVector::refreshTypes()
 {
-    mTypes = AUDIO_DEVICE_NONE;
+    mDeviceTypes = AUDIO_DEVICE_NONE;
     for(size_t i = 0; i < size(); i++) {
-        mTypes |= itemAt(i)->mType;
+        mDeviceTypes |= itemAt(i)->mDeviceType;
     }
-    ALOGV("DeviceVector::refreshTypes() mTypes %08x", mTypes);
+    ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
 }
 
 ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
@@ -4118,7 +4362,7 @@
             refreshTypes();
         }
     } else {
-        ALOGW("DeviceVector::add device %08x already in", item->mType);
+        ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
         ret = -1;
     }
     return ret;
@@ -4130,7 +4374,7 @@
     ssize_t ret = indexOf(item);
 
     if (ret < 0) {
-        ALOGW("DeviceVector::remove device %08x not in", item->mType);
+        ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
     } else {
         ret = SortedVector::removeAt(ret);
         if (ret >= 0) {
@@ -4155,6 +4399,61 @@
     }
 }
 
+sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDevice(
+                                                        audio_devices_t type, String8 address) const
+{
+    sp<DeviceDescriptor> device;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->mDeviceType == type) {
+            device = itemAt(i);
+            if (itemAt(i)->mAddress = address) {
+                break;
+            }
+        }
+    }
+    ALOGV("DeviceVector::getDevice() for type %d address %s found %p",
+          type, address.string(), device.get());
+    return device;
+}
+
+AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType(
+                                                                        audio_devices_t type) const
+{
+    DeviceVector devices;
+    for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
+        if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
+            devices.add(itemAt(i));
+            type &= ~itemAt(i)->mDeviceType;
+            ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
+                  itemAt(i)->mDeviceType, itemAt(i).get());
+        }
+    }
+    return devices;
+}
+
+void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig(struct audio_port_config *config) const
+{
+    config->id = mId;
+    config->role = audio_is_output_device(mDeviceType) ?
+                        AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+    config->type = AUDIO_PORT_TYPE_DEVICE;
+    config->sample_rate = 0;
+    config->channel_mask = mChannelMask;
+    config->format = AUDIO_FORMAT_DEFAULT;
+    config->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK;
+    config->gain.index = -1;
+    config->ext.device.type = mDeviceType;
+    strncpy(config->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const
+{
+    AudioPort::toAudioPort(port);
+    port->id = mId;
+    port->ext.device.type = mDeviceType;
+    strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
 void AudioPolicyManager::DeviceDescriptor::dumpHeader(int fd, int spaces)
 {
     const size_t SIZE = 256;
@@ -4174,7 +4473,7 @@
                          spaces, "",
                          enumToString(sDeviceNameToEnumTable,
                                       ARRAY_SIZE(sDeviceNameToEnumTable),
-                                      mType),
+                                      mDeviceType),
                          mId, mChannelMask, mAddress.string());
     write(fd, buffer, strlen(buffer));
 
@@ -4225,115 +4524,19 @@
     return device;
 }
 
-void AudioPolicyManager::loadSamplingRates(char *name, IOProfile *profile)
-{
-    char *str = strtok(name, "|");
-
-    // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
-    // rates should be read from the output stream after it is opened for the first time
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        profile->mSamplingRates.add(0);
-        return;
-    }
-
-    while (str != NULL) {
-        uint32_t rate = atoi(str);
-        if (rate != 0) {
-            ALOGV("loadSamplingRates() adding rate %d", rate);
-            profile->mSamplingRates.add(rate);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
-void AudioPolicyManager::loadFormats(char *name, IOProfile *profile)
-{
-    char *str = strtok(name, "|");
-
-    // by convention, "0' in the first entry in mFormats indicates the supported formats
-    // should be read from the output stream after it is opened for the first time
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
-        return;
-    }
-
-    while (str != NULL) {
-        audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
-                                                             ARRAY_SIZE(sFormatNameToEnumTable),
-                                                             str);
-        if (format != AUDIO_FORMAT_DEFAULT) {
-            profile->mFormats.add(format);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
-void AudioPolicyManager::loadInChannels(char *name, IOProfile *profile)
-{
-    const char *str = strtok(name, "|");
-
-    ALOGV("loadInChannels() %s", name);
-
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        profile->mChannelMasks.add(0);
-        return;
-    }
-
-    while (str != NULL) {
-        audio_channel_mask_t channelMask =
-                (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
-                                                   ARRAY_SIZE(sInChannelsNameToEnumTable),
-                                                   str);
-        if (channelMask != 0) {
-            ALOGV("loadInChannels() adding channelMask %04x", channelMask);
-            profile->mChannelMasks.add(channelMask);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
-void AudioPolicyManager::loadOutChannels(char *name, IOProfile *profile)
-{
-    const char *str = strtok(name, "|");
-
-    ALOGV("loadOutChannels() %s", name);
-
-    // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
-    // masks should be read from the output stream after it is opened for the first time
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        profile->mChannelMasks.add(0);
-        return;
-    }
-
-    while (str != NULL) {
-        audio_channel_mask_t channelMask =
-                (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
-                                                   ARRAY_SIZE(sOutChannelsNameToEnumTable),
-                                                   str);
-        if (channelMask != 0) {
-            profile->mChannelMasks.add(channelMask);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
 status_t AudioPolicyManager::loadInput(cnode *root, HwModule *module)
 {
     cnode *node = root->first_child;
 
-    IOProfile *profile = new IOProfile(module);
+    sp<IOProfile> profile = new IOProfile(AUDIO_PORT_ROLE_SINK, module);
 
     while (node) {
         if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
-            loadSamplingRates((char *)node->value, profile);
+            profile->loadSamplingRates((char *)node->value);
         } else if (strcmp(node->name, FORMATS_TAG) == 0) {
-            loadFormats((char *)node->value, profile);
+            profile->loadFormats((char *)node->value);
         } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
-            loadInChannels((char *)node->value, profile);
+            profile->loadInChannels((char *)node->value);
         } else if (strcmp(node->name, DEVICES_TAG) == 0) {
             profile->mSupportedDevices.loadDevicesFromType(parseDeviceNames((char *)node->value));
         }
@@ -4358,7 +4561,6 @@
         module->mInputProfiles.add(profile);
         return NO_ERROR;
     } else {
-        delete profile;
         return BAD_VALUE;
     }
 }
@@ -4367,15 +4569,15 @@
 {
     cnode *node = root->first_child;
 
-    IOProfile *profile = new IOProfile(module);
+    sp<IOProfile> profile = new IOProfile(AUDIO_PORT_ROLE_SOURCE, module);
 
     while (node) {
         if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
-            loadSamplingRates((char *)node->value, profile);
+            profile->loadSamplingRates((char *)node->value);
         } else if (strcmp(node->name, FORMATS_TAG) == 0) {
-            loadFormats((char *)node->value, profile);
+            profile->loadFormats((char *)node->value);
         } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
-            loadOutChannels((char *)node->value, profile);
+            profile->loadOutChannels((char *)node->value);
         } else if (strcmp(node->name, DEVICES_TAG) == 0) {
             profile->mSupportedDevices.loadDevicesFromType(parseDeviceNames((char *)node->value));
         } else if (strcmp(node->name, FLAGS_TAG) == 0) {
@@ -4402,7 +4604,6 @@
         module->mOutputProfiles.add(profile);
         return NO_ERROR;
     } else {
-        delete profile;
         return BAD_VALUE;
     }
 }
@@ -4480,7 +4681,7 @@
             } else {
                 ALOGW("loadGlobalConfig() default device not specified");
             }
-            ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mType);
+            ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType);
         } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
             mAvailableInputDevices.loadDevicesFromType(parseDeviceNames((char *)node->value));
             ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
@@ -4519,14 +4720,14 @@
 void AudioPolicyManager::defaultAudioPolicyConfig(void)
 {
     HwModule *module;
-    IOProfile *profile;
+    sp<IOProfile> profile;
     sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(AUDIO_DEVICE_IN_BUILTIN_MIC);
     mAvailableOutputDevices.add(mDefaultOutputDevice);
     mAvailableInputDevices.add(defaultInputDevice);
 
     module = new HwModule("primary");
 
-    profile = new IOProfile(module);
+    profile = new IOProfile(AUDIO_PORT_ROLE_SOURCE, module);
     profile->mSamplingRates.add(44100);
     profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
     profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
@@ -4534,7 +4735,7 @@
     profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
     module->mOutputProfiles.add(profile);
 
-    profile = new IOProfile(module);
+    profile = new IOProfile(AUDIO_PORT_ROLE_SINK, module);
     profile->mSamplingRates.add(8000);
     profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
     profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
index f00fa8a..905a3c8 100644
--- a/services/audiopolicy/AudioPolicyManager.h
+++ b/services/audiopolicy/AudioPolicyManager.h
@@ -175,47 +175,6 @@
 
         class IOProfile;
 
-        class DeviceDescriptor: public RefBase
-        {
-        public:
-            DeviceDescriptor(audio_devices_t type, String8 address,
-                             audio_channel_mask_t channelMask) :
-                                 mType(type), mAddress(address),
-                                 mChannelMask(channelMask), mId(0) {}
-
-            DeviceDescriptor(audio_devices_t type) :
-                                 mType(type), mAddress(""),
-                                 mChannelMask(0), mId(0) {}
-
-            status_t dump(int fd, int spaces) const;
-            static void dumpHeader(int fd, int spaces);
-
-            bool equals(const sp<DeviceDescriptor>& other) const;
-
-            audio_devices_t mType;
-            String8 mAddress;
-            audio_channel_mask_t mChannelMask;
-            uint32_t mId;
-        };
-
-        class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
-        {
-        public:
-            DeviceVector() : SortedVector(), mTypes(AUDIO_DEVICE_NONE) {}
-
-            ssize_t         add(const sp<DeviceDescriptor>& item);
-            ssize_t         remove(const sp<DeviceDescriptor>& item);
-            ssize_t         indexOf(const sp<DeviceDescriptor>& item) const;
-
-            audio_devices_t types() const { return mTypes; }
-
-            void loadDevicesFromType(audio_devices_t types);
-
-        private:
-            void refreshTypes();
-            audio_devices_t mTypes;
-        };
-
         class HwModule {
         public:
                     HwModule(const char *name);
@@ -225,8 +184,88 @@
 
             const char *const mName; // base name of the audio HW module (primary, a2dp ...)
             audio_module_handle_t mHandle;
-            Vector <IOProfile *> mOutputProfiles; // output profiles exposed by this module
-            Vector <IOProfile *> mInputProfiles;  // input profiles exposed by this module
+            Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
+            Vector < sp<IOProfile> > mInputProfiles;  // input profiles exposed by this module
+        };
+
+        class AudioPort: public RefBase
+        {
+        public:
+            AudioPort(audio_port_type_t type, audio_port_role_t role, HwModule *module) :
+                mType(type), mRole(role), mModule(module) {}
+            virtual ~AudioPort() {}
+
+            virtual void toAudioPort(struct audio_port *port) const;
+
+            void loadSamplingRates(char *name);
+            void loadFormats(char *name);
+            void loadOutChannels(char *name);
+            void loadInChannels(char *name);
+
+            audio_port_type_t        mType;
+            audio_port_role_t        mRole;
+            // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
+            // indicates the supported parameters should be read from the output stream
+            // after it is opened for the first time
+            Vector <uint32_t> mSamplingRates; // supported sampling rates
+            Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
+            Vector <audio_format_t> mFormats; // supported audio formats
+            HwModule *mModule;                     // audio HW module exposing this I/O stream
+        };
+
+
+        class DeviceDescriptor: public AudioPort
+        {
+        public:
+            DeviceDescriptor(audio_devices_t type, String8 address,
+                             audio_channel_mask_t channelMask) :
+                                 AudioPort(AUDIO_PORT_TYPE_DEVICE,
+                                           audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+                                                                          AUDIO_PORT_ROLE_SOURCE,
+                                         NULL),
+                                 mDeviceType(type), mAddress(address),
+                                 mChannelMask(channelMask), mId(0) {}
+
+            DeviceDescriptor(audio_devices_t type) :
+                                AudioPort(AUDIO_PORT_TYPE_DEVICE,
+                                          audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+                                                                         AUDIO_PORT_ROLE_SOURCE,
+                                        NULL),
+                                mDeviceType(type), mAddress(""),
+                                mChannelMask(0), mId(0) {}
+            virtual ~DeviceDescriptor() {}
+
+            bool equals(const sp<DeviceDescriptor>& other) const;
+            void toAudioPortConfig(struct audio_port_config *config) const;
+            virtual void toAudioPort(struct audio_port *port) const;
+
+            status_t dump(int fd, int spaces) const;
+            static void dumpHeader(int fd, int spaces);
+
+            audio_devices_t mDeviceType;
+            String8 mAddress;
+            audio_channel_mask_t mChannelMask;
+            audio_port_handle_t mId;
+        };
+
+        class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
+        {
+        public:
+            DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
+
+            ssize_t         add(const sp<DeviceDescriptor>& item);
+            ssize_t         remove(const sp<DeviceDescriptor>& item);
+            ssize_t         indexOf(const sp<DeviceDescriptor>& item) const;
+
+            audio_devices_t types() const { return mDeviceTypes; }
+
+            void loadDevicesFromType(audio_devices_t types);
+            sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
+            DeviceVector getDevicesFromType(audio_devices_t types) const;
+
+        private:
+            void refreshTypes();
+            audio_devices_t mDeviceTypes;
         };
 
         // the IOProfile class describes the capabilities of an output or input stream.
@@ -234,11 +273,11 @@
         // It is used by the policy manager to determine if an output or input is suitable for
         // a given use case,  open/close it accordingly and connect/disconnect audio tracks
         // to/from it.
-        class IOProfile
+        class IOProfile : public AudioPort
         {
         public:
-            IOProfile(HwModule *module);
-            ~IOProfile();
+            IOProfile(audio_port_role_t role, HwModule *module);
+            virtual ~IOProfile();
 
             bool isCompatibleProfile(audio_devices_t device,
                                      uint32_t samplingRate,
@@ -249,17 +288,10 @@
             void dump(int fd);
             void log();
 
-            // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
-            // indicates the supported parameters should be read from the output stream
-            // after it is opened for the first time
-            Vector <uint32_t> mSamplingRates; // supported sampling rates
-            Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
-            Vector <audio_format_t> mFormats; // supported audio formats
             DeviceVector  mSupportedDevices; // supported devices
                                              // (devices this output can be routed to)
             audio_output_flags_t mFlags; // attribute flags (e.g primary output,
                                                 // direct output...). For outputs only.
-            HwModule *mModule;                     // audio HW module exposing this I/O stream
         };
 
         // default volume curve
@@ -284,7 +316,7 @@
         class AudioOutputDescriptor
         {
         public:
-            AudioOutputDescriptor(const IOProfile *profile);
+            AudioOutputDescriptor(const sp<IOProfile>& profile);
 
             status_t    dump(int fd);
 
@@ -303,20 +335,25 @@
                              uint32_t inPastMs = 0,
                              nsecs_t sysTime = 0) const;
 
-            audio_io_handle_t mId;              // output handle
+            void toAudioPortConfig(struct audio_port_config *config) const;
+            void toAudioPort(struct audio_port *port) const;
+
+            audio_port_handle_t mId;
+            audio_io_handle_t mIoHandle;              // output handle
             uint32_t mSamplingRate;             //
             audio_format_t mFormat;             //
             audio_channel_mask_t mChannelMask;     // output configuration
             uint32_t mLatency;                  //
             audio_output_flags_t mFlags;   //
             audio_devices_t mDevice;                   // current device this output is routed to
+            audio_patch_handle_t mPatchHandle;
             uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
             nsecs_t mStopTime[AUDIO_STREAM_CNT];
             AudioOutputDescriptor *mOutput1;    // used by duplicated outputs: first output
             AudioOutputDescriptor *mOutput2;    // used by duplicated outputs: second output
             float mCurVolume[AUDIO_STREAM_CNT];   // current stream volume
             int mMuteCount[AUDIO_STREAM_CNT];     // mute request counter
-            const IOProfile *mProfile;          // I/O profile this output derives from
+            const sp<IOProfile> mProfile;          // I/O profile this output derives from
             bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
                                                 // device selection. See checkDeviceMuteStrategies()
             uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
@@ -327,18 +364,23 @@
         class AudioInputDescriptor
         {
         public:
-            AudioInputDescriptor(const IOProfile *profile);
+            AudioInputDescriptor(const sp<IOProfile>& profile);
 
             status_t    dump(int fd);
 
-            audio_io_handle_t mId;                      // input handle
+            audio_port_handle_t mId;
+            audio_io_handle_t mIoHandle;              // input handle
             uint32_t mSamplingRate;                     //
             audio_format_t mFormat;                     // input configuration
             audio_channel_mask_t mChannelMask;             //
             audio_devices_t mDevice;                    // current device this input is routed to
+            audio_patch_handle_t mPatchHandle;
             uint32_t mRefCount;                         // number of AudioRecord clients using this output
             audio_source_t mInputSource;                // input source selected by application (mediarecorder.h)
-            const IOProfile *mProfile;                  // I/O profile this output derives from
+            const sp<IOProfile> mProfile;                  // I/O profile this output derives from
+
+            void toAudioPortConfig(struct audio_port_config *config) const;
+            void toAudioPort(struct audio_port *port) const;
         };
 
         // stream descriptor used for volume control
@@ -372,8 +414,8 @@
             bool mEnabled;              // enabled state: CPU load being used or not
         };
 
-        void addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc);
-        void addInput(audio_io_handle_t id, AudioInputDescriptor *inputDesc);
+        void addOutput(audio_io_handle_t output, AudioOutputDescriptor *outputDesc);
+        void addInput(audio_io_handle_t input, AudioInputDescriptor *inputDesc);
 
         // return the strategy corresponding to a given stream type
         static routing_strategy getStrategy(audio_stream_type_t stream);
@@ -398,6 +440,12 @@
                              audio_devices_t device,
                              bool force = false,
                              int delayMs = 0);
+        status_t resetOutputDevice(audio_io_handle_t output,
+                                   int delayMs = 0);
+        status_t setInputDevice(audio_io_handle_t input,
+                                audio_devices_t device,
+                                bool force = false);
+        status_t resetInputDevice(audio_io_handle_t input);
 
         // select input device corresponding to requested audio source
         virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
@@ -484,16 +532,18 @@
         // must be called every time a condition that affects the device choice for a given output is
         // changed: connected device, phone state, force use, output start, output stop..
         // see getDeviceForStrategy() for the use of fromCache parameter
+        audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
 
-        audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache);
         // updates cache of device used by all strategies (mDeviceForStrategy[])
         // must be called every time a condition that affects the device choice for a given strategy is
         // changed: connected device, phone state, force use...
         // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
          // Must be called after checkOutputForAllStrategies()
-
         void updateDevicesAndOutputs();
 
+        // selects the most appropriate device on input for current state
+        audio_devices_t getNewInputDevice(audio_io_handle_t input);
+
         virtual uint32_t getMaxEffectsCpuLoad();
         virtual uint32_t getMaxEffectsMemory();
 #ifdef AUDIO_POLICY_TEST
@@ -525,11 +575,11 @@
 
         audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
                                        audio_output_flags_t flags);
-        IOProfile *getInputProfile(audio_devices_t device,
+        sp<IOProfile> getInputProfile(audio_devices_t device,
                                    uint32_t samplingRate,
                                    audio_format_t format,
                                    audio_channel_mask_t channelMask);
-        IOProfile *getProfileForDirectOutput(audio_devices_t device,
+        sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
                                                        uint32_t samplingRate,
                                                        audio_format_t format,
                                                        audio_channel_mask_t channelMask,
@@ -551,10 +601,6 @@
         static bool stringToBool(const char *value);
         static audio_output_flags_t parseFlagNames(char *name);
         static audio_devices_t parseDeviceNames(char *name);
-        void loadSamplingRates(char *name, IOProfile *profile);
-        void loadFormats(char *name, IOProfile *profile);
-        void loadOutChannels(char *name, IOProfile *profile);
-        void loadInChannels(char *name, IOProfile *profile);
         status_t loadOutput(cnode *root,  HwModule *module);
         status_t loadInput(cnode *root,  HwModule *module);
         void loadHwModule(cnode *root);
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp
index 4e9a2f0..ea573a4 100644
--- a/services/audiopolicy/AudioPolicyService.cpp
+++ b/services/audiopolicy/AudioPolicyService.cpp
@@ -150,6 +150,19 @@
 #endif
 }
 
+status_t AudioPolicyService::clientCreateAudioPatch(const struct audio_patch *patch,
+                                                audio_patch_handle_t *handle,
+                                                int delayMs)
+{
+    return mAudioCommandThread->createAudioPatchCommand(patch, handle, delayMs);
+}
+
+status_t AudioPolicyService::clientReleaseAudioPatch(audio_patch_handle_t handle,
+                                                 int delayMs)
+{
+    return mAudioCommandThread->releaseAudioPatchCommand(handle, delayMs);
+}
+
 
 void AudioPolicyService::binderDied(const wp<IBinder>& who) {
     ALOGW("binderDied() %p, calling pid %d", who.unsafe_get(),
@@ -357,6 +370,26 @@
                     svc->doReleaseOutput(data->mIO);
                     mLock.lock();
                     }break;
+                case CREATE_AUDIO_PATCH: {
+                    CreateAudioPatchData *data = (CreateAudioPatchData *)command->mParam.get();
+                    ALOGV("AudioCommandThread() processing create audio patch");
+                    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+                    if (af == 0) {
+                        command->mStatus = PERMISSION_DENIED;
+                    } else {
+                        command->mStatus = af->createAudioPatch(&data->mPatch, &data->mHandle);
+                    }
+                    } break;
+                case RELEASE_AUDIO_PATCH: {
+                    ReleaseAudioPatchData *data = (ReleaseAudioPatchData *)command->mParam.get();
+                    ALOGV("AudioCommandThread() processing release audio patch");
+                    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+                    if (af == 0) {
+                        command->mStatus = PERMISSION_DENIED;
+                    } else {
+                        command->mStatus = af->releaseAudioPatch(data->mHandle);
+                    }
+                    } break;
                 default:
                     ALOGW("AudioCommandThread() unknown command %d", command->mCommand);
                 }
@@ -516,6 +549,41 @@
     sendCommand(command);
 }
 
+status_t AudioPolicyService::AudioCommandThread::createAudioPatchCommand(
+                                                const struct audio_patch *patch,
+                                                audio_patch_handle_t *handle,
+                                                int delayMs)
+{
+    status_t status = NO_ERROR;
+
+    sp<AudioCommand> command = new AudioCommand();
+    command->mCommand = CREATE_AUDIO_PATCH;
+    CreateAudioPatchData *data = new CreateAudioPatchData();
+    data->mPatch = *patch;
+    data->mHandle = *handle;
+    command->mParam = data;
+    command->mWaitStatus = true;
+    ALOGV("AudioCommandThread() adding create patch delay %d", delayMs);
+    status = sendCommand(command, delayMs);
+    if (status == NO_ERROR) {
+        *handle = data->mHandle;
+    }
+    return status;
+}
+
+status_t AudioPolicyService::AudioCommandThread::releaseAudioPatchCommand(audio_patch_handle_t handle,
+                                                 int delayMs)
+{
+    sp<AudioCommand> command = new AudioCommand();
+    command->mCommand = RELEASE_AUDIO_PATCH;
+    ReleaseAudioPatchData *data = new ReleaseAudioPatchData();
+    data->mHandle = handle;
+    command->mParam = data;
+    command->mWaitStatus = true;
+    ALOGV("AudioCommandThread() adding release patch delay %d", delayMs);
+    return sendCommand(command, delayMs);
+}
+
 status_t AudioPolicyService::AudioCommandThread::sendCommand(sp<AudioCommand>& command, int delayMs)
 {
     {
@@ -534,6 +602,7 @@
     return command->mStatus;
 }
 
+
 // insertCommand_l() must be called with mLock held
 void AudioPolicyService::AudioCommandThread::insertCommand_l(sp<AudioCommand>& command, int delayMs)
 {
diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/AudioPolicyService.h
index 26037e4..a7ed9e8 100644
--- a/services/audiopolicy/AudioPolicyService.h
+++ b/services/audiopolicy/AudioPolicyService.h
@@ -145,6 +145,12 @@
                                   int session = 0);
             void doReleaseOutput(audio_io_handle_t output);
 
+            status_t clientCreateAudioPatch(const struct audio_patch *patch,
+                                      audio_patch_handle_t *handle,
+                                      int delayMs);
+            status_t clientReleaseAudioPatch(audio_patch_handle_t handle,
+                                             int delayMs);
+
 private:
                         AudioPolicyService() ANDROID_API;
     virtual             ~AudioPolicyService();
@@ -169,7 +175,9 @@
             SET_PARAMETERS,
             SET_VOICE_VOLUME,
             STOP_OUTPUT,
-            RELEASE_OUTPUT
+            RELEASE_OUTPUT,
+            CREATE_AUDIO_PATCH,
+            RELEASE_AUDIO_PATCH,
         };
 
         AudioCommandThread (String8 name, const wp<AudioPolicyService>& service);
@@ -196,6 +204,13 @@
                     void        releaseOutputCommand(audio_io_handle_t output);
                     status_t    sendCommand(sp<AudioCommand>& command, int delayMs = 0);
                     void        insertCommand_l(sp<AudioCommand>& command, int delayMs = 0);
+                    status_t    createAudioPatchCommand(const struct audio_patch *patch,
+                                                        audio_patch_handle_t *handle,
+                                                        int delayMs);
+                    status_t    releaseAudioPatchCommand(audio_patch_handle_t handle,
+                                                         int delayMs);
+
+                    void        insertCommand_l(AudioCommand *command, int delayMs = 0);
 
     private:
         class AudioCommandData;
@@ -261,6 +276,17 @@
             audio_io_handle_t mIO;
         };
 
+        class CreateAudioPatchData : public AudioCommandData {
+        public:
+            struct audio_patch mPatch;
+            audio_patch_handle_t mHandle;
+        };
+
+        class ReleaseAudioPatchData : public AudioCommandData {
+        public:
+            audio_patch_handle_t mHandle;
+        };
+
         Mutex   mLock;
         Condition mWaitWorkCV;
         Vector < sp<AudioCommand> > mAudioCommands; // list of pending commands
@@ -405,6 +431,15 @@
                                          audio_io_handle_t srcOutput,
                                          audio_io_handle_t dstOutput);
 
+        /* Create a patch between several source and sink ports */
+        virtual status_t createAudioPatch(const struct audio_patch *patch,
+                                           audio_patch_handle_t *handle,
+                                           int delayMs);
+
+        /* Release a patch */
+        virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
+                                           int delayMs);
+
      private:
         AudioPolicyService *mAudioPolicyService;
     };