diff --git a/services/audiopolicy/AudioPolicyManagerBase.h b/services/audiopolicy/AudioPolicyManagerBase.h
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+++ b/services/audiopolicy/AudioPolicyManagerBase.h
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+/*
+ * Copyright (C) 2009 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.
+ */
+
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+
+
+namespace android_audio_legacy {
+    using android::KeyedVector;
+    using android::DefaultKeyedVector;
+    using android::SortedVector;
+
+// ----------------------------------------------------------------------------
+
+#define MAX_DEVICE_ADDRESS_LEN 20
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN  0.016
+// Time in milliseconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY  5000
+// Time in milliseconds after media stopped playing during which we consider that the
+// sonification should be as unobtrusive as during the time media was playing.
+#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
+// Time in milliseconds during witch some streams are muted while the audio path
+// is switched
+#define MUTE_TIME_MS 2000
+
+#define NUM_TEST_OUTPUTS 5
+
+#define NUM_VOL_CURVE_KNEES 2
+
+// Default minimum length allowed for offloading a compressed track
+// Can be overridden by the audio.offload.min.duration.secs property
+#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManagerBase implements audio policy manager behavior common to all platforms.
+// Each platform must implement an AudioPolicyManager class derived from AudioPolicyManagerBase
+// and override methods for which the platform specific behavior differs from the implementation
+// in AudioPolicyManagerBase. Even if no specific behavior is required, the AudioPolicyManager
+// class must be implemented as well as the class factory function createAudioPolicyManager()
+// and provided in a shared library libaudiopolicy.so.
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManagerBase: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+    , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+                AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface);
+        virtual ~AudioPolicyManagerBase();
+
+        // AudioPolicyInterface
+        virtual status_t setDeviceConnectionState(audio_devices_t device,
+                                                          AudioSystem::device_connection_state state,
+                                                          const char *device_address);
+        virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device,
+                                                                              const char *device_address);
+        virtual void setPhoneState(int state);
+        virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+        virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
+        virtual void setSystemProperty(const char* property, const char* value);
+        virtual status_t initCheck();
+        virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            AudioSystem::output_flags flags,
+                                            const audio_offload_info_t *offloadInfo);
+        virtual status_t startOutput(audio_io_handle_t output,
+                                     AudioSystem::stream_type stream,
+                                     int session = 0);
+        virtual status_t stopOutput(audio_io_handle_t output,
+                                    AudioSystem::stream_type stream,
+                                    int session = 0);
+        virtual void releaseOutput(audio_io_handle_t output);
+        virtual audio_io_handle_t getInput(int inputSource,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            AudioSystem::audio_in_acoustics acoustics);
+
+        // indicates to the audio policy manager that the input starts being used.
+        virtual status_t startInput(audio_io_handle_t input);
+
+        // indicates to the audio policy manager that the input stops being used.
+        virtual status_t stopInput(audio_io_handle_t input);
+        virtual void releaseInput(audio_io_handle_t input);
+        virtual void initStreamVolume(AudioSystem::stream_type stream,
+                                                    int indexMin,
+                                                    int indexMax);
+        virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream,
+                                              int index,
+                                              audio_devices_t device);
+        virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream,
+                                              int *index,
+                                              audio_devices_t device);
+
+        // return the strategy corresponding to a given stream type
+        virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream);
+
+        // return the enabled output devices for the given stream type
+        virtual audio_devices_t getDevicesForStream(AudioSystem::stream_type stream);
+
+        virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
+        virtual status_t registerEffect(const effect_descriptor_t *desc,
+                                        audio_io_handle_t io,
+                                        uint32_t strategy,
+                                        int session,
+                                        int id);
+        virtual status_t unregisterEffect(int id);
+        virtual status_t setEffectEnabled(int id, bool enabled);
+
+        virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const;
+        // return whether a stream is playing remotely, override to change the definition of
+        //   local/remote playback, used for instance by notification manager to not make
+        //   media players lose audio focus when not playing locally
+        virtual bool isStreamActiveRemotely(int stream, uint32_t inPastMs = 0) const;
+        virtual bool isSourceActive(audio_source_t source) const;
+
+        virtual status_t dump(int fd);
+
+        virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+protected:
+
+        enum routing_strategy {
+            STRATEGY_MEDIA,
+            STRATEGY_PHONE,
+            STRATEGY_SONIFICATION,
+            STRATEGY_SONIFICATION_RESPECTFUL,
+            STRATEGY_DTMF,
+            STRATEGY_ENFORCED_AUDIBLE,
+            NUM_STRATEGIES
+        };
+
+        // 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};
+
+        class VolumeCurvePoint
+        {
+        public:
+            int mIndex;
+            float mDBAttenuation;
+        };
+
+        // device categories used for volume curve management.
+        enum device_category {
+            DEVICE_CATEGORY_HEADSET,
+            DEVICE_CATEGORY_SPEAKER,
+            DEVICE_CATEGORY_EARPIECE,
+            DEVICE_CATEGORY_CNT
+        };
+
+        class IOProfile;
+
+        class HwModule {
+        public:
+                    HwModule(const char *name);
+                    ~HwModule();
+
+            void dump(int fd);
+
+            const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+            audio_module_handle_t mHandle;
+            Vector <IOProfile *> mOutputProfiles; // output profiles exposed by this module
+            Vector <IOProfile *> mInputProfiles;  // input profiles exposed by this module
+        };
+
+        // 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:
+            IOProfile(HwModule *module);
+            ~IOProfile();
+
+            bool isCompatibleProfile(audio_devices_t device,
+                                     uint32_t samplingRate,
+                                     audio_format_t format,
+                                     audio_channel_mask_t channelMask,
+                                     audio_output_flags_t flags) const;
+
+            void dump(int fd);
+
+            // 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
+            audio_devices_t mSupportedDevices; // supported devices (devices this output can be
+                                               // routed to)
+            audio_output_flags_t mFlags; // attribute flags (e.g primary output,
+                                                // direct output...). For outputs only.
+            HwModule *mModule;                     // audio HW module exposing this I/O stream
+        };
+
+        // default volume curve
+        static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        // default volume curve for media strategy
+        static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        // volume curve for media strategy on speakers
+        static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        // volume curve for sonification strategy on speakers
+        static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+        // default volume curves per stream and device category. See initializeVolumeCurves()
+        static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT];
+
+        // 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:
+            AudioOutputDescriptor(const IOProfile *profile);
+
+            status_t    dump(int fd);
+
+            audio_devices_t device() const;
+            void changeRefCount(AudioSystem::stream_type stream, int delta);
+
+            bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+            audio_devices_t supportedDevices();
+            uint32_t latency();
+            bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc);
+            bool isActive(uint32_t inPastMs = 0) const;
+            bool isStreamActive(AudioSystem::stream_type stream,
+                                uint32_t inPastMs = 0,
+                                nsecs_t sysTime = 0) const;
+            bool isStrategyActive(routing_strategy strategy,
+                             uint32_t inPastMs = 0,
+                             nsecs_t sysTime = 0) const;
+
+            audio_io_handle_t mId;              // output handle
+            uint32_t mSamplingRate;             //
+            audio_format_t mFormat;             //
+            audio_channel_mask_t mChannelMask;     // output configuration
+            uint32_t mLatency;                  //
+            audio_output_flags_t mFlags;   //
+            audio_devices_t mDevice;                   // current device this output is routed to
+            uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output
+            nsecs_t mStopTime[AudioSystem::NUM_STREAM_TYPES];
+            AudioOutputDescriptor *mOutput1;    // used by duplicated outputs: first output
+            AudioOutputDescriptor *mOutput2;    // used by duplicated outputs: second output
+            float mCurVolume[AudioSystem::NUM_STREAM_TYPES];   // current stream volume
+            int mMuteCount[AudioSystem::NUM_STREAM_TYPES];     // mute request counter
+            const 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)
+        };
+
+        // 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:
+            AudioInputDescriptor(const IOProfile *profile);
+
+            status_t    dump(int fd);
+
+            uint32_t mSamplingRate;                     //
+            audio_format_t mFormat;                     // input configuration
+            audio_channel_mask_t mChannelMask;             //
+            audio_devices_t mDevice;                    // current device this input is routed to
+            uint32_t mRefCount;                         // number of AudioRecord clients using this output
+            int      mInputSource;                      // input source selected by application (mediarecorder.h)
+            const IOProfile *mProfile;                  // I/O profile this output derives from
+        };
+
+        // 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[DEVICE_CATEGORY_CNT];
+        };
+
+        // stream descriptor used for volume control
+        class EffectDescriptor
+        {
+        public:
+
+            status_t dump(int fd);
+
+            int mIo;                // io the effect is attached to
+            routing_strategy mStrategy; // routing strategy the effect is associated to
+            int mSession;               // audio session the effect is on
+            effect_descriptor_t mDesc;  // effect descriptor
+            bool mEnabled;              // enabled state: CPU load being used or not
+        };
+
+        void addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc);
+
+        // return the strategy corresponding to a given stream type
+        static routing_strategy getStrategy(AudioSystem::stream_type stream);
+
+        // return appropriate device for streams handled by the specified strategy according to current
+        // phone state, connected devices...
+        // if fromCache is true, the device is returned from mDeviceForStrategy[],
+        // otherwise it is determine by current state
+        // (device connected,phone state, force use, a2dp output...)
+        // This allows to:
+        //  1 speed up process when the state is stable (when starting or stopping an output)
+        //  2 access to either current device selection (fromCache == true) or
+        // "future" device selection (fromCache == false) when called from a context
+        //  where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+        //  before updateDevicesAndOutputs() is called.
+        virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+                                                     bool fromCache);
+
+        // change the route of the specified output. Returns the number of ms we have slept to
+        // allow new routing to take effect in certain cases.
+        uint32_t setOutputDevice(audio_io_handle_t output,
+                             audio_devices_t device,
+                             bool force = false,
+                             int delayMs = 0);
+
+        // select input device corresponding to requested audio source
+        virtual audio_devices_t getDeviceForInputSource(int inputSource);
+
+        // return io handle of active input or 0 if no input is active
+        //    Only considers inputs from physical devices (e.g. main mic, headset mic) when
+        //    ignoreVirtualInputs is true.
+        audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+        // initialize volume curves for each strategy and device category
+        void initializeVolumeCurves();
+
+        // compute the actual volume for a given stream according to the requested index and a particular
+        // device
+        virtual float computeVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device);
+
+        // check that volume change is permitted, compute and send new volume to audio hardware
+        status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+        // apply all stream volumes to the specified output and device
+        void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+        // Mute or unmute all streams handled by the specified strategy on the specified output
+        void setStrategyMute(routing_strategy strategy,
+                             bool on,
+                             audio_io_handle_t output,
+                             int delayMs = 0,
+                             audio_devices_t device = (audio_devices_t)0);
+
+        // Mute or unmute the stream on the specified output
+        void setStreamMute(int stream,
+                           bool on,
+                           audio_io_handle_t output,
+                           int delayMs = 0,
+                           audio_devices_t device = (audio_devices_t)0);
+
+        // handle special cases for sonification strategy while in call: mute streams or replace by
+        // a special tone in the device used for communication
+        void handleIncallSonification(int stream, bool starting, bool stateChange);
+
+        // true if device is in a telephony or VoIP call
+        virtual bool isInCall();
+
+        // true if given state represents a device in a telephony or VoIP call
+        virtual bool isStateInCall(int state);
+
+        // when a device is connected, checks if an open output can be routed
+        // to this device. If none is open, tries to open one of the available outputs.
+        // Returns an output suitable to this device or 0.
+        // when a device is disconnected, checks if an output is not used any more and
+        // returns its handle if any.
+        // transfers the audio tracks and effects from one output thread to another accordingly.
+        status_t checkOutputsForDevice(audio_devices_t device,
+                                       AudioSystem::device_connection_state state,
+                                       SortedVector<audio_io_handle_t>& outputs,
+                                       const String8 paramStr);
+
+        // close an output and its companion duplicating output.
+        void closeOutput(audio_io_handle_t output);
+
+        // checks and if necessary changes outputs used for all strategies.
+        // must be called every time a condition that affects the output choice for a given strategy
+        // changes: connected device, phone state, force use...
+        // Must be called before updateDevicesAndOutputs()
+        void checkOutputForStrategy(routing_strategy strategy);
+
+        // Same as checkOutputForStrategy() but for a all strategies in order of priority
+        void checkOutputForAllStrategies();
+
+        // manages A2DP output suspend/restore according to phone state and BT SCO usage
+        void checkA2dpSuspend();
+
+        // returns the A2DP output handle if it is open or 0 otherwise
+        audio_io_handle_t getA2dpOutput();
+
+        // selects the most appropriate device on output for current state
+        // must be called every time a condition that affects the device choice for a given output is
+        // changed: connected device, phone state, force use, output start, output stop..
+        // see getDeviceForStrategy() for the use of fromCache parameter
+
+        audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache);
+        // updates cache of device used by all strategies (mDeviceForStrategy[])
+        // must be called every time a condition that affects the device choice for a given strategy is
+        // changed: connected device, phone state, force use...
+        // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
+         // Must be called after checkOutputForAllStrategies()
+
+        void updateDevicesAndOutputs();
+
+        virtual uint32_t getMaxEffectsCpuLoad();
+        virtual uint32_t getMaxEffectsMemory();
+#ifdef AUDIO_POLICY_TEST
+        virtual     bool        threadLoop();
+                    void        exit();
+        int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+        status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled);
+
+        // returns the category the device belongs to with regard to volume curve management
+        static 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);
+
+        SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+                        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs);
+        bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+                                           SortedVector<audio_io_handle_t>& outputs2);
+
+        // mute/unmute strategies using an incompatible device combination
+        // if muting, wait for the audio in pcm buffer to be drained before proceeding
+        // if unmuting, unmute only after the specified delay
+        // Returns the number of ms waited
+        uint32_t  checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+                                            audio_devices_t prevDevice,
+                                            uint32_t delayMs);
+
+        audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+                                       AudioSystem::output_flags flags);
+        IOProfile *getInputProfile(audio_devices_t device,
+                                   uint32_t samplingRate,
+                                   audio_format_t format,
+                                   audio_channel_mask_t channelMask);
+        IOProfile *getProfileForDirectOutput(audio_devices_t device,
+                                                       uint32_t samplingRate,
+                                                       audio_format_t format,
+                                                       audio_channel_mask_t channelMask,
+                                                       audio_output_flags_t flags);
+
+        audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
+
+        bool isNonOffloadableEffectEnabled();
+
+        //
+        // Audio policy configuration file parsing (audio_policy.conf)
+        //
+        static uint32_t stringToEnum(const struct StringToEnum *table,
+                                     size_t size,
+                                     const char *name);
+        static bool stringToBool(const char *value);
+        static audio_output_flags_t parseFlagNames(char *name);
+        static audio_devices_t parseDeviceNames(char *name);
+        void loadSamplingRates(char *name, IOProfile *profile);
+        void loadFormats(char *name, IOProfile *profile);
+        void loadOutChannels(char *name, IOProfile *profile);
+        void loadInChannels(char *name, IOProfile *profile);
+        status_t loadOutput(cnode *root,  HwModule *module);
+        status_t loadInput(cnode *root,  HwModule *module);
+        void loadHwModule(cnode *root);
+        void loadHwModules(cnode *root);
+        void loadGlobalConfig(cnode *root);
+        status_t loadAudioPolicyConfig(const char *path);
+        void defaultAudioPolicyConfig(void);
+
+
+        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
+        audio_io_handle_t mPrimaryOutput;              // primary output handle
+        // list of descriptors for outputs currently opened
+        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;
+        // copy of mOutputs before setDeviceConnectionState() opens new outputs
+        // reset to mOutputs when updateDevicesAndOutputs() is called.
+        DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs;
+        DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs;     // list of input descriptors
+        audio_devices_t mAvailableOutputDevices; // bit field of all available output devices
+        audio_devices_t mAvailableInputDevices; // bit field of all available input devices
+                                                // without AUDIO_DEVICE_BIT_IN to allow direct bit
+                                                // field comparisons
+        int mPhoneState;                                                    // current phone state
+        AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE];   // current forced use configuration
+
+        StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES];           // stream descriptors for volume control
+        String8 mA2dpDeviceAddress;                                         // A2DP device MAC address
+        String8 mScoDeviceAddress;                                          // SCO device MAC address
+        String8 mUsbCardAndDevice; // USB audio ALSA card and device numbers:
+                                   // card=<card_number>;device=<><device_number>
+        bool    mLimitRingtoneVolume;                                       // limit ringtone volume to music volume if headset connected
+        audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+        float   mLastVoiceVolume;                                           // last voice volume value sent to audio HAL
+
+        // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
+        static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
+        // Maximum memory allocated to audio effects in KB
+        static const uint32_t MAX_EFFECTS_MEMORY = 512;
+        uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
+        uint32_t mTotalEffectsMemory;  // current memory used by effects
+        KeyedVector<int, EffectDescriptor *> mEffects;  // list of registered audio effects
+        bool    mA2dpSuspended;  // true if A2DP output is suspended
+        bool mHasA2dp; // true on platforms with support for bluetooth A2DP
+        bool mHasUsb; // true on platforms with support for USB audio
+        bool mHasRemoteSubmix; // true on platforms with support for remote presentation of a submix
+        audio_devices_t mAttachedOutputDevices; // output devices always available on the platform
+        audio_devices_t mDefaultOutputDevice; // output device selected by default at boot time
+                                              // (must be in mAttachedOutputDevices)
+        bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
+                                // to boost soft sounds, used to adjust volume curves accordingly
+
+        Vector <HwModule *> mHwModules;
+
+#ifdef AUDIO_POLICY_TEST
+        Mutex   mLock;
+        Condition mWaitWorkCV;
+
+        int             mCurOutput;
+        bool            mDirectOutput;
+        audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
+        int             mTestInput;
+        uint32_t        mTestDevice;
+        uint32_t        mTestSamplingRate;
+        uint32_t        mTestFormat;
+        uint32_t        mTestChannels;
+        uint32_t        mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+private:
+        static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+                int indexInUi);
+        // updates device caching and output for streams that can influence the
+        //    routing of notifications
+        void handleNotificationRoutingForStream(AudioSystem::stream_type stream);
+        static bool isVirtualInputDevice(audio_devices_t device);
+};
+
+};