Split ManagerDefault into Manager and ManagerDefinitions

This patch splits the managerdefault into a manager and a manager
defintion library that contains all pillar elements of a policy manager.

It renames the file with the name of the main class they contains.

It splits the AudioPort into AudioPort and AudioPatch.

Change-Id: I992cf0b8aed895805cc003ba0980d2c9e92c985b
Signed-off-by: François Gaffie <francois.gaffie@intel.com>

22 files changed, 3430 insertions, 0 deletions

diff --git a/services/audiopolicy/common/Android.mk b/services/audiopolicy/common/Android.mk
new file mode 100644
index 0000000..dcce8e3
--- /dev/null
+++ b/services/audiopolicy/common/Android.mk
@@ -0,0 +1,9 @@
+
+LOCAL_PATH := $(call my-dir)
+include $(CLEAR_VARS)
+
+#######################################################################
+# Recursive call sub-folder Android.mk
+#
+include $(call all-makefiles-under,$(LOCAL_PATH))
+
diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk
new file mode 100644
index 0000000..d9d1477
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/Android.mk
@@ -0,0 +1,29 @@
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+    src/DeviceDescriptor.cpp \
+    src/AudioGain.cpp \
+    src/HwModule.cpp \
+    src/IOProfile.cpp \
+    src/AudioPort.cpp \
+    src/AudioPatch.cpp \
+    src/AudioInputDescriptor.cpp \
+    src/AudioOutputDescriptor.cpp \
+    src/ConfigParsingUtils.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libutils \
+    liblog \
+
+LOCAL_C_INCLUDES += \
+    $(LOCAL_PATH)/include \
+
+LOCAL_EXPORT_C_INCLUDE_DIRS := \
+    $(LOCAL_PATH)/include
+
+LOCAL_MODULE := libaudiopolicycomponents
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/services/audiopolicy/common/managerdefinitions/include/ApmImplDefinitions.h b/services/audiopolicy/common/managerdefinitions/include/ApmImplDefinitions.h
new file mode 100644
index 0000000..62927da
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/ApmImplDefinitions.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+namespace android {
+
+enum routing_strategy {
+    STRATEGY_MEDIA,
+    STRATEGY_PHONE,
+    STRATEGY_SONIFICATION,
+    STRATEGY_SONIFICATION_RESPECTFUL,
+    STRATEGY_DTMF,
+    STRATEGY_ENFORCED_AUDIBLE,
+    STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
+    STRATEGY_ACCESSIBILITY,
+    STRATEGY_REROUTING,
+    NUM_STRATEGIES
+};
+
+}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h b/services/audiopolicy/common/managerdefinitions/include/AudioGain.h
new file mode 100644
index 0000000..f638c8e
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioGain.h
@@ -0,0 +1,119 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <system/audio.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+class VolumeCurvePoint
+{
+public:
+    int mIndex;
+    float mDBAttenuation;
+};
+
+class StreamDescriptor;
+
+class ApmGains
+{
+public :
+    // 4 points to define the volume attenuation curve, each characterized by the volume
+    // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
+    // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+    enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
+
+    // device categories used for volume curve management.
+    enum device_category {
+        DEVICE_CATEGORY_HEADSET,
+        DEVICE_CATEGORY_SPEAKER,
+        DEVICE_CATEGORY_EARPIECE,
+        DEVICE_CATEGORY_EXT_MEDIA,
+        DEVICE_CATEGORY_CNT
+    };
+
+    // returns the category the device belongs to with regard to volume curve management
+    static ApmGains::device_category getDeviceCategory(audio_devices_t device);
+
+    // extract one device relevant for volume control from multiple device selection
+    static audio_devices_t getDeviceForVolume(audio_devices_t device);
+
+    static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+                    int indexInUi);
+
+    // default volume curve
+    static const VolumeCurvePoint sDefaultVolumeCurve[ApmGains::VOLCNT];
+    // default volume curve for media strategy
+    static const VolumeCurvePoint sDefaultMediaVolumeCurve[ApmGains::VOLCNT];
+    // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
+    static const VolumeCurvePoint sExtMediaSystemVolumeCurve[ApmGains::VOLCNT];
+    // volume curve for media strategy on speakers
+    static const VolumeCurvePoint sSpeakerMediaVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT];
+    // volume curve for sonification strategy on speakers
+    static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sDefaultSystemVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sHeadsetSystemVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sDefaultVoiceVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sLinearVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sSilentVolumeCurve[ApmGains::VOLCNT];
+    static const VolumeCurvePoint sFullScaleVolumeCurve[ApmGains::VOLCNT];
+    // default volume curves per stream and device category. See initializeVolumeCurves()
+    static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+
+class AudioGain: public RefBase
+{
+public:
+    AudioGain(int index, bool useInChannelMask);
+    virtual ~AudioGain() {}
+
+    void dump(int fd, int spaces, int index) const;
+
+    void getDefaultConfig(struct audio_gain_config *config);
+    status_t checkConfig(const struct audio_gain_config *config);
+    int               mIndex;
+    struct audio_gain mGain;
+    bool              mUseInChannelMask;
+};
+
+
+// stream descriptor used for volume control
+class StreamDescriptor
+{
+public:
+    StreamDescriptor();
+
+    int getVolumeIndex(audio_devices_t device);
+    void dump(int fd);
+
+    int mIndexMin;      // min volume index
+    int mIndexMax;      // max volume index
+    KeyedVector<audio_devices_t, int> mIndexCur;   // current volume index per device
+    bool mCanBeMuted;   // true is the stream can be muted
+
+    const VolumeCurvePoint *mVolumeCurve[ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
new file mode 100644
index 0000000..599c295
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "AudioPort.h"
+#include <utils/Errors.h>
+#include <system/audio.h>
+#include <utils/SortedVector.h>
+
+namespace android {
+
+class IOProfile;
+class AudioMix;
+
+// descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+// and keep track of the usage of this input.
+class AudioInputDescriptor: public AudioPortConfig
+{
+public:
+    AudioInputDescriptor(const sp<IOProfile>& profile);
+    void setIoHandle(audio_io_handle_t ioHandle);
+
+    status_t    dump(int fd);
+
+    audio_port_handle_t           mId;
+    audio_io_handle_t             mIoHandle;       // input handle
+    audio_devices_t               mDevice;         // current device this input is routed to
+    AudioMix                      *mPolicyMix;     // non NULL when used by a dynamic policy
+    audio_patch_handle_t          mPatchHandle;
+    uint32_t                      mRefCount;       // number of AudioRecord clients using
+    // this input
+    uint32_t                      mOpenRefCount;
+    audio_source_t                mInputSource;    // input source selected by application
+    //(mediarecorder.h)
+    const sp<IOProfile>           mProfile;        // I/O profile this output derives from
+    SortedVector<audio_session_t> mSessions;       // audio sessions attached to this input
+    bool                          mIsSoundTrigger; // used by a soundtrigger capture
+
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+            const struct audio_port_config *srcConfig = NULL) const;
+    virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+    void toAudioPort(struct audio_port *port) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
new file mode 100644
index 0000000..8469b81
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "AudioPort.h"
+#include "ApmImplDefinitions.h"
+#include <utils/Errors.h>
+#include <utils/Timers.h>
+#include <system/audio.h>
+
+namespace android {
+
+class IOProfile;
+class AudioMix;
+
+// descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+// and keep track of the usage of this output by each audio stream type.
+class AudioOutputDescriptor: public AudioPortConfig
+{
+public:
+    AudioOutputDescriptor(const sp<IOProfile>& profile);
+
+    status_t    dump(int fd);
+
+    audio_devices_t device() const;
+    void changeRefCount(audio_stream_type_t stream, int delta);
+
+    void setIoHandle(audio_io_handle_t ioHandle);
+    bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+    audio_devices_t supportedDevices();
+    uint32_t latency();
+    bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
+    bool isActive(uint32_t inPastMs = 0) const;
+    bool isStreamActive(audio_stream_type_t stream,
+                        uint32_t inPastMs = 0,
+                        nsecs_t sysTime = 0) const;
+
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+                           const struct audio_port_config *srcConfig = NULL) const;
+    virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+    void toAudioPort(struct audio_port *port) const;
+
+    audio_port_handle_t mId;
+    audio_io_handle_t mIoHandle;              // output handle
+    uint32_t mLatency;                  //
+    audio_output_flags_t mFlags;   //
+    audio_devices_t mDevice;                   // current device this output is routed to
+    AudioMix *mPolicyMix;             // non NULL when used by a dynamic policy
+    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];
+    sp<AudioOutputDescriptor> mOutput1;    // used by duplicated outputs: first output
+    sp<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 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)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
new file mode 100644
index 0000000..6e8fb32
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <system/audio.h>
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+
+namespace android {
+
+class AudioPatch : public RefBase
+{
+public:
+    AudioPatch(const struct audio_patch *patch, uid_t uid);
+
+    status_t dump(int fd, int spaces, int index) const;
+
+    audio_patch_handle_t mHandle;
+    struct audio_patch mPatch;
+    uid_t mUid;
+    audio_patch_handle_t mAfPatchHandle;
+
+private:
+    static volatile int32_t mNextUniqueId;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
new file mode 100644
index 0000000..52e845c
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -0,0 +1,126 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <utils/String8.h>
+#include <utils/Vector.h>
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <system/audio.h>
+#include <cutils/config_utils.h>
+
+#define MAX_MIXER_SAMPLING_RATE 48000
+#define MAX_MIXER_CHANNEL_COUNT 8
+
+namespace android {
+
+class HwModule;
+class AudioGain;
+
+class AudioPort: public virtual RefBase
+{
+public:
+    AudioPort(const String8& name, audio_port_type_t type,
+            audio_port_role_t role, const sp<HwModule>& module);
+    virtual ~AudioPort() {}
+
+    audio_port_handle_t getHandle() { return mId; }
+
+    void attach(const sp<HwModule>& module);
+    bool isAttached() { return mId != 0; }
+
+    static audio_port_handle_t getNextUniqueId();
+
+    virtual void toAudioPort(struct audio_port *port) const;
+
+    void importAudioPort(const sp<AudioPort> port);
+    void clearCapabilities();
+
+    void loadSamplingRates(char *name);
+    void loadFormats(char *name);
+    void loadOutChannels(char *name);
+    void loadInChannels(char *name);
+
+    audio_gain_mode_t loadGainMode(char *name);
+    void loadGain(cnode *root, int index);
+    virtual void loadGains(cnode *root);
+
+    // searches for an exact match
+    status_t checkExactSamplingRate(uint32_t samplingRate) const;
+    // searches for a compatible match, and returns the best match via updatedSamplingRate
+    status_t checkCompatibleSamplingRate(uint32_t samplingRate,
+            uint32_t *updatedSamplingRate) const;
+    // searches for an exact match
+    status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
+    // searches for a compatible match, currently implemented for input channel masks only
+    status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
+    status_t checkFormat(audio_format_t format) const;
+    status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
+
+    uint32_t pickSamplingRate() const;
+    audio_channel_mask_t pickChannelMask() const;
+    audio_format_t pickFormat() const;
+
+    static const audio_format_t sPcmFormatCompareTable[];
+    static int compareFormats(audio_format_t format1, audio_format_t format2);
+
+    void dump(int fd, int spaces) const;
+
+    String8           mName;
+    audio_port_type_t mType;
+    audio_port_role_t mRole;
+    bool              mUseInChannelMask;
+    // 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
+    Vector < sp<AudioGain> > mGains; // gain controllers
+    sp<HwModule> mModule;                 // audio HW module exposing this I/O stream
+    uint32_t mFlags; // attribute flags (e.g primary output,
+                     // direct output...).
+
+
+protected:
+    //TODO - clarify the role of mId in this case, both an "attached" indicator
+    // and a unique ID for identifying a port to the (upcoming) selection API,
+    // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor.
+    audio_port_handle_t mId;
+
+private:
+    static volatile int32_t mNextUniqueId;
+};
+
+class AudioPortConfig: public virtual RefBase
+{
+public:
+    AudioPortConfig();
+    virtual ~AudioPortConfig() {}
+
+    status_t applyAudioPortConfig(const struct audio_port_config *config,
+            struct audio_port_config *backupConfig = NULL);
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+            const struct audio_port_config *srcConfig = NULL) const = 0;
+    virtual sp<AudioPort> getAudioPort() const = 0;
+    uint32_t mSamplingRate;
+    audio_format_t mFormat;
+    audio_channel_mask_t mChannelMask;
+    struct audio_gain_config mGain;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h
new file mode 100644
index 0000000..45e96d9
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h
@@ -0,0 +1,166 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <system/audio.h>
+#include <sys/types.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// Definitions for audio_policy.conf file parsing
+// ----------------------------------------------------------------------------
+
+struct StringToEnum {
+    const char *name;
+    uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+#endif
+
+const StringToEnum sDeviceNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
+    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
+    STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
+};
+
+const StringToEnum sOutputFlagNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+    STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
+};
+
+const StringToEnum sInputFlagNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
+    STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
+};
+
+const StringToEnum sFormatNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+    STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+    STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
+    STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
+    STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+    STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
+    STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
+    STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
+    STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+    STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
+};
+
+const StringToEnum sOutChannelsNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const StringToEnum sInChannelsNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+const StringToEnum sGainModeNameToEnumTable[] = {
+    STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
+    STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
+    STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
+};
+
+class ConfigParsingUtils
+{
+public:
+    static uint32_t stringToEnum(const struct StringToEnum *table,
+            size_t size,
+            const char *name);
+    static const char *enumToString(const struct StringToEnum *table,
+            size_t size,
+            uint32_t value);
+    static bool stringToBool(const char *value);
+    static uint32_t parseOutputFlagNames(char *name);
+    static uint32_t parseInputFlagNames(char *name);
+    static audio_devices_t parseDeviceNames(char *name);
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
new file mode 100644
index 0000000..e63ea2b
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "AudioPort.h"
+#include <utils/Errors.h>
+#include <utils/String8.h>
+#include <utils/SortedVector.h>
+#include <cutils/config_utils.h>
+#include <system/audio.h>
+
+namespace android {
+
+class DeviceDescriptor: public AudioPort, public AudioPortConfig
+{
+public:
+    DeviceDescriptor(const String8& name, audio_devices_t type);
+
+    virtual ~DeviceDescriptor() {}
+
+    bool equals(const sp<DeviceDescriptor>& other) const;
+
+    // AudioPortConfig
+    virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+            const struct audio_port_config *srcConfig = NULL) const;
+
+    // AudioPort
+    virtual void loadGains(cnode *root);
+    virtual void toAudioPort(struct audio_port *port) const;
+
+    status_t dump(int fd, int spaces, int index) const;
+
+    audio_devices_t mDeviceType;
+    String8 mAddress;
+
+    static String8  emptyNameStr;
+};
+
+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);
+    void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
+
+    sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
+    DeviceVector getDevicesFromType(audio_devices_t types) const;
+    sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
+    sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
+    DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
+    const;
+
+private:
+    void refreshTypes();
+    audio_devices_t mDeviceTypes;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
new file mode 100644
index 0000000..3d4e2ad
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "DeviceDescriptor.h"
+#include <utils/RefBase.h>
+#include <utils/String8.h>
+#include <utils/Errors.h>
+#include <utils/Vector.h>
+#include <system/audio.h>
+#include <cutils/config_utils.h>
+
+namespace android {
+
+class IOProfile;
+
+class HwModule : public RefBase
+{
+public:
+    HwModule(const char *name);
+    ~HwModule();
+
+    status_t loadOutput(cnode *root);
+    status_t loadInput(cnode *root);
+    status_t loadDevice(cnode *root);
+
+    status_t addOutputProfile(String8 name, const audio_config_t *config,
+            audio_devices_t device, String8 address);
+    status_t removeOutputProfile(String8 name);
+    status_t addInputProfile(String8 name, const audio_config_t *config,
+            audio_devices_t device, String8 address);
+    status_t removeInputProfile(String8 name);
+
+    void dump(int fd);
+
+    const char *const        mName; // base name of the audio HW module (primary, a2dp ...)
+    uint32_t                 mHalVersion; // audio HAL API version
+    audio_module_handle_t    mHandle;
+    Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
+    Vector < sp<IOProfile> > mInputProfiles;  // input profiles exposed by this module
+    DeviceVector             mDeclaredDevices; // devices declared in audio_policy.conf
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
new file mode 100644
index 0000000..095e759
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "AudioPort.h"
+#include "DeviceDescriptor.h"
+#include <utils/String8.h>
+#include <system/audio.h>
+
+namespace android {
+
+class HwModule;
+
+// the IOProfile class describes the capabilities of an output or input stream.
+// It is currently assumed that all combination of listed parameters are supported.
+// 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 : public AudioPort
+{
+public:
+    IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
+    virtual ~IOProfile();
+
+    // This method is used for both output and input.
+    // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
+    // For input, flags is interpreted as audio_input_flags_t.
+    // TODO: merge audio_output_flags_t and audio_input_flags_t.
+    bool isCompatibleProfile(audio_devices_t device,
+                             String8 address,
+                             uint32_t samplingRate,
+                             uint32_t *updatedSamplingRate,
+                             audio_format_t format,
+                             audio_channel_mask_t channelMask,
+                             uint32_t flags) const;
+
+    void dump(int fd);
+    void log();
+
+    DeviceVector  mSupportedDevices; // supported devices
+                                     // (devices this output can be routed to)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
new file mode 100644
index 0000000..441bf7b
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+#pragma once
+
+/////////////////////////////////////////////////
+//      Definitions for audio policy configuration file (audio_policy.conf)
+/////////////////////////////////////////////////
+
+#define AUDIO_HARDWARE_MODULE_ID_MAX_LEN 32
+
+#define AUDIO_POLICY_CONFIG_FILE "/system/etc/audio_policy.conf"
+#define AUDIO_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_policy.conf"
+
+// global configuration
+#define GLOBAL_CONFIG_TAG "global_configuration"
+
+#define ATTACHED_OUTPUT_DEVICES_TAG "attached_output_devices"
+#define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device"
+#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices"
+#define SPEAKER_DRC_ENABLED_TAG "speaker_drc_enabled"
+#define AUDIO_HAL_VERSION_TAG "audio_hal_version"
+
+// hw modules descriptions
+#define AUDIO_HW_MODULE_TAG "audio_hw_modules"
+
+#define OUTPUTS_TAG "outputs"
+#define INPUTS_TAG "inputs"
+
+#define SAMPLING_RATES_TAG "sampling_rates"
+#define FORMATS_TAG "formats"
+#define CHANNELS_TAG "channel_masks"
+#define DEVICES_TAG "devices"
+#define FLAGS_TAG "flags"
+
+#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and
+                                    // "formats" in outputs descriptors indicating that supported
+                                    // values should be queried after opening the output.
+
+#define DEVICES_TAG "devices"
+#define DEVICE_TYPE "type"
+#define DEVICE_ADDRESS "address"
+
+#define MIXERS_TAG "mixers"
+#define MIXER_TYPE "type"
+#define MIXER_TYPE_MUX "mux"
+#define MIXER_TYPE_MIX "mix"
+
+#define GAINS_TAG "gains"
+#define GAIN_MODE "mode"
+#define GAIN_CHANNELS "channel_mask"
+#define GAIN_MIN_VALUE "min_value_mB"
+#define GAIN_MAX_VALUE "max_value_mB"
+#define GAIN_DEFAULT_VALUE "default_value_mB"
+#define GAIN_STEP_VALUE "step_value_mB"
+#define GAIN_MIN_RAMP_MS "min_ramp_ms"
+#define GAIN_MAX_RAMP_MS "max_ramp_ms"
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
new file mode 100644
index 0000000..ebd3548
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
@@ -0,0 +1,448 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioGain"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "AudioGain.h"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <math.h>
+
+namespace android {
+
+const VolumeCurvePoint
+ApmGains::sDefaultVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+
+const VolumeCurvePoint
+ApmGains::sDefaultMediaVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sExtMediaSystemVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT] = {
+    {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT] = {
+    {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT] = {
+    {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sHeadsetSystemVolumeCurve[ApmGains::VOLCNT] = {
+    {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultVoiceVolumeCurve[ApmGains::VOLCNT] = {
+    {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT] = {
+    {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sLinearVolumeCurve[ApmGains::VOLCNT] = {
+    {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSilentVolumeCurve[ApmGains::VOLCNT] = {
+    {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sFullScaleVolumeCurve[ApmGains::VOLCNT] = {
+    {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint *ApmGains::sVolumeProfiles[AUDIO_STREAM_CNT]
+                                                  [ApmGains::DEVICE_CATEGORY_CNT] = {
+    { // AUDIO_STREAM_VOICE_CALL
+        ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_SYSTEM
+        ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultSystemVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_RING
+        ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_MUSIC
+        ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_ALARM
+        ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_NOTIFICATION
+        ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultVolumeCurve,  // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_BLUETOOTH_SCO
+        ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_ENFORCED_AUDIBLE
+        ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    {  // AUDIO_STREAM_DTMF
+        ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sExtMediaSystemVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_TTS
+      // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
+        ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sSilentVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_ACCESSIBILITY
+        ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_REROUTING
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sFullScaleVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+    { // AUDIO_STREAM_PATCH
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+        ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+        ApmGains::sFullScaleVolumeCurve  // DEVICE_CATEGORY_EXT_MEDIA
+    },
+};
+
+//static
+audio_devices_t ApmGains::getDeviceForVolume(audio_devices_t device)
+{
+    if (device == AUDIO_DEVICE_NONE) {
+        // this happens when forcing a route update and no track is active on an output.
+        // In this case the returned category is not important.
+        device =  AUDIO_DEVICE_OUT_SPEAKER;
+    } else if (popcount(device) > 1) {
+        // Multiple device selection is either:
+        //  - speaker + one other device: give priority to speaker in this case.
+        //  - one A2DP device + another device: happens with duplicated output. In this case
+        // retain the device on the A2DP output as the other must not correspond to an active
+        // selection if not the speaker.
+        //  - HDMI-CEC system audio mode only output: give priority to available item in order.
+        if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+            device = AUDIO_DEVICE_OUT_SPEAKER;
+        } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
+            device = AUDIO_DEVICE_OUT_HDMI_ARC;
+        } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
+            device = AUDIO_DEVICE_OUT_AUX_LINE;
+        } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
+            device = AUDIO_DEVICE_OUT_SPDIF;
+        } else {
+            device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+        }
+    }
+
+    /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
+    if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
+        device = AUDIO_DEVICE_OUT_SPEAKER;
+
+    ALOGW_IF(popcount(device) != 1,
+            "getDeviceForVolume() invalid device combination: %08x",
+            device);
+
+    return device;
+}
+
+//static
+ApmGains::device_category ApmGains::getDeviceCategory(audio_devices_t device)
+{
+    switch(getDeviceForVolume(device)) {
+        case AUDIO_DEVICE_OUT_EARPIECE:
+            return ApmGains::DEVICE_CATEGORY_EARPIECE;
+        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+            return ApmGains::DEVICE_CATEGORY_HEADSET;
+        case AUDIO_DEVICE_OUT_LINE:
+        case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+        /*USB?  Remote submix?*/
+            return ApmGains::DEVICE_CATEGORY_EXT_MEDIA;
+        case AUDIO_DEVICE_OUT_SPEAKER:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+        case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+        case AUDIO_DEVICE_OUT_USB_DEVICE:
+        case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+        default:
+            return ApmGains::DEVICE_CATEGORY_SPEAKER;
+    }
+}
+
+//static
+float ApmGains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+        int indexInUi)
+{
+    ApmGains::device_category deviceCategory = ApmGains::getDeviceCategory(device);
+    const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
+
+    // the volume index in the UI is relative to the min and max volume indices for this stream type
+    int nbSteps = 1 + curve[ApmGains::VOLMAX].mIndex -
+            curve[ApmGains::VOLMIN].mIndex;
+    int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
+            (streamDesc.mIndexMax - streamDesc.mIndexMin);
+
+    // find what part of the curve this index volume belongs to, or if it's out of bounds
+    int segment = 0;
+    if (volIdx < curve[ApmGains::VOLMIN].mIndex) {         // out of bounds
+        return 0.0f;
+    } else if (volIdx < curve[ApmGains::VOLKNEE1].mIndex) {
+        segment = 0;
+    } else if (volIdx < curve[ApmGains::VOLKNEE2].mIndex) {
+        segment = 1;
+    } else if (volIdx <= curve[ApmGains::VOLMAX].mIndex) {
+        segment = 2;
+    } else {                                                               // out of bounds
+        return 1.0f;
+    }
+
+    // linear interpolation in the attenuation table in dB
+    float decibels = curve[segment].mDBAttenuation +
+            ((float)(volIdx - curve[segment].mIndex)) *
+                ( (curve[segment+1].mDBAttenuation -
+                        curve[segment].mDBAttenuation) /
+                    ((float)(curve[segment+1].mIndex -
+                            curve[segment].mIndex)) );
+
+    float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+    ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+            curve[segment].mIndex, volIdx,
+            curve[segment+1].mIndex,
+            curve[segment].mDBAttenuation,
+            decibels,
+            curve[segment+1].mDBAttenuation,
+            amplification);
+
+    return amplification;
+}
+
+
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+    mIndex = index;
+    mUseInChannelMask = useInChannelMask;
+    memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+    config->index = mIndex;
+    config->mode = mGain.mode;
+    config->channel_mask = mGain.channel_mask;
+    if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+        config->values[0] = mGain.default_value;
+    } else {
+        uint32_t numValues;
+        if (mUseInChannelMask) {
+            numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+        } else {
+            numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+        }
+        for (size_t i = 0; i < numValues; i++) {
+            config->values[i] = mGain.default_value;
+        }
+    }
+    if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+        config->ramp_duration_ms = mGain.min_ramp_ms;
+    }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+    if ((config->mode & ~mGain.mode) != 0) {
+        return BAD_VALUE;
+    }
+    if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+        if ((config->values[0] < mGain.min_value) ||
+                    (config->values[0] > mGain.max_value)) {
+            return BAD_VALUE;
+        }
+    } else {
+        if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+            return BAD_VALUE;
+        }
+        uint32_t numValues;
+        if (mUseInChannelMask) {
+            numValues = audio_channel_count_from_in_mask(config->channel_mask);
+        } else {
+            numValues = audio_channel_count_from_out_mask(config->channel_mask);
+        }
+        for (size_t i = 0; i < numValues; i++) {
+            if ((config->values[i] < mGain.min_value) ||
+                    (config->values[i] > mGain.max_value)) {
+                return BAD_VALUE;
+            }
+        }
+    }
+    if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+        if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+                    (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+            return BAD_VALUE;
+        }
+    }
+    return NO_ERROR;
+}
+
+void AudioGain::dump(int fd, int spaces, int index) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
+    result.append(buffer);
+
+    write(fd, result.string(), result.size());
+}
+
+
+// --- StreamDescriptor class implementation
+
+StreamDescriptor::StreamDescriptor()
+    :   mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+    mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int StreamDescriptor::getVolumeIndex(audio_devices_t device)
+{
+    device = ApmGains::getDeviceForVolume(device);
+    // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+    if (mIndexCur.indexOfKey(device) < 0) {
+        device = AUDIO_DEVICE_OUT_DEFAULT;
+    }
+    return mIndexCur.valueFor(device);
+}
+
+void StreamDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%s         %02d         %02d         ",
+             mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+    result.append(buffer);
+    for (size_t i = 0; i < mIndexCur.size(); i++) {
+        snprintf(buffer, SIZE, "%04x : %02d, ",
+                 mIndexCur.keyAt(i),
+                 mIndexCur.valueAt(i));
+        result.append(buffer);
+    }
+    result.append("\n");
+
+    write(fd, result.string(), result.size());
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
new file mode 100644
index 0000000..fa7761a
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -0,0 +1,110 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioInputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioInputDescriptor.h"
+#include "IOProfile.h"
+#include "AudioGain.h"
+#include "HwModule.h"
+#include <media/AudioPolicy.h>
+
+namespace android {
+
+AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
+    : mId(0), mIoHandle(0),
+      mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
+      mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
+{
+    if (profile != NULL) {
+        mSamplingRate = profile->pickSamplingRate();
+        mFormat = profile->pickFormat();
+        mChannelMask = profile->pickChannelMask();
+        if (profile->mGains.size() > 0) {
+            profile->mGains[0]->getDefaultConfig(&mGain);
+        }
+    }
+}
+
+void AudioInputDescriptor::setIoHandle(audio_io_handle_t ioHandle)
+{
+    mId = AudioPort::getNextUniqueId();
+    mIoHandle = ioHandle;
+}
+
+void AudioInputDescriptor::toAudioPortConfig(
+                                                   struct audio_port_config *dstConfig,
+                                                   const struct audio_port_config *srcConfig) const
+{
+    ALOG_ASSERT(mProfile != 0,
+                "toAudioPortConfig() called on input with null profile %d", mIoHandle);
+    dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+                            AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+    if (srcConfig != NULL) {
+        dstConfig->config_mask |= srcConfig->config_mask;
+    }
+
+    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+    dstConfig->id = mId;
+    dstConfig->role = AUDIO_PORT_ROLE_SINK;
+    dstConfig->type = AUDIO_PORT_TYPE_MIX;
+    dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+    dstConfig->ext.mix.handle = mIoHandle;
+    dstConfig->ext.mix.usecase.source = mInputSource;
+}
+
+void AudioInputDescriptor::toAudioPort(
+                                                    struct audio_port *port) const
+{
+    ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
+
+    mProfile->toAudioPort(port);
+    port->id = mId;
+    toAudioPortConfig(&port->active_config);
+    port->ext.mix.hw_module = mProfile->mModule->mHandle;
+    port->ext.mix.handle = mIoHandle;
+    port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioInputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " ID: %d\n", mId);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
+    result.append(buffer);
+
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
new file mode 100644
index 0000000..4c5509d
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -0,0 +1,223 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioOutputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioOutputDescriptor.h"
+#include "IOProfile.h"
+#include "AudioGain.h"
+#include "HwModule.h"
+#include <media/AudioPolicy.h>
+
+namespace android {
+
+AudioOutputDescriptor::AudioOutputDescriptor(
+        const sp<IOProfile>& profile)
+    : mId(0), mIoHandle(0), mLatency(0),
+    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
+    mPatchHandle(0),
+    mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+    // clear usage count for all stream types
+    for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+        mRefCount[i] = 0;
+        mCurVolume[i] = -1.0;
+        mMuteCount[i] = 0;
+        mStopTime[i] = 0;
+    }
+    for (int i = 0; i < NUM_STRATEGIES; i++) {
+        mStrategyMutedByDevice[i] = false;
+    }
+    if (profile != NULL) {
+        mFlags = (audio_output_flags_t)profile->mFlags;
+        mSamplingRate = profile->pickSamplingRate();
+        mFormat = profile->pickFormat();
+        mChannelMask = profile->pickChannelMask();
+        if (profile->mGains.size() > 0) {
+            profile->mGains[0]->getDefaultConfig(&mGain);
+        }
+    }
+}
+
+audio_devices_t AudioOutputDescriptor::device() const
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+    } else {
+        return mDevice;
+    }
+}
+
+void AudioOutputDescriptor::setIoHandle(audio_io_handle_t ioHandle)
+{
+    mId = AudioPort::getNextUniqueId();
+    mIoHandle = ioHandle;
+}
+
+uint32_t AudioOutputDescriptor::latency()
+{
+    if (isDuplicated()) {
+        return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+    } else {
+        return mLatency;
+    }
+}
+
+bool AudioOutputDescriptor::sharesHwModuleWith(
+        const sp<AudioOutputDescriptor> outputDesc)
+{
+    if (isDuplicated()) {
+        return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+    } else if (outputDesc->isDuplicated()){
+        return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+    } else {
+        return (mProfile->mModule == outputDesc->mProfile->mModule);
+    }
+}
+
+void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+                                                                   int delta)
+{
+    // forward usage count change to attached outputs
+    if (isDuplicated()) {
+        mOutput1->changeRefCount(stream, delta);
+        mOutput2->changeRefCount(stream, delta);
+    }
+    if ((delta + (int)mRefCount[stream]) < 0) {
+        ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
+              delta, stream, mRefCount[stream]);
+        mRefCount[stream] = 0;
+        return;
+    }
+    mRefCount[stream] += delta;
+    ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioOutputDescriptor::supportedDevices()
+{
+    if (isDuplicated()) {
+        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+    } else {
+        return mProfile->mSupportedDevices.types() ;
+    }
+}
+
+bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+    nsecs_t sysTime = 0;
+    if (inPastMs != 0) {
+        sysTime = systemTime();
+    }
+    for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+        if (i == AUDIO_STREAM_PATCH) {
+            continue;
+        }
+        if (isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
+                                                                       uint32_t inPastMs,
+                                                                       nsecs_t sysTime) const
+{
+    if (mRefCount[stream] != 0) {
+        return true;
+    }
+    if (inPastMs == 0) {
+        return false;
+    }
+    if (sysTime == 0) {
+        sysTime = systemTime();
+    }
+    if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+        return true;
+    }
+    return false;
+}
+
+void AudioOutputDescriptor::toAudioPortConfig(
+                                                 struct audio_port_config *dstConfig,
+                                                 const struct audio_port_config *srcConfig) const
+{
+    ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
+
+    dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+                            AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+    if (srcConfig != NULL) {
+        dstConfig->config_mask |= srcConfig->config_mask;
+    }
+    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+    dstConfig->id = mId;
+    dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
+    dstConfig->type = AUDIO_PORT_TYPE_MIX;
+    dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+    dstConfig->ext.mix.handle = mIoHandle;
+    dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
+}
+
+void AudioOutputDescriptor::toAudioPort(
+                                                    struct audio_port *port) const
+{
+    ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
+    mProfile->toAudioPort(port);
+    port->id = mId;
+    toAudioPortConfig(&port->active_config);
+    port->ext.mix.hw_module = mProfile->mModule->mHandle;
+    port->ext.mix.handle = mIoHandle;
+    port->ext.mix.latency_class =
+            mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioOutputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " ID: %d\n", mId);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", device());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+    result.append(buffer);
+    for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+        snprintf(buffer, SIZE, " %02d     %.03f     %02d       %02d\n",
+                 i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+        result.append(buffer);
+    }
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+
+
+}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
new file mode 100644
index 0000000..89e7f81
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioPatch"
+//#define LOG_NDEBUG 0
+
+#include "AudioPatch.h"
+#include "ConfigParsingUtils.h"
+#include <cutils/log.h>
+#include <utils/String8.h>
+
+namespace android {
+
+int32_t volatile AudioPatch::mNextUniqueId = 1;
+
+AudioPatch::AudioPatch(const struct audio_patch *patch, uid_t uid) :
+    mHandle(static_cast<audio_patch_handle_t>(android_atomic_inc(&mNextUniqueId))),
+    mPatch(*patch),
+    mUid(uid),
+    mAfPatchHandle(0)
+{
+}
+
+status_t AudioPatch::dump(int fd, int spaces, int index) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
+    result.append(buffer);
+    for (size_t i = 0; i < mPatch.num_sources; i++) {
+        if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
+            snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+                     mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+                                                        ARRAY_SIZE(sDeviceNameToEnumTable),
+                                                        mPatch.sources[i].ext.device.type));
+        } else {
+            snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+                     mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
+        }
+        result.append(buffer);
+    }
+    snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
+    result.append(buffer);
+    for (size_t i = 0; i < mPatch.num_sinks; i++) {
+        if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
+            snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+                     mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+                                                        ARRAY_SIZE(sDeviceNameToEnumTable),
+                                                        mPatch.sinks[i].ext.device.type));
+        } else {
+            snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+                     mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
+        }
+        result.append(buffer);
+    }
+
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
new file mode 100644
index 0000000..d34aac1
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -0,0 +1,798 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioPort"
+//#define LOG_NDEBUG 0
+
+#include "AudioPort.h"
+#include "HwModule.h"
+#include "AudioGain.h"
+#include "ConfigParsingUtils.h"
+#include "audio_policy_conf.h"
+
+namespace android {
+
+int32_t volatile AudioPort::mNextUniqueId = 1;
+
+// --- AudioPort class implementation
+
+AudioPort::AudioPort(const String8& name, audio_port_type_t type,
+          audio_port_role_t role, const sp<HwModule>& module) :
+    mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0)
+{
+    mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
+                    ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
+}
+
+void AudioPort::attach(const sp<HwModule>& module)
+{
+    mId = getNextUniqueId();
+    mModule = module;
+}
+
+audio_port_handle_t AudioPort::getNextUniqueId()
+{
+    return static_cast<audio_port_handle_t>(android_atomic_inc(&mNextUniqueId));
+}
+
+void AudioPort::toAudioPort(struct audio_port *port) const
+{
+    port->role = mRole;
+    port->type = mType;
+    strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
+    unsigned int i;
+    for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
+        if (mSamplingRates[i] != 0) {
+            port->sample_rates[i] = mSamplingRates[i];
+        }
+    }
+    port->num_sample_rates = i;
+    for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
+        if (mChannelMasks[i] != 0) {
+            port->channel_masks[i] = mChannelMasks[i];
+        }
+    }
+    port->num_channel_masks = i;
+    for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
+        if (mFormats[i] != 0) {
+            port->formats[i] = mFormats[i];
+        }
+    }
+    port->num_formats = i;
+
+    ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+
+    for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
+        port->gains[i] = mGains[i]->mGain;
+    }
+    port->num_gains = i;
+}
+
+void AudioPort::importAudioPort(const sp<AudioPort> port) {
+    for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
+        const uint32_t rate = port->mSamplingRates.itemAt(k);
+        if (rate != 0) { // skip "dynamic" rates
+            bool hasRate = false;
+            for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
+                if (rate == mSamplingRates.itemAt(l)) {
+                    hasRate = true;
+                    break;
+                }
+            }
+            if (!hasRate) { // never import a sampling rate twice
+                mSamplingRates.add(rate);
+            }
+        }
+    }
+    for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
+        const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
+        if (mask != 0) { // skip "dynamic" masks
+            bool hasMask = false;
+            for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
+                if (mask == mChannelMasks.itemAt(l)) {
+                    hasMask = true;
+                    break;
+                }
+            }
+            if (!hasMask) { // never import a channel mask twice
+                mChannelMasks.add(mask);
+            }
+        }
+    }
+    for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
+        const audio_format_t format = port->mFormats.itemAt(k);
+        if (format != 0) { // skip "dynamic" formats
+            bool hasFormat = false;
+            for (size_t l = 0 ; l < mFormats.size() ; l++) {
+                if (format == mFormats.itemAt(l)) {
+                    hasFormat = true;
+                    break;
+                }
+            }
+            if (!hasFormat) { // never import a channel mask twice
+                mFormats.add(format);
+            }
+        }
+    }
+    for (size_t k = 0 ; k < port->mGains.size() ; k++) {
+        sp<AudioGain> gain = port->mGains.itemAt(k);
+        if (gain != 0) {
+            bool hasGain = false;
+            for (size_t l = 0 ; l < mGains.size() ; l++) {
+                if (gain == mGains.itemAt(l)) {
+                    hasGain = true;
+                    break;
+                }
+            }
+            if (!hasGain) { // never import a gain twice
+                mGains.add(gain);
+            }
+        }
+    }
+}
+
+void AudioPort::clearCapabilities() {
+    mChannelMasks.clear();
+    mFormats.clear();
+    mSamplingRates.clear();
+    mGains.clear();
+}
+
+void 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, "|");
+    }
+}
+
+void 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)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable,
+                                                             ARRAY_SIZE(sFormatNameToEnumTable),
+                                                             str);
+        if (format != AUDIO_FORMAT_DEFAULT) {
+            mFormats.add(format);
+        }
+        str = strtok(NULL, "|");
+    }
+}
+
+void 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)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sInChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+            mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+}
+
+void 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)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+                                                   ARRAY_SIZE(sOutChannelsNameToEnumTable),
+                                                   str);
+        if (channelMask != 0) {
+            mChannelMasks.add(channelMask);
+        }
+        str = strtok(NULL, "|");
+    }
+    return;
+}
+
+audio_gain_mode_t AudioPort::loadGainMode(char *name)
+{
+    const char *str = strtok(name, "|");
+
+    ALOGV("loadGainMode() %s", name);
+    audio_gain_mode_t mode = 0;
+    while (str != NULL) {
+        mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable,
+                                                ARRAY_SIZE(sGainModeNameToEnumTable),
+                                                str);
+        str = strtok(NULL, "|");
+    }
+    return mode;
+}
+
+void AudioPort::loadGain(cnode *root, int index)
+{
+    cnode *node = root->first_child;
+
+    sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
+
+    while (node) {
+        if (strcmp(node->name, GAIN_MODE) == 0) {
+            gain->mGain.mode = loadGainMode((char *)node->value);
+        } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
+            if (mUseInChannelMask) {
+                gain->mGain.channel_mask =
+                        (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+                                                           ARRAY_SIZE(sInChannelsNameToEnumTable),
+                                                           (char *)node->value);
+            } else {
+                gain->mGain.channel_mask =
+                        (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+                                                           ARRAY_SIZE(sOutChannelsNameToEnumTable),
+                                                           (char *)node->value);
+            }
+        } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
+            gain->mGain.min_value = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
+            gain->mGain.max_value = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
+            gain->mGain.default_value = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
+            gain->mGain.step_value = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
+            gain->mGain.min_ramp_ms = atoi((char *)node->value);
+        } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
+            gain->mGain.max_ramp_ms = atoi((char *)node->value);
+        }
+        node = node->next;
+    }
+
+    ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
+          gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
+
+    if (gain->mGain.mode == 0) {
+        return;
+    }
+    mGains.add(gain);
+}
+
+void AudioPort::loadGains(cnode *root)
+{
+    cnode *node = root->first_child;
+    int index = 0;
+    while (node) {
+        ALOGV("loadGains() loading gain %s", node->name);
+        loadGain(node, index++);
+        node = node->next;
+    }
+}
+
+status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
+{
+    if (mSamplingRates.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+        if (mSamplingRates[i] == samplingRate) {
+            return NO_ERROR;
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
+        uint32_t *updatedSamplingRate) const
+{
+    if (mSamplingRates.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    // Search for the closest supported sampling rate that is above (preferred)
+    // or below (acceptable) the desired sampling rate, within a permitted ratio.
+    // The sampling rates do not need to be sorted in ascending order.
+    ssize_t maxBelow = -1;
+    ssize_t minAbove = -1;
+    uint32_t candidate;
+    for (size_t i = 0; i < mSamplingRates.size(); i++) {
+        candidate = mSamplingRates[i];
+        if (candidate == samplingRate) {
+            if (updatedSamplingRate != NULL) {
+                *updatedSamplingRate = candidate;
+            }
+            return NO_ERROR;
+        }
+        // candidate < desired
+        if (candidate < samplingRate) {
+            if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
+                maxBelow = i;
+            }
+        // candidate > desired
+        } else {
+            if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
+                minAbove = i;
+            }
+        }
+    }
+    // This uses hard-coded knowledge about AudioFlinger resampling ratios.
+    // TODO Move these assumptions out.
+    static const uint32_t kMaxDownSampleRatio = 6;  // beyond this aliasing occurs
+    static const uint32_t kMaxUpSampleRatio = 256;  // beyond this sample rate inaccuracies occur
+                                                    // due to approximation by an int32_t of the
+                                                    // phase increments
+    // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
+    if (minAbove >= 0) {
+        candidate = mSamplingRates[minAbove];
+        if (candidate / kMaxDownSampleRatio <= samplingRate) {
+            if (updatedSamplingRate != NULL) {
+                *updatedSamplingRate = candidate;
+            }
+            return NO_ERROR;
+        }
+    }
+    // But if we have to up-sample from a lower sampling rate, that's OK.
+    if (maxBelow >= 0) {
+        candidate = mSamplingRates[maxBelow];
+        if (candidate * kMaxUpSampleRatio >= samplingRate) {
+            if (updatedSamplingRate != NULL) {
+                *updatedSamplingRate = candidate;
+            }
+            return NO_ERROR;
+        }
+    }
+    // leave updatedSamplingRate unmodified
+    return BAD_VALUE;
+}
+
+status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
+{
+    if (mChannelMasks.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    for (size_t i = 0; i < mChannelMasks.size(); i++) {
+        if (mChannelMasks[i] == channelMask) {
+            return NO_ERROR;
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
+        const
+{
+    if (mChannelMasks.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+    for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+        // FIXME Does not handle multi-channel automatic conversions yet
+        audio_channel_mask_t supported = mChannelMasks[i];
+        if (supported == channelMask) {
+            return NO_ERROR;
+        }
+        if (isRecordThread) {
+            // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
+            // FIXME Abstract this out to a table.
+            if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
+                    && channelMask == AUDIO_CHANNEL_IN_MONO) ||
+                (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
+                    || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
+                return NO_ERROR;
+            }
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t AudioPort::checkFormat(audio_format_t format) const
+{
+    if (mFormats.isEmpty()) {
+        return NO_ERROR;
+    }
+
+    for (size_t i = 0; i < mFormats.size(); i ++) {
+        if (mFormats[i] == format) {
+            return NO_ERROR;
+        }
+    }
+    return BAD_VALUE;
+}
+
+
+uint32_t AudioPort::pickSamplingRate() const
+{
+    // special case for uninitialized dynamic profile
+    if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
+        return 0;
+    }
+
+    // For direct outputs, pick minimum sampling rate: this helps ensuring that the
+    // channel count / sampling rate combination chosen will be supported by the connected
+    // sink
+    if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+            (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+        uint32_t samplingRate = UINT_MAX;
+        for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+            if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
+                samplingRate = mSamplingRates[i];
+            }
+        }
+        return (samplingRate == UINT_MAX) ? 0 : samplingRate;
+    }
+
+    uint32_t samplingRate = 0;
+    uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
+
+    // For mixed output and inputs, use max mixer sampling rates. Do not
+    // limit sampling rate otherwise
+    if (mType != AUDIO_PORT_TYPE_MIX) {
+        maxRate = UINT_MAX;
+    }
+    for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+        if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
+            samplingRate = mSamplingRates[i];
+        }
+    }
+    return samplingRate;
+}
+
+audio_channel_mask_t AudioPort::pickChannelMask() const
+{
+    // special case for uninitialized dynamic profile
+    if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
+        return AUDIO_CHANNEL_NONE;
+    }
+    audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
+
+    // For direct outputs, pick minimum channel count: this helps ensuring that the
+    // channel count / sampling rate combination chosen will be supported by the connected
+    // sink
+    if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+            (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+        uint32_t channelCount = UINT_MAX;
+        for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+            uint32_t cnlCount;
+            if (mUseInChannelMask) {
+                cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+            } else {
+                cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+            }
+            if ((cnlCount < channelCount) && (cnlCount > 0)) {
+                channelMask = mChannelMasks[i];
+                channelCount = cnlCount;
+            }
+        }
+        return channelMask;
+    }
+
+    uint32_t channelCount = 0;
+    uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
+
+    // For mixed output and inputs, use max mixer channel count. Do not
+    // limit channel count otherwise
+    if (mType != AUDIO_PORT_TYPE_MIX) {
+        maxCount = UINT_MAX;
+    }
+    for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+        uint32_t cnlCount;
+        if (mUseInChannelMask) {
+            cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+        } else {
+            cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+        }
+        if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+            channelMask = mChannelMasks[i];
+            channelCount = cnlCount;
+        }
+    }
+    return channelMask;
+}
+
+/* format in order of increasing preference */
+const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
+        AUDIO_FORMAT_DEFAULT,
+        AUDIO_FORMAT_PCM_16_BIT,
+        AUDIO_FORMAT_PCM_8_24_BIT,
+        AUDIO_FORMAT_PCM_24_BIT_PACKED,
+        AUDIO_FORMAT_PCM_32_BIT,
+        AUDIO_FORMAT_PCM_FLOAT,
+};
+
+int AudioPort::compareFormats(audio_format_t format1,
+                                                  audio_format_t format2)
+{
+    // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
+    // compressed format and better than any PCM format. This is by design of pickFormat()
+    if (!audio_is_linear_pcm(format1)) {
+        if (!audio_is_linear_pcm(format2)) {
+            return 0;
+        }
+        return 1;
+    }
+    if (!audio_is_linear_pcm(format2)) {
+        return -1;
+    }
+
+    int index1 = -1, index2 = -1;
+    for (size_t i = 0;
+            (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
+            i ++) {
+        if (sPcmFormatCompareTable[i] == format1) {
+            index1 = i;
+        }
+        if (sPcmFormatCompareTable[i] == format2) {
+            index2 = i;
+        }
+    }
+    // format1 not found => index1 < 0 => format2 > format1
+    // format2 not found => index2 < 0 => format2 < format1
+    return index1 - index2;
+}
+
+audio_format_t AudioPort::pickFormat() const
+{
+    // special case for uninitialized dynamic profile
+    if (mFormats.size() == 1 && mFormats[0] == 0) {
+        return AUDIO_FORMAT_DEFAULT;
+    }
+
+    audio_format_t format = AUDIO_FORMAT_DEFAULT;
+    audio_format_t bestFormat =
+            AudioPort::sPcmFormatCompareTable[
+                ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1];
+    // For mixed output and inputs, use best mixer output format. Do not
+    // limit format otherwise
+    if ((mType != AUDIO_PORT_TYPE_MIX) ||
+            ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+             (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
+        bestFormat = AUDIO_FORMAT_INVALID;
+    }
+
+    for (size_t i = 0; i < mFormats.size(); i ++) {
+        if ((compareFormats(mFormats[i], format) > 0) &&
+                (compareFormats(mFormats[i], bestFormat) <= 0)) {
+            format = mFormats[i];
+        }
+    }
+    return format;
+}
+
+status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig,
+                                                  int index) const
+{
+    if (index < 0 || (size_t)index >= mGains.size()) {
+        return BAD_VALUE;
+    }
+    return mGains[index]->checkConfig(gainConfig);
+}
+
+void AudioPort::dump(int fd, int spaces) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    if (mName.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
+        result.append(buffer);
+    }
+
+    if (mSamplingRates.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
+        result.append(buffer);
+        for (size_t i = 0; i < mSamplingRates.size(); i++) {
+            if (i == 0 && mSamplingRates[i] == 0) {
+                snprintf(buffer, SIZE, "Dynamic");
+            } else {
+                snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+            }
+            result.append(buffer);
+            result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
+        }
+        result.append("\n");
+    }
+
+    if (mChannelMasks.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
+        result.append(buffer);
+        for (size_t i = 0; i < mChannelMasks.size(); i++) {
+            ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
+
+            if (i == 0 && mChannelMasks[i] == 0) {
+                snprintf(buffer, SIZE, "Dynamic");
+            } else {
+                snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+            }
+            result.append(buffer);
+            result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
+        }
+        result.append("\n");
+    }
+
+    if (mFormats.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
+        result.append(buffer);
+        for (size_t i = 0; i < mFormats.size(); i++) {
+            const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable,
+                                                 ARRAY_SIZE(sFormatNameToEnumTable),
+                                                 mFormats[i]);
+            if (i == 0 && strcmp(formatStr, "") == 0) {
+                snprintf(buffer, SIZE, "Dynamic");
+            } else {
+                snprintf(buffer, SIZE, "%s", formatStr);
+            }
+            result.append(buffer);
+            result.append(i == (mFormats.size() - 1) ? "" : ", ");
+        }
+        result.append("\n");
+    }
+    write(fd, result.string(), result.size());
+    if (mGains.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
+        write(fd, buffer, strlen(buffer) + 1);
+        result.append(buffer);
+        for (size_t i = 0; i < mGains.size(); i++) {
+            mGains[i]->dump(fd, spaces + 2, i);
+        }
+    }
+}
+
+
+// --- AudioPortConfig class implementation
+
+AudioPortConfig::AudioPortConfig()
+{
+    mSamplingRate = 0;
+    mChannelMask = AUDIO_CHANNEL_NONE;
+    mFormat = AUDIO_FORMAT_INVALID;
+    mGain.index = -1;
+}
+
+status_t AudioPortConfig::applyAudioPortConfig(
+                                                        const struct audio_port_config *config,
+                                                        struct audio_port_config *backupConfig)
+{
+    struct audio_port_config localBackupConfig;
+    status_t status = NO_ERROR;
+
+    localBackupConfig.config_mask = config->config_mask;
+    toAudioPortConfig(&localBackupConfig);
+
+    sp<AudioPort> audioport = getAudioPort();
+    if (audioport == 0) {
+        status = NO_INIT;
+        goto exit;
+    }
+    if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+        status = audioport->checkExactSamplingRate(config->sample_rate);
+        if (status != NO_ERROR) {
+            goto exit;
+        }
+        mSamplingRate = config->sample_rate;
+    }
+    if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+        status = audioport->checkExactChannelMask(config->channel_mask);
+        if (status != NO_ERROR) {
+            goto exit;
+        }
+        mChannelMask = config->channel_mask;
+    }
+    if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+        status = audioport->checkFormat(config->format);
+        if (status != NO_ERROR) {
+            goto exit;
+        }
+        mFormat = config->format;
+    }
+    if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+        status = audioport->checkGain(&config->gain, config->gain.index);
+        if (status != NO_ERROR) {
+            goto exit;
+        }
+        mGain = config->gain;
+    }
+
+exit:
+    if (status != NO_ERROR) {
+        applyAudioPortConfig(&localBackupConfig);
+    }
+    if (backupConfig != NULL) {
+        *backupConfig = localBackupConfig;
+    }
+    return status;
+}
+
+void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
+                                        const struct audio_port_config *srcConfig) const
+{
+    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+        dstConfig->sample_rate = mSamplingRate;
+        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
+            dstConfig->sample_rate = srcConfig->sample_rate;
+        }
+    } else {
+        dstConfig->sample_rate = 0;
+    }
+    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+        dstConfig->channel_mask = mChannelMask;
+        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
+            dstConfig->channel_mask = srcConfig->channel_mask;
+        }
+    } else {
+        dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
+    }
+    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+        dstConfig->format = mFormat;
+        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
+            dstConfig->format = srcConfig->format;
+        }
+    } else {
+        dstConfig->format = AUDIO_FORMAT_INVALID;
+    }
+    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+        dstConfig->gain = mGain;
+        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
+            dstConfig->gain = srcConfig->gain;
+        }
+    } else {
+        dstConfig->gain.index = -1;
+    }
+    if (dstConfig->gain.index != -1) {
+        dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
+    } else {
+        dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+    }
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
new file mode 100644
index 0000000..300f35a
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
@@ -0,0 +1,122 @@
+/*
+ * Copyright (C) 2015 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 "APM::ConfigParsingUtils"
+//#define LOG_NDEBUG 0
+
+#include "ConfigParsingUtils.h"
+#include <utils/Log.h>
+
+namespace android {
+
+//static
+uint32_t ConfigParsingUtils::stringToEnum(const struct StringToEnum *table,
+                                              size_t size,
+                                              const char *name)
+{
+    for (size_t i = 0; i < size; i++) {
+        if (strcmp(table[i].name, name) == 0) {
+            ALOGV("stringToEnum() found %s", table[i].name);
+            return table[i].value;
+        }
+    }
+    return 0;
+}
+
+//static
+const char *ConfigParsingUtils::enumToString(const struct StringToEnum *table,
+                                              size_t size,
+                                              uint32_t value)
+{
+    for (size_t i = 0; i < size; i++) {
+        if (table[i].value == value) {
+            return table[i].name;
+        }
+    }
+    return "";
+}
+
+//static
+bool ConfigParsingUtils::stringToBool(const char *value)
+{
+    return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+
+// --- audio_policy.conf file parsing
+//static
+uint32_t ConfigParsingUtils::parseOutputFlagNames(char *name)
+{
+    uint32_t flag = 0;
+
+    // it is OK to cast name to non const here as we are not going to use it after
+    // strtok() modifies it
+    char *flagName = strtok(name, "|");
+    while (flagName != NULL) {
+        if (strlen(flagName) != 0) {
+            flag |= ConfigParsingUtils::stringToEnum(sOutputFlagNameToEnumTable,
+                               ARRAY_SIZE(sOutputFlagNameToEnumTable),
+                               flagName);
+        }
+        flagName = strtok(NULL, "|");
+    }
+    //force direct flag if offload flag is set: offloading implies a direct output stream
+    // and all common behaviors are driven by checking only the direct flag
+    // this should normally be set appropriately in the policy configuration file
+    if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+    }
+
+    return flag;
+}
+
+//static
+uint32_t ConfigParsingUtils::parseInputFlagNames(char *name)
+{
+    uint32_t flag = 0;
+
+    // it is OK to cast name to non const here as we are not going to use it after
+    // strtok() modifies it
+    char *flagName = strtok(name, "|");
+    while (flagName != NULL) {
+        if (strlen(flagName) != 0) {
+            flag |= stringToEnum(sInputFlagNameToEnumTable,
+                               ARRAY_SIZE(sInputFlagNameToEnumTable),
+                               flagName);
+        }
+        flagName = strtok(NULL, "|");
+    }
+    return flag;
+}
+
+//static
+audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name)
+{
+    uint32_t device = 0;
+
+    char *devName = strtok(name, "|");
+    while (devName != NULL) {
+        if (strlen(devName) != 0) {
+            device |= stringToEnum(sDeviceNameToEnumTable,
+                                 ARRAY_SIZE(sDeviceNameToEnumTable),
+                                 devName);
+         }
+        devName = strtok(NULL, "|");
+     }
+    return device;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
new file mode 100644
index 0000000..7f098ab
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -0,0 +1,289 @@
+/*
+ * Copyright (C) 2015 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 "APM::Devices"
+//#define LOG_NDEBUG 0
+
+#include "DeviceDescriptor.h"
+#include "AudioGain.h"
+#include "HwModule.h"
+#include "ConfigParsingUtils.h"
+
+namespace android {
+
+String8 DeviceDescriptor::emptyNameStr = String8("");
+
+DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
+                     AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
+                               audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+                                                              AUDIO_PORT_ROLE_SOURCE,
+                             NULL),
+                     mDeviceType(type), mAddress("")
+{
+
+}
+
+bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
+{
+    // Devices are considered equal if they:
+    // - 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 (mDeviceType == other->mDeviceType) &&
+           (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
+           (mChannelMask == 0 || other->mChannelMask == 0 ||
+                mChannelMask == other->mChannelMask);
+}
+
+void DeviceDescriptor::loadGains(cnode *root)
+{
+    AudioPort::loadGains(root);
+    if (mGains.size() > 0) {
+        mGains[0]->getDefaultConfig(&mGain);
+    }
+}
+
+void DeviceVector::refreshTypes()
+{
+    mDeviceTypes = AUDIO_DEVICE_NONE;
+    for(size_t i = 0; i < size(); i++) {
+        mDeviceTypes |= itemAt(i)->mDeviceType;
+    }
+    ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
+}
+
+ssize_t DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
+{
+    for(size_t i = 0; i < size(); i++) {
+        if (item->equals(itemAt(i))) {
+            return i;
+        }
+    }
+    return -1;
+}
+
+ssize_t DeviceVector::add(const sp<DeviceDescriptor>& item)
+{
+    ssize_t ret = indexOf(item);
+
+    if (ret < 0) {
+        ret = SortedVector::add(item);
+        if (ret >= 0) {
+            refreshTypes();
+        }
+    } else {
+        ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
+        ret = -1;
+    }
+    return ret;
+}
+
+ssize_t DeviceVector::remove(const sp<DeviceDescriptor>& item)
+{
+    size_t i;
+    ssize_t ret = indexOf(item);
+
+    if (ret < 0) {
+        ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
+    } else {
+        ret = SortedVector::removeAt(ret);
+        if (ret >= 0) {
+            refreshTypes();
+        }
+    }
+    return ret;
+}
+
+void DeviceVector::loadDevicesFromType(audio_devices_t types)
+{
+    DeviceVector deviceList;
+
+    uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
+    types &= ~role_bit;
+
+    while (types) {
+        uint32_t i = 31 - __builtin_clz(types);
+        uint32_t type = 1 << i;
+        types &= ~type;
+        add(new DeviceDescriptor(String8("device_type"), type | role_bit));
+    }
+}
+
+void DeviceVector::loadDevicesFromName(char *name,
+                                       const DeviceVector& declaredDevices)
+{
+    char *devName = strtok(name, "|");
+    while (devName != NULL) {
+        if (strlen(devName) != 0) {
+            audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable,
+                                 ARRAY_SIZE(sDeviceNameToEnumTable),
+                                 devName);
+            if (type != AUDIO_DEVICE_NONE) {
+                sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type);
+                if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
+                        type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
+                    dev->mAddress = String8("0");
+                }
+                add(dev);
+            } else {
+                sp<DeviceDescriptor> deviceDesc =
+                        declaredDevices.getDeviceFromName(String8(devName));
+                if (deviceDesc != 0) {
+                    add(deviceDesc);
+                }
+            }
+         }
+         devName = strtok(NULL, "|");
+     }
+}
+
+sp<DeviceDescriptor> 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) {
+            if (address == "" || itemAt(i)->mAddress == address) {
+                device = itemAt(i);
+                if (itemAt(i)->mAddress == address) {
+                    break;
+                }
+            }
+        }
+    }
+    ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
+          type, address.string(), device.get());
+    return device;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const
+{
+    sp<DeviceDescriptor> device;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->getHandle() == id) {
+            device = itemAt(i);
+            break;
+        }
+    }
+    return device;
+}
+
+DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const
+{
+    DeviceVector devices;
+    bool isOutput = audio_is_output_devices(type);
+    type &= ~AUDIO_DEVICE_BIT_IN;
+    for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
+        bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType);
+        audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN;
+        if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
+            devices.add(itemAt(i));
+            type &= ~curType;
+            ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
+                  itemAt(i)->mDeviceType, itemAt(i).get());
+        }
+    }
+    return devices;
+}
+
+DeviceVector DeviceVector::getDevicesFromTypeAddr(
+        audio_devices_t type, String8 address) const
+{
+    DeviceVector devices;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->mDeviceType == type) {
+            if (itemAt(i)->mAddress == address) {
+                devices.add(itemAt(i));
+            }
+        }
+    }
+    return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromName(const String8& name) const
+{
+    sp<DeviceDescriptor> device;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->mName == name) {
+            device = itemAt(i);
+            break;
+        }
+    }
+    return device;
+}
+
+void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+                                         const struct audio_port_config *srcConfig) const
+{
+    dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
+    if (srcConfig != NULL) {
+        dstConfig->config_mask |= srcConfig->config_mask;
+    }
+
+    AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+    dstConfig->id = mId;
+    dstConfig->role = audio_is_output_device(mDeviceType) ?
+                        AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+    dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
+    dstConfig->ext.device.type = mDeviceType;
+
+    //TODO Understand why this test is necessary. i.e. why at boot time does it crash
+    // without the test?
+    // This has been demonstrated to NOT be true (at start up)
+    // ALOG_ASSERT(mModule != NULL);
+    dstConfig->ext.device.hw_module = mModule != 0 ? mModule->mHandle : AUDIO_IO_HANDLE_NONE;
+    strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+void DeviceDescriptor::toAudioPort(struct audio_port *port) const
+{
+    ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
+    AudioPort::toAudioPort(port);
+    port->id = mId;
+    toAudioPortConfig(&port->active_config);
+    port->ext.device.type = mDeviceType;
+    port->ext.device.hw_module = mModule->mHandle;
+    strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+status_t DeviceDescriptor::dump(int fd, int spaces, int index) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
+    result.append(buffer);
+    if (mId != 0) {
+        snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
+        result.append(buffer);
+    }
+    snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
+            ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+                    ARRAY_SIZE(sDeviceNameToEnumTable),
+                    mDeviceType));
+    result.append(buffer);
+    if (mAddress.size() != 0) {
+        snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
+        result.append(buffer);
+    }
+    write(fd, result.string(), result.size());
+    AudioPort::dump(fd, spaces);
+
+    return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
new file mode 100644
index 0000000..a0ea185
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -0,0 +1,282 @@
+/*
+ * Copyright (C) 2015 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 "APM::HwModule"
+//#define LOG_NDEBUG 0
+
+#include "HwModule.h"
+#include "IOProfile.h"
+#include "AudioGain.h"
+#include "ConfigParsingUtils.h"
+#include "audio_policy_conf.h"
+#include <hardware/audio.h>
+
+namespace android {
+
+HwModule::HwModule(const char *name)
+    : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
+      mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
+{
+}
+
+HwModule::~HwModule()
+{
+    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+        mOutputProfiles[i]->mSupportedDevices.clear();
+    }
+    for (size_t i = 0; i < mInputProfiles.size(); i++) {
+        mInputProfiles[i]->mSupportedDevices.clear();
+    }
+    free((void *)mName);
+}
+
+status_t HwModule::loadInput(cnode *root)
+{
+    cnode *node = root->first_child;
+
+    sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
+
+    while (node) {
+        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+            profile->loadSamplingRates((char *)node->value);
+        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+            profile->loadFormats((char *)node->value);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            profile->loadInChannels((char *)node->value);
+        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+            profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+                                                           mDeclaredDevices);
+        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+            profile->mFlags = ConfigParsingUtils::parseInputFlagNames((char *)node->value);
+        } else if (strcmp(node->name, GAINS_TAG) == 0) {
+            profile->loadGains(node);
+        }
+        node = node->next;
+    }
+    ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+            "loadInput() invalid supported devices");
+    ALOGW_IF(profile->mChannelMasks.size() == 0,
+            "loadInput() invalid supported channel masks");
+    ALOGW_IF(profile->mSamplingRates.size() == 0,
+            "loadInput() invalid supported sampling rates");
+    ALOGW_IF(profile->mFormats.size() == 0,
+            "loadInput() invalid supported formats");
+    if (!profile->mSupportedDevices.isEmpty() &&
+            (profile->mChannelMasks.size() != 0) &&
+            (profile->mSamplingRates.size() != 0) &&
+            (profile->mFormats.size() != 0)) {
+
+        ALOGV("loadInput() adding input Supported Devices %04x",
+              profile->mSupportedDevices.types());
+
+        mInputProfiles.add(profile);
+        return NO_ERROR;
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t HwModule::loadOutput(cnode *root)
+{
+    cnode *node = root->first_child;
+
+    sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
+
+    while (node) {
+        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+            profile->loadSamplingRates((char *)node->value);
+        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+            profile->loadFormats((char *)node->value);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            profile->loadOutChannels((char *)node->value);
+        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+            profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+                                                           mDeclaredDevices);
+        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+            profile->mFlags = ConfigParsingUtils::parseOutputFlagNames((char *)node->value);
+        } else if (strcmp(node->name, GAINS_TAG) == 0) {
+            profile->loadGains(node);
+        }
+        node = node->next;
+    }
+    ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+            "loadOutput() invalid supported devices");
+    ALOGW_IF(profile->mChannelMasks.size() == 0,
+            "loadOutput() invalid supported channel masks");
+    ALOGW_IF(profile->mSamplingRates.size() == 0,
+            "loadOutput() invalid supported sampling rates");
+    ALOGW_IF(profile->mFormats.size() == 0,
+            "loadOutput() invalid supported formats");
+    if (!profile->mSupportedDevices.isEmpty() &&
+            (profile->mChannelMasks.size() != 0) &&
+            (profile->mSamplingRates.size() != 0) &&
+            (profile->mFormats.size() != 0)) {
+
+        ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
+              profile->mSupportedDevices.types(), profile->mFlags);
+
+        mOutputProfiles.add(profile);
+        return NO_ERROR;
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t HwModule::loadDevice(cnode *root)
+{
+    cnode *node = root->first_child;
+
+    audio_devices_t type = AUDIO_DEVICE_NONE;
+    while (node) {
+        if (strcmp(node->name, DEVICE_TYPE) == 0) {
+            type = ConfigParsingUtils::parseDeviceNames((char *)node->value);
+            break;
+        }
+        node = node->next;
+    }
+    if (type == AUDIO_DEVICE_NONE ||
+            (!audio_is_input_device(type) && !audio_is_output_device(type))) {
+        ALOGW("loadDevice() bad type %08x", type);
+        return BAD_VALUE;
+    }
+    sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
+    deviceDesc->mModule = this;
+
+    node = root->first_child;
+    while (node) {
+        if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
+            deviceDesc->mAddress = String8((char *)node->value);
+        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+            if (audio_is_input_device(type)) {
+                deviceDesc->loadInChannels((char *)node->value);
+            } else {
+                deviceDesc->loadOutChannels((char *)node->value);
+            }
+        } else if (strcmp(node->name, GAINS_TAG) == 0) {
+            deviceDesc->loadGains(node);
+        }
+        node = node->next;
+    }
+
+    ALOGV("loadDevice() adding device name %s type %08x address %s",
+          deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
+
+    mDeclaredDevices.add(deviceDesc);
+
+    return NO_ERROR;
+}
+
+status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config,
+                                                  audio_devices_t device, String8 address)
+{
+    sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
+
+    profile->mSamplingRates.add(config->sample_rate);
+    profile->mChannelMasks.add(config->channel_mask);
+    profile->mFormats.add(config->format);
+
+    sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+    devDesc->mAddress = address;
+    profile->mSupportedDevices.add(devDesc);
+
+    mOutputProfiles.add(profile);
+
+    return NO_ERROR;
+}
+
+status_t HwModule::removeOutputProfile(String8 name)
+{
+    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+        if (mOutputProfiles[i]->mName == name) {
+            mOutputProfiles.removeAt(i);
+            break;
+        }
+    }
+
+    return NO_ERROR;
+}
+
+status_t HwModule::addInputProfile(String8 name, const audio_config_t *config,
+                                                  audio_devices_t device, String8 address)
+{
+    sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
+
+    profile->mSamplingRates.add(config->sample_rate);
+    profile->mChannelMasks.add(config->channel_mask);
+    profile->mFormats.add(config->format);
+
+    sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+    devDesc->mAddress = address;
+    profile->mSupportedDevices.add(devDesc);
+
+    ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
+
+    mInputProfiles.add(profile);
+
+    return NO_ERROR;
+}
+
+status_t HwModule::removeInputProfile(String8 name)
+{
+    for (size_t i = 0; i < mInputProfiles.size(); i++) {
+        if (mInputProfiles[i]->mName == name) {
+            mInputProfiles.removeAt(i);
+            break;
+        }
+    }
+
+    return NO_ERROR;
+}
+
+
+void HwModule::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "  - name: %s\n", mName);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "  - handle: %d\n", mHandle);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "  - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    if (mOutputProfiles.size()) {
+        write(fd, "  - outputs:\n", strlen("  - outputs:\n"));
+        for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+            snprintf(buffer, SIZE, "    output %zu:\n", i);
+            write(fd, buffer, strlen(buffer));
+            mOutputProfiles[i]->dump(fd);
+        }
+    }
+    if (mInputProfiles.size()) {
+        write(fd, "  - inputs:\n", strlen("  - inputs:\n"));
+        for (size_t i = 0; i < mInputProfiles.size(); i++) {
+            snprintf(buffer, SIZE, "    input %zu:\n", i);
+            write(fd, buffer, strlen(buffer));
+            mInputProfiles[i]->dump(fd);
+        }
+    }
+    if (mDeclaredDevices.size()) {
+        write(fd, "  - devices:\n", strlen("  - devices:\n"));
+        for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
+            mDeclaredDevices[i]->dump(fd, 4, i);
+        }
+    }
+}
+
+} //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
new file mode 100644
index 0000000..0214a2b
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -0,0 +1,149 @@
+/*
+ * Copyright (C) 2015 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 "APM::IOProfile"
+//#define LOG_NDEBUG 0
+
+#include "IOProfile.h"
+#include "HwModule.h"
+#include "AudioGain.h"
+
+namespace android {
+
+IOProfile::IOProfile(const String8& name, audio_port_role_t role,
+                                         const sp<HwModule>& module)
+    : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
+{
+}
+
+IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool IOProfile::isCompatibleProfile(audio_devices_t device,
+                                                        String8 address,
+                                                        uint32_t samplingRate,
+                                                        uint32_t *updatedSamplingRate,
+                                                        audio_format_t format,
+                                                        audio_channel_mask_t channelMask,
+                                                        uint32_t flags) const
+{
+    const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
+    const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+    ALOG_ASSERT(isPlaybackThread != isRecordThread);
+
+
+    if (device != AUDIO_DEVICE_NONE) {
+        // just check types if multiple devices are selected
+        if (popcount(device & ~AUDIO_DEVICE_BIT_IN) > 1) {
+            if ((mSupportedDevices.types() & device) != device) {
+                return false;
+            }
+        } else if (mSupportedDevices.getDevice(device, address) == 0) {
+            return false;
+        }
+    }
+
+    if (samplingRate == 0) {
+         return false;
+    }
+    uint32_t myUpdatedSamplingRate = samplingRate;
+    if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
+         return false;
+    }
+    if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
+            NO_ERROR) {
+         return false;
+    }
+
+    if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
+        return false;
+    }
+
+    if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
+            checkExactChannelMask(channelMask) != NO_ERROR)) {
+        return false;
+    }
+    if (isRecordThread && (!audio_is_input_channel(channelMask) ||
+            checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
+        return false;
+    }
+
+    if (isPlaybackThread && (mFlags & flags) != flags) {
+        return false;
+    }
+    // The only input flag that is allowed to be different is the fast flag.
+    // An existing fast stream is compatible with a normal track request.
+    // An existing normal stream is compatible with a fast track request,
+    // but the fast request will be denied by AudioFlinger and converted to normal track.
+    if (isRecordThread && ((mFlags ^ flags) &
+            ~AUDIO_INPUT_FLAG_FAST)) {
+        return false;
+    }
+
+    if (updatedSamplingRate != NULL) {
+        *updatedSamplingRate = myUpdatedSamplingRate;
+    }
+    return true;
+}
+
+void IOProfile::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    AudioPort::dump(fd, 4);
+
+    snprintf(buffer, SIZE, "    - flags: 0x%04x\n", mFlags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "    - devices:\n");
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    for (size_t i = 0; i < mSupportedDevices.size(); i++) {
+        mSupportedDevices[i]->dump(fd, 6, i);
+    }
+}
+
+void IOProfile::log()
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    ALOGV("    - sampling rates: ");
+    for (size_t i = 0; i < mSamplingRates.size(); i++) {
+        ALOGV("  %d", mSamplingRates[i]);
+    }
+
+    ALOGV("    - channel masks: ");
+    for (size_t i = 0; i < mChannelMasks.size(); i++) {
+        ALOGV("  0x%04x", mChannelMasks[i]);
+    }
+
+    ALOGV("    - formats: ");
+    for (size_t i = 0; i < mFormats.size(); i++) {
+        ALOGV("  0x%08x", mFormats[i]);
+    }
+
+    ALOGV("    - devices: 0x%04x\n", mSupportedDevices.types());
+    ALOGV("    - flags: 0x%04x\n", mFlags);
+}
+
+}; // namespace android