Moving things around for ICS leak

29 files changed, 54 insertions, 4739 deletions

diff --git a/BoardConfig.mk b/BoardConfig.mk
index f6ecfe5..cd6657d 100755
--- a/BoardConfig.mk
+++ b/BoardConfig.mk
@@ -29,8 +29,8 @@ TARGET_ARCH_VARIANT_CPU := cortex-a9
 ARCH_ARM_HAVE_TLS_REGISTER := true
 TARGET_GLOBAL_CFLAGS += -mtune=cortex-a9 -mfpu=neon -mfloat-abi=softfp
 TARGET_GLOBAL_CPPFLAGS += -mtune=cortex-a9 -mfpu=neon -mfloat-abi=softfp
-TARGET_BOARD_PLATFORM := smdkv310
-TARGET_BOOTLOADER_BOARD_NAME := GT-N7000
+TARGET_BOARD_PLATFORM := s5pc210
+TARGET_BOOTLOADER_BOARD_NAME := smdk4210
 TARGET_BOARD_INFO_FILE := device/samsung/galaxynote/board-info.txt
 
 TARGET_NO_BOOTLOADER := true
@@ -60,12 +60,12 @@ TARGET_RELEASETOOL_IMG_FROM_TARGET_SCRIPT := ./device/samsung/galaxynote/release
 # Graphics (Mali 400)
 BOARD_EGL_CFG := device/samsung/galaxynote/configs/egl.cfg
 USE_OPENGL_RENDERER := true
+
+# HWComposer
 BOARD_USES_HWCOMPOSER := true
-BOARD_USES_LEGACY_EGL := true
-COMMON_GLOBAL_CFLAGS += -DMISSING_EGL_EXTERNAL_IMAGE -DMISSING_EGL_PIXEL_FORMAT_YV12 -DMISSING_GRALLOC_BUFFERS
+BOARD_USE_SECTVOUT := true
 
 # Audio
-BOARD_USES_AUDIO_LEGACY := true
 BOARD_USE_YAMAHAPLAYER := true
 
 # Camera
diff --git a/audio/Android.mk b/audio/Android.mk
deleted file mode 100755
index 5026c54..0000000
--- a/audio/Android.mk
+++ /dev/null
@@ -1,103 +0,0 @@
-#
-# Copyright (C) 2011 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.
-
-ifeq ($(BOARD_USES_AUDIO_LEGACY),true)
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    libmedia \
-    libhardware_legacy
-
-LOCAL_SHARED_LIBRARIES += libdl
-
-LOCAL_SHARED_LIBRARIES += libaudio
-
-ifeq ($(BOARD_FORCE_STATIC_A2DP),true)
-  LOCAL_SHARED_LIBRARIES += liba2dp
-endif
-
-
-LOCAL_STATIC_LIBRARIES := \
-    libmedia_helper
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
-    libaudiohw_legacy
-
-include $(BUILD_SHARED_LIBRARY)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
-    AudioPolicyManagerBase.cpp \
-    AudioPolicyCompatClient.cpp \
-    audio_policy_hal.cpp
-
-ifeq ($(AUDIO_POLICY_TEST),true)
-  LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
-endif
-
-ifeq ($(BOARD_HAVE_BLUETOOTH),true)
-  LOCAL_CFLAGS += -DWITH_A2DP
-endif
-
-LOCAL_STATIC_LIBRARIES := libmedia_helper
-LOCAL_MODULE := libaudiopolicy_legacy2
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_STATIC_LIBRARY)
-
-
-# The default audio policy, for now still implemented on top of legacy
-# policy code
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
-    AudioPolicyManagerDefault.cpp
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    libmedia
-
-LOCAL_STATIC_LIBRARIES := \
-    libmedia_helper
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
-    libaudiopolicy_legacy2
-
-ifeq ($(BOARD_USES_AUDIO_LEGACY),true)
-LOCAL_SHARED_LIBRARIES += libaudiopolicy
-endif
-
-LOCAL_C_INCLUDES := $(LOCAL_PATH)
-LOCAL_MODULE := audio_policy.$(TARGET_BOARD_PLATFORM)
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
-LOCAL_MODULE_TAGS := optional
-
-ifeq ($(BOARD_HAVE_BLUETOOTH),true)
-  LOCAL_CFLAGS += -DWITH_A2DP
-endif
-
-include $(BUILD_SHARED_LIBRARY)
-
-endif ## AUDIOPOLICY
diff --git a/audio/AudioPolicyCompatClient.cpp b/audio/AudioPolicyCompatClient.cpp
deleted file mode 100755
index 08c31c5..0000000
--- a/audio/AudioPolicyCompatClient.cpp
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- * Copyright (C) 2011 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 "AudioPolicyCompatClient"
-//#define LOG_NDEBUG 0
-
-#include <stdint.h>
-
-#include <hardware/hardware.h>
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioSystemLegacy.h>
-
-#include "AudioPolicyCompatClient.h"
-
-namespace android_audio_legacy {
-
-audio_io_handle_t AudioPolicyCompatClient::openOutput(uint32_t *pDevices,
-                                uint32_t *pSamplingRate,
-                                uint32_t *pFormat,
-                                uint32_t *pChannels,
-                                uint32_t *pLatencyMs,
-                                AudioSystem::output_flags flags)
-{
-    return mServiceOps->open_output(mService, pDevices, pSamplingRate, pFormat,
-                                    pChannels, pLatencyMs,
-                                    (audio_policy_output_flags_t)flags);
-}
-
-audio_io_handle_t AudioPolicyCompatClient::openDuplicateOutput(audio_io_handle_t output1,
-                                                          audio_io_handle_t output2)
-{
-    return mServiceOps->open_duplicate_output(mService, output1, output2);
-}
-
-status_t AudioPolicyCompatClient::closeOutput(audio_io_handle_t output)
-{
-    return mServiceOps->close_output(mService, output);
-}
-
-status_t AudioPolicyCompatClient::suspendOutput(audio_io_handle_t output)
-{
-    return mServiceOps->suspend_output(mService, output);
-}
-
-status_t AudioPolicyCompatClient::restoreOutput(audio_io_handle_t output)
-{
-    return mServiceOps->restore_output(mService, output);
-}
-
-audio_io_handle_t AudioPolicyCompatClient::openInput(uint32_t *pDevices,
-                                uint32_t *pSamplingRate,
-                                uint32_t *pFormat,
-                                uint32_t *pChannels,
-                                uint32_t acoustics)
-{
-    return mServiceOps->open_input(mService, pDevices, pSamplingRate, pFormat,
-                                   pChannels, acoustics);
-}
-
-status_t AudioPolicyCompatClient::closeInput(audio_io_handle_t input)
-{
-    return mServiceOps->close_input(mService, input);
-}
-
-status_t AudioPolicyCompatClient::setStreamOutput(AudioSystem::stream_type stream,
-                                             audio_io_handle_t output)
-{
-    return mServiceOps->set_stream_output(mService, (audio_stream_type_t)stream,
-                                          output);
-}
-
-status_t AudioPolicyCompatClient::moveEffects(int session, audio_io_handle_t srcOutput,
-                                               audio_io_handle_t dstOutput)
-{
-    return mServiceOps->move_effects(mService, session, srcOutput, dstOutput);
-}
-
-String8 AudioPolicyCompatClient::getParameters(audio_io_handle_t ioHandle, const String8& keys)
-{
-    char *str;
-    String8 out_str8;
-
-    str = mServiceOps->get_parameters(mService, ioHandle, keys.string());
-    out_str8 = String8(str);
-    free(str);
-
-    return out_str8;
-}
-
-void AudioPolicyCompatClient::setParameters(audio_io_handle_t ioHandle,
-                                            const String8& keyValuePairs,
-                                            int delayMs)
-{
-    mServiceOps->set_parameters(mService, ioHandle, keyValuePairs.string(),
-                           delayMs);
-}
-
-status_t AudioPolicyCompatClient::setStreamVolume(
-                                             AudioSystem::stream_type stream,
-                                             float volume,
-                                             audio_io_handle_t output,
-                                             int delayMs)
-{
-    return mServiceOps->set_stream_volume(mService, (audio_stream_type_t)stream,
-                                          volume, output, delayMs);
-}
-
-status_t AudioPolicyCompatClient::startTone(ToneGenerator::tone_type tone,
-                                       AudioSystem::stream_type stream)
-{
-    return mServiceOps->start_tone(mService,
-                                   AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
-                                   (audio_stream_type_t)stream);
-}
-
-status_t AudioPolicyCompatClient::stopTone()
-{
-    return mServiceOps->stop_tone(mService);
-}
-
-status_t AudioPolicyCompatClient::setVoiceVolume(float volume, int delayMs)
-{
-    return mServiceOps->set_voice_volume(mService, volume, delayMs);
-}
-
-}; // namespace android_audio_legacy
diff --git a/audio/AudioPolicyCompatClient.h b/audio/AudioPolicyCompatClient.h
deleted file mode 100755
index 4feee37..0000000
--- a/audio/AudioPolicyCompatClient.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_AUDIOPOLICYCLIENTLEGACY_H
-#define ANDROID_AUDIOPOLICYCLIENTLEGACY_H
-
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioSystemLegacy.h>
-#include <hardware_legacy/AudioPolicyInterface.h>
-
-/************************************/
-/* FOR BACKWARDS COMPATIBILITY ONLY */
-/************************************/
-namespace android_audio_legacy {
-
-class AudioPolicyCompatClient : public AudioPolicyClientInterface {
-public:
-    AudioPolicyCompatClient(struct audio_policy_service_ops *serviceOps,
-                            void *service) :
-            mServiceOps(serviceOps) , mService(service) {}
-
-    virtual audio_io_handle_t openOutput(uint32_t *pDevices,
-                                         uint32_t *pSamplingRate,
-                                         uint32_t *pFormat,
-                                         uint32_t *pChannels,
-                                         uint32_t *pLatencyMs,
-                                         AudioSystem::output_flags flags);
-    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
-                                                  audio_io_handle_t output2);
-    virtual status_t closeOutput(audio_io_handle_t output);
-    virtual status_t suspendOutput(audio_io_handle_t output);
-    virtual status_t restoreOutput(audio_io_handle_t output);
-    virtual audio_io_handle_t openInput(uint32_t *pDevices,
-                                        uint32_t *pSamplingRate,
-                                        uint32_t *pFormat,
-                                        uint32_t *pChannels,
-                                        uint32_t acoustics);
-    virtual status_t closeInput(audio_io_handle_t input);
-    virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output);
-    virtual status_t moveEffects(int session,
-                                 audio_io_handle_t srcOutput,
-                                 audio_io_handle_t dstOutput);
-
-    virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
-    virtual void setParameters(audio_io_handle_t ioHandle,
-                               const String8& keyValuePairs,
-                               int delayMs = 0);
-    virtual status_t setStreamVolume(AudioSystem::stream_type stream,
-                                     float volume,
-                                     audio_io_handle_t output,
-                                     int delayMs = 0);
-    virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream);
-    virtual status_t stopTone();
-    virtual status_t setVoiceVolume(float volume, int delayMs = 0);
-
-private:
-    struct audio_policy_service_ops* mServiceOps;
-    void*                            mService;
-};
-
-}; // namespace android_audio_legacy
-
-#endif // ANDROID_AUDIOPOLICYCLIENTLEGACY_H
diff --git a/audio/AudioPolicyInterface.h b/audio/AudioPolicyInterface.h
deleted file mode 100755
index 78f87da..0000000
--- a/audio/AudioPolicyInterface.h
+++ /dev/null
@@ -1,238 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef ANDROID_AUDIOPOLICYINTERFACE_H
-#define ANDROID_AUDIOPOLICYINTERFACE_H
-
-#include <media/AudioSystem.h>
-#include <media/ToneGenerator.h>
-#include <utils/String8.h>
-
-#include <hardware_legacy/AudioSystemLegacy.h>
-
-namespace android_audio_legacy {
-    using android::Vector;
-    using android::String8;
-    using android::ToneGenerator;
-
-// ----------------------------------------------------------------------------
-
-// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces
-// between the platform specific audio policy manager and Android generic audio policy manager.
-// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class.
-// This implementation makes use of the AudioPolicyClientInterface to control the activity and
-// configuration of audio input and output streams.
-//
-// The platform specific audio policy manager is in charge of the audio routing and volume control
-// policies for a given platform.
-// The main roles of this module are:
-//   - keep track of current system state (removable device connections, phone state, user requests...).
-//   System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface.
-//   - process getOutput() queries received when AudioTrack objects are created: Those queries
-//   return a handler on an output that has been selected, configured and opened by the audio policy manager and that
-//   must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method.
-//   When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide
-//   to close or reconfigure the output depending on other streams using this output and current system state.
-//   - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs.
-//   - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value
-//   applicable to each output as a function of platform specific settings and current output route (destination device). It
-//   also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries).
-//
-// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so)
-// and is linked with libaudioflinger.so
-
-
-//    Audio Policy Manager Interface
-class AudioPolicyInterface
-{
-
-public:
-    virtual ~AudioPolicyInterface() {}
-    //
-    // configuration functions
-    //
-
-    // indicate a change in device connection status
-    virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
-                                          AudioSystem::device_connection_state state,
-                                          const char *device_address) = 0;
-    // retreive a device connection status
-    virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
-                                                                          const char *device_address) = 0;
-    // indicate a change in phone state. Valid phones states are defined by AudioSystem::audio_mode
-    virtual void setPhoneState(int state) = 0;
-    // indicate a change in ringer mode
-    virtual void setRingerMode(uint32_t mode, uint32_t mask) = 0;
-    // force using a specific device category for the specified usage
-    virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0;
-    // retreive current device category forced for a given usage
-    virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0;
-    // set a system property (e.g. camera sound always audible)
-    virtual void setSystemProperty(const char* property, const char* value) = 0;
-    // check proper initialization
-    virtual status_t initCheck() = 0;
-
-    //
-    // Audio routing query functions
-    //
-
-    // request an output appriate for playback of the supplied stream type and parameters
-    virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
-                                        uint32_t samplingRate = 0,
-                                        uint32_t format = AudioSystem::FORMAT_DEFAULT,
-                                        uint32_t channels = 0,
-                                        AudioSystem::output_flags flags = AudioSystem::OUTPUT_FLAG_INDIRECT) = 0;
-    // indicates to the audio policy manager that the output starts being used by corresponding stream.
-    virtual status_t startOutput(audio_io_handle_t output,
-                                 AudioSystem::stream_type stream,
-                                 int session = 0) = 0;
-    // indicates to the audio policy manager that the output stops being used by corresponding stream.
-    virtual status_t stopOutput(audio_io_handle_t output,
-                                AudioSystem::stream_type stream,
-                                int session = 0) = 0;
-    // releases the output.
-    virtual void releaseOutput(audio_io_handle_t output) = 0;
-
-    // request an input appriate for record from the supplied device with supplied parameters.
-    virtual audio_io_handle_t getInput(int inputSource,
-                                    uint32_t samplingRate = 0,
-                                    uint32_t Format = AudioSystem::FORMAT_DEFAULT,
-                                    uint32_t channels = 0,
-                                    AudioSystem::audio_in_acoustics acoustics = (AudioSystem::audio_in_acoustics)0) = 0;
-    // indicates to the audio policy manager that the input starts being used.
-    virtual status_t startInput(audio_io_handle_t input) = 0;
-    // indicates to the audio policy manager that the input stops being used.
-    virtual status_t stopInput(audio_io_handle_t input) = 0;
-    // releases the input.
-    virtual void releaseInput(audio_io_handle_t input) = 0;
-
-    //
-    // volume control functions
-    //
-
-    // initialises stream volume conversion parameters by specifying volume index range.
-    virtual void initStreamVolume(AudioSystem::stream_type stream,
-                                      int indexMin,
-                                      int indexMax) = 0;
-
-    // sets the new stream volume at a level corresponding to the supplied index
-    virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index) = 0;
-    // retreive current volume index for the specified stream
-    virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) = 0;
-
-    // return the strategy corresponding to a given stream type
-    virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream) = 0;
-
-    // return the enabled output devices for the given stream type
-    virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream) = 0;
-
-    // Audio effect management
-    virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc) = 0;
-    virtual status_t registerEffect(effect_descriptor_t *desc,
-                                    audio_io_handle_t io,
-                                    uint32_t strategy,
-                                    int session,
-                                    int id) = 0;
-    virtual status_t unregisterEffect(int id) = 0;
-
-    virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const = 0;
-
-    //dump state
-    virtual status_t    dump(int fd) = 0;
-};
-
-
-// Audio Policy client Interface
-class AudioPolicyClientInterface
-{
-public:
-    virtual ~AudioPolicyClientInterface() {}
-
-    //
-    // Audio output Control functions
-    //
-
-    // opens an audio output with the requested parameters. The parameter values can indicate to use the default values
-    // in case the audio policy manager has no specific requirements for the output being opened.
-    // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream.
-    // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
-    virtual audio_io_handle_t openOutput(uint32_t *pDevices,
-                                    uint32_t *pSamplingRate,
-                                    uint32_t *pFormat,
-                                    uint32_t *pChannels,
-                                    uint32_t *pLatencyMs,
-                                    AudioSystem::output_flags flags) = 0;
-    // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
-    // a special mixer thread in the AudioFlinger.
-    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0;
-    // closes the output stream
-    virtual status_t closeOutput(audio_io_handle_t output) = 0;
-    // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in
-    // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded.
-    virtual status_t suspendOutput(audio_io_handle_t output) = 0;
-    // restores a suspended output.
-    virtual status_t restoreOutput(audio_io_handle_t output) = 0;
-
-    //
-    // Audio input Control functions
-    //
-
-    // opens an audio input
-    virtual audio_io_handle_t openInput(uint32_t *pDevices,
-                                    uint32_t *pSamplingRate,
-                                    uint32_t *pFormat,
-                                    uint32_t *pChannels,
-                                    uint32_t acoustics) = 0;
-    // closes an audio input
-    virtual status_t closeInput(audio_io_handle_t input) = 0;
-    //
-    // misc control functions
-    //
-
-    // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
-    // for each output (destination device) it is attached to.
-    virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0;
-
-    // reroute a given stream type to the specified output
-    virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output) = 0;
-
-    // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
-    virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0) = 0;
-    // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
-    virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0;
-
-    // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing
-    // over a telephony device during a phone call.
-    virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream) = 0;
-    virtual status_t stopTone() = 0;
-
-    // set down link audio volume.
-    virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0;
-
-    // move effect to the specified output
-    virtual status_t moveEffects(int session,
-                                     audio_io_handle_t srcOutput,
-                                     audio_io_handle_t dstOutput) = 0;
-
-};
-
-extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface);
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface);
-
-
-}; // namespace android
-
-#endif // ANDROID_AUDIOPOLICYINTERFACE_H
diff --git a/audio/AudioPolicyManagerBase.cpp b/audio/AudioPolicyManagerBase.cpp
deleted file mode 100755
index ac31ad4..0000000
--- a/audio/AudioPolicyManagerBase.cpp
+++ /dev/null
@@ -1,2434 +0,0 @@
-/*
- * 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.
- */
-
-#define LOG_TAG "AudioPolicyManagerBase"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-#include <hardware_legacy/AudioPolicyManagerBase.h>
-#include <hardware/audio_effect.h>
-#include <math.h>
-
-namespace android_audio_legacy {
-
-// ----------------------------------------------------------------------------
-// AudioPolicyInterface implementation
-// ----------------------------------------------------------------------------
-
-
-status_t AudioPolicyManagerBase::setDeviceConnectionState(AudioSystem::audio_devices device,
-                                                  AudioSystem::device_connection_state state,
-                                                  const char *device_address)
-{
-
-    LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
-
-    // connect/disconnect only 1 device at a time
-    if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
-
-    if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
-        LOGE("setDeviceConnectionState() invalid address: %s", device_address);
-        return BAD_VALUE;
-    }
-
-    // handle output devices
-    if (AudioSystem::isOutputDevice(device)) {
-
-#ifndef WITH_A2DP
-        if (AudioSystem::isA2dpDevice(device)) {
-            LOGE("setDeviceConnectionState() invalid device: %x", device);
-            return BAD_VALUE;
-        }
-#endif
-
-        switch (state)
-        {
-        // handle output device connection
-        case AudioSystem::DEVICE_STATE_AVAILABLE:
-            if (mAvailableOutputDevices & device) {
-                LOGW("setDeviceConnectionState() device already connected: %x", device);
-                return INVALID_OPERATION;
-            }
-            LOGV("setDeviceConnectionState() connecting device %x", device);
-
-            // register new device as available
-            mAvailableOutputDevices |= device;
-
-#ifdef WITH_A2DP
-            // handle A2DP device connection
-            if (AudioSystem::isA2dpDevice(device)) {
-                status_t status = handleA2dpConnection(device, device_address);
-                if (status != NO_ERROR) {
-                    mAvailableOutputDevices &= ~device;
-                    return status;
-                }
-            } else
-#endif
-            {
-                if (AudioSystem::isBluetoothScoDevice(device)) {
-                    LOGV("setDeviceConnectionState() BT SCO  device, address %s", device_address);
-                    // keep track of SCO device address
-                    mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
-                }
-            }
-            break;
-        // handle output device disconnection
-        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
-            if (!(mAvailableOutputDevices & device)) {
-                LOGW("setDeviceConnectionState() device not connected: %x", device);
-                return INVALID_OPERATION;
-            }
-
-
-            LOGV("setDeviceConnectionState() disconnecting device %x", device);
-            // remove device from available output devices
-            mAvailableOutputDevices &= ~device;
-
-#ifdef WITH_A2DP
-            // handle A2DP device disconnection
-            if (AudioSystem::isA2dpDevice(device)) {
-                status_t status = handleA2dpDisconnection(device, device_address);
-                if (status != NO_ERROR) {
-                    mAvailableOutputDevices |= device;
-                    return status;
-                }
-            } else
-#endif
-            {
-                if (AudioSystem::isBluetoothScoDevice(device)) {
-                    mScoDeviceAddress = "";
-                }
-            }
-            } break;
-
-        default:
-            LOGE("setDeviceConnectionState() invalid state: %x", state);
-            return BAD_VALUE;
-        }
-
-        // request routing change if necessary
-        uint32_t newDevice = getNewDevice(mHardwareOutput, false);
-#ifdef WITH_A2DP
-        checkA2dpSuspend();
-        checkOutputForAllStrategies();
-        // A2DP outputs must be closed after checkOutputForAllStrategies() is executed
-        if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE && AudioSystem::isA2dpDevice(device)) {
-            closeA2dpOutputs();
-        }
-#endif
-        updateDeviceForStrategy();
-        setOutputDevice(mHardwareOutput, newDevice);
-
-        if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) {
-            device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
-        } else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO ||
-                   device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
-                   device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
-            device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
-        } else {
-            return NO_ERROR;
-        }
-    }
-    // handle input devices
-    if (AudioSystem::isInputDevice(device)) {
-
-        switch (state)
-        {
-        // handle input device connection
-        case AudioSystem::DEVICE_STATE_AVAILABLE: {
-            if (mAvailableInputDevices & device) {
-                LOGW("setDeviceConnectionState() device already connected: %d", device);
-                return INVALID_OPERATION;
-            }
-            mAvailableInputDevices |= device;
-            }
-            break;
-
-        // handle input device disconnection
-        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
-            if (!(mAvailableInputDevices & device)) {
-                LOGW("setDeviceConnectionState() device not connected: %d", device);
-                return INVALID_OPERATION;
-            }
-            mAvailableInputDevices &= ~device;
-            } break;
-
-        default:
-            LOGE("setDeviceConnectionState() invalid state: %x", state);
-            return BAD_VALUE;
-        }
-
-        audio_io_handle_t activeInput = getActiveInput();
-        if (activeInput != 0) {
-            AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
-            uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
-            if (newDevice != inputDesc->mDevice) {
-                LOGV("setDeviceConnectionState() changing device from %x to %x for input %d",
-                        inputDesc->mDevice, newDevice, activeInput);
-                inputDesc->mDevice = newDevice;
-                AudioParameter param = AudioParameter();
-                param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
-                mpClientInterface->setParameters(activeInput, param.toString());
-            }
-        }
-
-        return NO_ERROR;
-    }
-
-    LOGW("setDeviceConnectionState() invalid device: %x", device);
-    return BAD_VALUE;
-}
-
-AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(AudioSystem::audio_devices device,
-                                                  const char *device_address)
-{
-    AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
-    String8 address = String8(device_address);
-    if (AudioSystem::isOutputDevice(device)) {
-        if (device & mAvailableOutputDevices) {
-#ifdef WITH_A2DP
-            if (AudioSystem::isA2dpDevice(device) &&
-                address != "" && mA2dpDeviceAddress != address) {
-                return state;
-            }
-#endif
-            if (AudioSystem::isBluetoothScoDevice(device) &&
-                address != "" && mScoDeviceAddress != address) {
-                return state;
-            }
-            state = AudioSystem::DEVICE_STATE_AVAILABLE;
-        }
-    } else if (AudioSystem::isInputDevice(device)) {
-        if (device & mAvailableInputDevices) {
-            state = AudioSystem::DEVICE_STATE_AVAILABLE;
-        }
-    }
-
-    return state;
-}
-
-void AudioPolicyManagerBase::setPhoneState(int state)
-{
-    LOGV("setPhoneState() state %d", state);
-    uint32_t newDevice = 0;
-    if (state < 0 || state >= AudioSystem::NUM_MODES) {
-        LOGW("setPhoneState() invalid state %d", state);
-        return;
-    }
-
-    if (state == mPhoneState ) {
-        LOGW("setPhoneState() setting same state %d", state);
-        return;
-    }
-
-    // if leaving call state, handle special case of active streams
-    // pertaining to sonification strategy see handleIncallSonification()
-    if (isInCall()) {
-        LOGV("setPhoneState() in call state management: new state is %d", state);
-        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
-            handleIncallSonification(stream, false, true);
-        }
-    }
-
-    // store previous phone state for management of sonification strategy below
-    int oldState = mPhoneState;
-    mPhoneState = state;
-    bool force = false;
-
-    // are we entering or starting a call
-    if (!isStateInCall(oldState) && isStateInCall(state)) {
-        LOGV("  Entering call in setPhoneState()");
-        // force routing command to audio hardware when starting a call
-        // even if no device change is needed
-        force = true;
-    } else if (isStateInCall(oldState) && !isStateInCall(state)) {
-        LOGV("  Exiting call in setPhoneState()");
-        // force routing command to audio hardware when exiting a call
-        // even if no device change is needed
-        force = true;
-    } else if (isStateInCall(state) && (state != oldState)) {
-        LOGV("  Switching between telephony and VoIP in setPhoneState()");
-        // force routing command to audio hardware when switching between telephony and VoIP
-        // even if no device change is needed
-        force = true;
-    }
-
-    // check for device and output changes triggered by new phone state
-    newDevice = getNewDevice(mHardwareOutput, false);
-#ifdef WITH_A2DP
-    checkA2dpSuspend();
-    checkOutputForAllStrategies();
-#endif
-    updateDeviceForStrategy();
-
-    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
-
-    // force routing command to audio hardware when ending call
-    // even if no device change is needed
-    if (isStateInCall(oldState) && newDevice == 0) {
-        newDevice = hwOutputDesc->device();
-    }
-
-    // when changing from ring tone to in call mode, mute the ringing tone
-    // immediately and delay the route change to avoid sending the ring tone
-    // tail into the earpiece or headset.
-    int delayMs = 0;
-    if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) {
-        // delay the device change command by twice the output latency to have some margin
-        // and be sure that audio buffers not yet affected by the mute are out when
-        // we actually apply the route change
-        delayMs = hwOutputDesc->mLatency*2;
-        setStreamMute(AudioSystem::RING, true, mHardwareOutput);
-    }
-
-    // change routing is necessary
-    setOutputDevice(mHardwareOutput, newDevice, force, delayMs);
-
-    // if entering in call state, handle special case of active streams
-    // pertaining to sonification strategy see handleIncallSonification()
-    if (isStateInCall(state)) {
-        LOGV("setPhoneState() in call state management: new state is %d", state);
-        // unmute the ringing tone after a sufficient delay if it was muted before
-        // setting output device above
-        if (oldState == AudioSystem::MODE_RINGTONE) {
-            setStreamMute(AudioSystem::RING, false, mHardwareOutput, MUTE_TIME_MS);
-        }
-        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
-            handleIncallSonification(stream, true, true);
-        }
-    }
-
-    // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
-    if (state == AudioSystem::MODE_RINGTONE &&
-        isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
-        mLimitRingtoneVolume = true;
-    } else {
-        mLimitRingtoneVolume = false;
-    }
-}
-
-void AudioPolicyManagerBase::setRingerMode(uint32_t mode, uint32_t mask)
-{
-    LOGV("setRingerMode() mode %x, mask %x", mode, mask);
-
-    mRingerMode = mode;
-}
-
-void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
-{
-    LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
-
-    bool forceVolumeReeval = false;
-    switch(usage) {
-    case AudioSystem::FOR_COMMUNICATION:
-        if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
-            config != AudioSystem::FORCE_NONE) {
-            LOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
-            return;
-        }
-        forceVolumeReeval = true;
-        mForceUse[usage] = config;
-        break;
-    case AudioSystem::FOR_MEDIA:
-        if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
-            config != AudioSystem::FORCE_WIRED_ACCESSORY &&
-            config != AudioSystem::FORCE_ANALOG_DOCK &&
-            config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE) {
-            LOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
-            return;
-        }
-        mForceUse[usage] = config;
-        break;
-    case AudioSystem::FOR_RECORD:
-        if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY &&
-            config != AudioSystem::FORCE_NONE) {
-            LOGW("setForceUse() invalid config %d for FOR_RECORD", config);
-            return;
-        }
-        mForceUse[usage] = config;
-        break;
-    case AudioSystem::FOR_DOCK:
-        if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK &&
-            config != AudioSystem::FORCE_BT_DESK_DOCK &&
-            config != AudioSystem::FORCE_WIRED_ACCESSORY &&
-            config != AudioSystem::FORCE_ANALOG_DOCK &&
-            config != AudioSystem::FORCE_DIGITAL_DOCK) {
-            LOGW("setForceUse() invalid config %d for FOR_DOCK", config);
-        }
-        forceVolumeReeval = true;
-        mForceUse[usage] = config;
-        break;
-    default:
-        LOGW("setForceUse() invalid usage %d", usage);
-        break;
-    }
-
-    // check for device and output changes triggered by new phone state
-    uint32_t newDevice = getNewDevice(mHardwareOutput, false);
-#ifdef WITH_A2DP
-    checkA2dpSuspend();
-    checkOutputForAllStrategies();
-#endif
-    updateDeviceForStrategy();
-    setOutputDevice(mHardwareOutput, newDevice);
-    if (forceVolumeReeval) {
-        applyStreamVolumes(mHardwareOutput, newDevice, 0, true);
-    }
-
-    audio_io_handle_t activeInput = getActiveInput();
-    if (activeInput != 0) {
-        AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
-        newDevice = getDeviceForInputSource(inputDesc->mInputSource);
-        if (newDevice != inputDesc->mDevice) {
-            LOGV("setForceUse() changing device from %x to %x for input %d",
-                    inputDesc->mDevice, newDevice, activeInput);
-            inputDesc->mDevice = newDevice;
-            AudioParameter param = AudioParameter();
-            param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
-            mpClientInterface->setParameters(activeInput, param.toString());
-        }
-    }
-
-}
-
-AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
-{
-    return mForceUse[usage];
-}
-
-void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
-{
-    LOGV("setSystemProperty() property %s, value %s", property, value);
-    if (strcmp(property, "ro.camera.sound.forced") == 0) {
-        if (atoi(value)) {
-            LOGV("ENFORCED_AUDIBLE cannot be muted");
-            mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
-        } else {
-            LOGV("ENFORCED_AUDIBLE can be muted");
-            mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
-        }
-    }
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
-                                    uint32_t samplingRate,
-                                    uint32_t format,
-                                    uint32_t channels,
-                                    AudioSystem::output_flags flags)
-{
-    audio_io_handle_t output = 0;
-    uint32_t latency = 0;
-    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
-    uint32_t device = getDeviceForStrategy(strategy);
-    LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
-
-#ifdef AUDIO_POLICY_TEST
-    if (mCurOutput != 0) {
-        LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
-                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
-
-        if (mTestOutputs[mCurOutput] == 0) {
-            LOGV("getOutput() opening test output");
-            AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
-            outputDesc->mDevice = mTestDevice;
-            outputDesc->mSamplingRate = mTestSamplingRate;
-            outputDesc->mFormat = mTestFormat;
-            outputDesc->mChannels = mTestChannels;
-            outputDesc->mLatency = mTestLatencyMs;
-            outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
-            outputDesc->mRefCount[stream] = 0;
-            mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
-                                            &outputDesc->mSamplingRate,
-                                            &outputDesc->mFormat,
-                                            &outputDesc->mChannels,
-                                            &outputDesc->mLatency,
-                                            outputDesc->mFlags);
-            if (mTestOutputs[mCurOutput]) {
-                AudioParameter outputCmd = AudioParameter();
-                outputCmd.addInt(String8("set_id"),mCurOutput);
-                mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
-                addOutput(mTestOutputs[mCurOutput], outputDesc);
-            }
-        }
-        return mTestOutputs[mCurOutput];
-    }
-#endif //AUDIO_POLICY_TEST
-
-    // open a direct output if required by specified parameters
-    if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) {
-
-        LOGV("getOutput() opening direct output device %x", device);
-        AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
-        outputDesc->mDevice = device;
-        outputDesc->mSamplingRate = samplingRate;
-        outputDesc->mFormat = format;
-        outputDesc->mChannels = channels;
-        outputDesc->mLatency = 0;
-        outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT);
-        outputDesc->mRefCount[stream] = 0;
-        outputDesc->mStopTime[stream] = 0;
-        output = mpClientInterface->openOutput(&outputDesc->mDevice,
-                                        &outputDesc->mSamplingRate,
-                                        &outputDesc->mFormat,
-                                        &outputDesc->mChannels,
-                                        &outputDesc->mLatency,
-                                        outputDesc->mFlags);
-
-        // only accept an output with the requeted parameters
-        if (output == 0 ||
-            (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
-            (format != 0 && format != outputDesc->mFormat) ||
-            (channels != 0 && channels != outputDesc->mChannels)) {
-            LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d",
-                    samplingRate, format, channels);
-            if (output != 0) {
-                mpClientInterface->closeOutput(output);
-            }
-            delete outputDesc;
-            return 0;
-        }
-        addOutput(output, outputDesc);
-        return output;
-    }
-
-    if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO &&
-        channels != AudioSystem::CHANNEL_OUT_STEREO) {
-        return 0;
-    }
-    // open a non direct output
-
-    // get which output is suitable for the specified stream. The actual routing change will happen
-    // when startOutput() will be called
-    uint32_t a2dpDevice = device & AudioSystem::DEVICE_OUT_ALL_A2DP;
-    if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) {
-#ifdef WITH_A2DP
-        if (a2dpUsedForSonification() && a2dpDevice != 0) {
-            // if playing on 2 devices among which one is A2DP, use duplicated output
-            LOGV("getOutput() using duplicated output");
-            LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device);
-            output = mDuplicatedOutput;
-        } else
-#endif
-        {
-            // if playing on 2 devices among which none is A2DP, use hardware output
-            output = mHardwareOutput;
-        }
-        LOGV("getOutput() using output %d for 2 devices %x", output, device);
-    } else {
-#ifdef WITH_A2DP
-        if (a2dpDevice != 0) {
-            // if playing on A2DP device, use a2dp output
-            LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device);
-            output = mA2dpOutput;
-        } else
-#endif
-        {
-            // if playing on not A2DP device, use hardware output
-            output = mHardwareOutput;
-        }
-    }
-
-
-    LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
-                stream, samplingRate, format, channels, flags);
-
-    return output;
-}
-
-status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
-                                             AudioSystem::stream_type stream,
-                                             int session)
-{
-    LOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        LOGW("startOutput() unknow output %d", output);
-        return BAD_VALUE;
-    }
-
-    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
-
-#ifdef WITH_A2DP
-    if (mA2dpOutput != 0  && !a2dpUsedForSonification() &&
-            (strategy == STRATEGY_SONIFICATION || strategy == STRATEGY_ENFORCED_AUDIBLE)) {
-        setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
-    }
-#endif
-
-    // incremenent usage count for this stream on the requested output:
-    // NOTE that the usage count is the same for duplicated output and hardware output which is
-    // necassary for a correct control of hardware output routing by startOutput() and stopOutput()
-    outputDesc->changeRefCount(stream, 1);
-
-    uint32_t prevDevice = outputDesc->mDevice;
-    setOutputDevice(output, getNewDevice(output));
-
-    // handle special case for sonification while in call
-    if (isInCall()) {
-        handleIncallSonification(stream, true, false);
-    }
-
-    // apply volume rules for current stream and device if necessary
-    checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device());
-
-    // FIXME: need a delay to make sure that audio path switches to speaker before sound
-    // starts. Should be platform specific?
-    if (stream == AudioSystem::ENFORCED_AUDIBLE &&
-            prevDevice != outputDesc->mDevice) {
-        usleep(outputDesc->mLatency*4*1000);
-    }
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
-                                            AudioSystem::stream_type stream,
-                                            int session)
-{
-    LOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        LOGW("stopOutput() unknow output %d", output);
-        return BAD_VALUE;
-    }
-
-    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
-
-    // handle special case for sonification while in call
-    if (isInCall()) {
-        handleIncallSonification(stream, false, false);
-    }
-
-    if (outputDesc->mRefCount[stream] > 0) {
-        // decrement usage count of this stream on the output
-        outputDesc->changeRefCount(stream, -1);
-        // store time at which the stream was stopped - see isStreamActive()
-        outputDesc->mStopTime[stream] = systemTime();
-
-        setOutputDevice(output, getNewDevice(output), false, outputDesc->mLatency*2);
-
-#ifdef WITH_A2DP
-        if (mA2dpOutput != 0 && !a2dpUsedForSonification() &&
-                (strategy == STRATEGY_SONIFICATION || strategy == STRATEGY_ENFORCED_AUDIBLE)) {
-            setStrategyMute(STRATEGY_MEDIA,
-                            false,
-                            mA2dpOutput,
-                            mOutputs.valueFor(mHardwareOutput)->mLatency*2);
-        }
-#endif
-        if (output != mHardwareOutput) {
-            setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true);
-        }
-        return NO_ERROR;
-    } else {
-        LOGW("stopOutput() refcount is already 0 for output %d", output);
-        return INVALID_OPERATION;
-    }
-}
-
-void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
-{
-    LOGV("releaseOutput() %d", output);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        LOGW("releaseOutput() releasing unknown output %d", output);
-        return;
-    }
-
-#ifdef AUDIO_POLICY_TEST
-    int testIndex = testOutputIndex(output);
-    if (testIndex != 0) {
-        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-        if (outputDesc->refCount() == 0) {
-            mpClientInterface->closeOutput(output);
-            delete mOutputs.valueAt(index);
-            mOutputs.removeItem(output);
-            mTestOutputs[testIndex] = 0;
-        }
-        return;
-    }
-#endif //AUDIO_POLICY_TEST
-
-    if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
-        mpClientInterface->closeOutput(output);
-        delete mOutputs.valueAt(index);
-        mOutputs.removeItem(output);
-    }
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
-                                    uint32_t samplingRate,
-                                    uint32_t format,
-                                    uint32_t channels,
-                                    AudioSystem::audio_in_acoustics acoustics)
-{
-    audio_io_handle_t input = 0;
-    uint32_t device = getDeviceForInputSource(inputSource);
-
-    LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
-
-    if (device == 0) {
-        return 0;
-    }
-
-    // adapt channel selection to input source
-    switch(inputSource) {
-    case AUDIO_SOURCE_VOICE_UPLINK:
-        channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
-        break;
-    case AUDIO_SOURCE_VOICE_DOWNLINK:
-        channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
-        break;
-    case AUDIO_SOURCE_VOICE_CALL:
-        channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
-        break;
-    default:
-        break;
-    }
-
-    AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
-
-    inputDesc->mInputSource = inputSource;
-    inputDesc->mDevice = device;
-    inputDesc->mSamplingRate = samplingRate;
-    inputDesc->mFormat = format;
-    inputDesc->mChannels = channels;
-    inputDesc->mAcoustics = acoustics;
-    inputDesc->mRefCount = 0;
-    input = mpClientInterface->openInput(&inputDesc->mDevice,
-                                    &inputDesc->mSamplingRate,
-                                    &inputDesc->mFormat,
-                                    &inputDesc->mChannels,
-                                    inputDesc->mAcoustics);
-
-    // only accept input with the exact requested set of parameters
-    if (input == 0 ||
-        (samplingRate != inputDesc->mSamplingRate) ||
-        (format != inputDesc->mFormat) ||
-        (channels != inputDesc->mChannels)) {
-        LOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d",
-                samplingRate, format, channels);
-        if (input != 0) {
-            mpClientInterface->closeInput(input);
-        }
-        delete inputDesc;
-        return 0;
-    }
-    mInputs.add(input, inputDesc);
-    return input;
-}
-
-status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
-{
-    LOGV("startInput() input %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        LOGW("startInput() unknow input %d", input);
-        return BAD_VALUE;
-    }
-    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-#ifdef AUDIO_POLICY_TEST
-    if (mTestInput == 0)
-#endif //AUDIO_POLICY_TEST
-    {
-        // refuse 2 active AudioRecord clients at the same time
-        if (getActiveInput() != 0) {
-            LOGW("startInput() input %d failed: other input already started", input);
-            return INVALID_OPERATION;
-        }
-    }
-
-    AudioParameter param = AudioParameter();
-    param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
-
-    param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource);
-    LOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
-
-    mpClientInterface->setParameters(input, param.toString());
-
-    inputDesc->mRefCount = 1;
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
-{
-    LOGV("stopInput() input %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        LOGW("stopInput() unknow input %d", input);
-        return BAD_VALUE;
-    }
-    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-    if (inputDesc->mRefCount == 0) {
-        LOGW("stopInput() input %d already stopped", input);
-        return INVALID_OPERATION;
-    } else {
-        AudioParameter param = AudioParameter();
-        param.addInt(String8(AudioParameter::keyRouting), 0);
-        mpClientInterface->setParameters(input, param.toString());
-        inputDesc->mRefCount = 0;
-        return NO_ERROR;
-    }
-}
-
-void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input)
-{
-    LOGV("releaseInput() %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        LOGW("releaseInput() releasing unknown input %d", input);
-        return;
-    }
-    mpClientInterface->closeInput(input);
-    delete mInputs.valueAt(index);
-    mInputs.removeItem(input);
-    LOGV("releaseInput() exit");
-}
-
-void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream,
-                                            int indexMin,
-                                            int indexMax)
-{
-    LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
-    if (indexMin < 0 || indexMin >= indexMax) {
-        LOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
-        return;
-    }
-    mStreams[stream].mIndexMin = indexMin;
-    mStreams[stream].mIndexMax = indexMax;
-}
-
-status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
-{
-
-    if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
-        return BAD_VALUE;
-    }
-
-    // Force max volume if stream cannot be muted
-    if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
-
-    LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
-    mStreams[stream].mIndexCur = index;
-
-    // compute and apply stream volume on all outputs according to connected device
-    status_t status = NO_ERROR;
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), mOutputs.valueAt(i)->device());
-        if (volStatus != NO_ERROR) {
-            status = volStatus;
-        }
-    }
-    return status;
-}
-
-status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
-{
-    if (index == 0) {
-        return BAD_VALUE;
-    }
-    LOGV("getStreamVolumeIndex() stream %d", stream);
-    *index =  mStreams[stream].mIndexCur;
-    return NO_ERROR;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(effect_descriptor_t *desc)
-{
-    LOGV("getOutputForEffect()");
-    // apply simple rule where global effects are attached to the same output as MUSIC streams
-    return getOutput(AudioSystem::MUSIC);
-}
-
-status_t AudioPolicyManagerBase::registerEffect(effect_descriptor_t *desc,
-                                audio_io_handle_t io,
-                                uint32_t strategy,
-                                int session,
-                                int id)
-{
-    ssize_t index = mOutputs.indexOfKey(io);
-    if (index < 0) {
-        index = mInputs.indexOfKey(io);
-        if (index < 0) {
-            LOGW("registerEffect() unknown io %d", io);
-            return INVALID_OPERATION;
-        }
-    }
-
-    if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
-        LOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
-                desc->name, desc->memoryUsage);
-        return INVALID_OPERATION;
-    }
-    mTotalEffectsMemory += desc->memoryUsage;
-    LOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
-            desc->name, io, strategy, session, id);
-    LOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
-
-    EffectDescriptor *pDesc = new EffectDescriptor();
-    memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
-    pDesc->mIo = io;
-    pDesc->mStrategy = (routing_strategy)strategy;
-    pDesc->mSession = session;
-    pDesc->mEnabled = false;
-
-    mEffects.add(id, pDesc);
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::unregisterEffect(int id)
-{
-    ssize_t index = mEffects.indexOfKey(id);
-    if (index < 0) {
-        LOGW("unregisterEffect() unknown effect ID %d", id);
-        return INVALID_OPERATION;
-    }
-
-    EffectDescriptor *pDesc = mEffects.valueAt(index);
-
-    setEffectEnabled(pDesc, false);
-
-    if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
-        LOGW("unregisterEffect() memory %d too big for total %d",
-                pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
-        pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
-    }
-    mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
-    LOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
-            pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
-
-    mEffects.removeItem(id);
-    delete pDesc;
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::setEffectEnabled(int id, bool enabled)
-{
-    ssize_t index = mEffects.indexOfKey(id);
-    if (index < 0) {
-        LOGW("unregisterEffect() unknown effect ID %d", id);
-        return INVALID_OPERATION;
-    }
-
-    return setEffectEnabled(mEffects.valueAt(index), enabled);
-}
-
-status_t AudioPolicyManagerBase::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
-{
-    if (enabled == pDesc->mEnabled) {
-        LOGV("setEffectEnabled(%s) effect already %s",
-             enabled?"true":"false", enabled?"enabled":"disabled");
-        return INVALID_OPERATION;
-    }
-
-    if (enabled) {
-        if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
-            LOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
-                 pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
-            return INVALID_OPERATION;
-        }
-        mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
-        LOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
-    } else {
-        if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
-            LOGW("setEffectEnabled(false) CPU load %d too high for total %d",
-                    pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
-            pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
-        }
-        mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
-        LOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
-    }
-    pDesc->mEnabled = enabled;
-    return NO_ERROR;
-}
-
-bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const
-{
-    nsecs_t sysTime = systemTime();
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        if (mOutputs.valueAt(i)->mRefCount[stream] != 0 ||
-            ns2ms(sysTime - mOutputs.valueAt(i)->mStopTime[stream]) < inPastMs) {
-            return true;
-        }
-    }
-    return false;
-}
-
-
-status_t AudioPolicyManagerBase::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Hardware Output: %d\n", mHardwareOutput);
-    result.append(buffer);
-#ifdef WITH_A2DP
-    snprintf(buffer, SIZE, " A2DP Output: %d\n", mA2dpOutput);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Duplicated Output: %d\n", mDuplicatedOutput);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
-    result.append(buffer);
-#endif
-    snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Ringer mode: %d\n", mRingerMode);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-    snprintf(buffer, SIZE, "\nOutputs dump:\n");
-    write(fd, buffer, strlen(buffer));
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
-        write(fd, buffer, strlen(buffer));
-        mOutputs.valueAt(i)->dump(fd);
-    }
-
-    snprintf(buffer, SIZE, "\nInputs dump:\n");
-    write(fd, buffer, strlen(buffer));
-    for (size_t i = 0; i < mInputs.size(); i++) {
-        snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
-        write(fd, buffer, strlen(buffer));
-        mInputs.valueAt(i)->dump(fd);
-    }
-
-    snprintf(buffer, SIZE, "\nStreams dump:\n");
-    write(fd, buffer, strlen(buffer));
-    snprintf(buffer, SIZE, " Stream  Index Min  Index Max  Index Cur  Can be muted\n");
-    write(fd, buffer, strlen(buffer));
-    for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
-        snprintf(buffer, SIZE, " %02d", i);
-        mStreams[i].dump(buffer + 3, SIZE);
-        write(fd, buffer, strlen(buffer));
-    }
-
-    snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
-            (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
-    write(fd, buffer, strlen(buffer));
-
-    snprintf(buffer, SIZE, "Registered effects:\n");
-    write(fd, buffer, strlen(buffer));
-    for (size_t i = 0; i < mEffects.size(); i++) {
-        snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
-        write(fd, buffer, strlen(buffer));
-        mEffects.valueAt(i)->dump(fd);
-    }
-
-
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManagerBase
-// ----------------------------------------------------------------------------
-
-AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
-    :
-#ifdef AUDIO_POLICY_TEST
-    Thread(false),
-#endif //AUDIO_POLICY_TEST
-    mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0),
-    mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
-    mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
-    mA2dpSuspended(false)
-{
-    mpClientInterface = clientInterface;
-
-    for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
-        mForceUse[i] = AudioSystem::FORCE_NONE;
-    }
-
-    initializeVolumeCurves();
-
-    // devices available by default are speaker, ear piece and microphone
-    mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE |
-                        AudioSystem::DEVICE_OUT_SPEAKER;
-    mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
-
-#ifdef WITH_A2DP
-    mA2dpOutput = 0;
-    mDuplicatedOutput = 0;
-    mA2dpDeviceAddress = String8("");
-#endif
-    mScoDeviceAddress = String8("");
-
-    // open hardware output
-    AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
-    outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
-    mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
-                                    &outputDesc->mSamplingRate,
-                                    &outputDesc->mFormat,
-                                    &outputDesc->mChannels,
-                                    &outputDesc->mLatency,
-                                    outputDesc->mFlags);
-
-    if (mHardwareOutput == 0) {
-        LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
-                outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
-    } else {
-        addOutput(mHardwareOutput, outputDesc);
-        setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true);
-        //TODO: configure audio effect output stage here
-    }
-
-    updateDeviceForStrategy();
-#ifdef AUDIO_POLICY_TEST
-    if (mHardwareOutput != 0) {
-        AudioParameter outputCmd = AudioParameter();
-        outputCmd.addInt(String8("set_id"), 0);
-        mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
-
-        mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
-        mTestSamplingRate = 44100;
-        mTestFormat = AudioSystem::PCM_16_BIT;
-        mTestChannels =  AudioSystem::CHANNEL_OUT_STEREO;
-        mTestLatencyMs = 0;
-        mCurOutput = 0;
-        mDirectOutput = false;
-        for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
-            mTestOutputs[i] = 0;
-        }
-
-        const size_t SIZE = 256;
-        char buffer[SIZE];
-        snprintf(buffer, SIZE, "AudioPolicyManagerTest");
-        run(buffer, ANDROID_PRIORITY_AUDIO);
-    }
-#endif //AUDIO_POLICY_TEST
-}
-
-AudioPolicyManagerBase::~AudioPolicyManagerBase()
-{
-#ifdef AUDIO_POLICY_TEST
-    exit();
-#endif //AUDIO_POLICY_TEST
-   for (size_t i = 0; i < mOutputs.size(); i++) {
-        mpClientInterface->closeOutput(mOutputs.keyAt(i));
-        delete mOutputs.valueAt(i);
-   }
-   mOutputs.clear();
-   for (size_t i = 0; i < mInputs.size(); i++) {
-        mpClientInterface->closeInput(mInputs.keyAt(i));
-        delete mInputs.valueAt(i);
-   }
-   mInputs.clear();
-}
-
-status_t AudioPolicyManagerBase::initCheck()
-{
-    return (mHardwareOutput == 0) ? NO_INIT : NO_ERROR;
-}
-
-#ifdef AUDIO_POLICY_TEST
-bool AudioPolicyManagerBase::threadLoop()
-{
-    LOGV("entering threadLoop()");
-    while (!exitPending())
-    {
-        String8 command;
-        int valueInt;
-        String8 value;
-
-        Mutex::Autolock _l(mLock);
-        mWaitWorkCV.waitRelative(mLock, milliseconds(50));
-
-        command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
-        AudioParameter param = AudioParameter(command);
-
-        if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
-            valueInt != 0) {
-            LOGV("Test command %s received", command.string());
-            String8 target;
-            if (param.get(String8("target"), target) != NO_ERROR) {
-                target = "Manager";
-            }
-            if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
-                param.remove(String8("test_cmd_policy_output"));
-                mCurOutput = valueInt;
-            }
-            if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
-                param.remove(String8("test_cmd_policy_direct"));
-                if (value == "false") {
-                    mDirectOutput = false;
-                } else if (value == "true") {
-                    mDirectOutput = true;
-                }
-            }
-            if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
-                param.remove(String8("test_cmd_policy_input"));
-                mTestInput = valueInt;
-            }
-
-            if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
-                param.remove(String8("test_cmd_policy_format"));
-                int format = AudioSystem::INVALID_FORMAT;
-                if (value == "PCM 16 bits") {
-                    format = AudioSystem::PCM_16_BIT;
-                } else if (value == "PCM 8 bits") {
-                    format = AudioSystem::PCM_8_BIT;
-                } else if (value == "Compressed MP3") {
-                    format = AudioSystem::MP3;
-                }
-                if (format != AudioSystem::INVALID_FORMAT) {
-                    if (target == "Manager") {
-                        mTestFormat = format;
-                    } else if (mTestOutputs[mCurOutput] != 0) {
-                        AudioParameter outputParam = AudioParameter();
-                        outputParam.addInt(String8("format"), format);
-                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
-                    }
-                }
-            }
-            if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
-                param.remove(String8("test_cmd_policy_channels"));
-                int channels = 0;
-
-                if (value == "Channels Stereo") {
-                    channels =  AudioSystem::CHANNEL_OUT_STEREO;
-                } else if (value == "Channels Mono") {
-                    channels =  AudioSystem::CHANNEL_OUT_MONO;
-                }
-                if (channels != 0) {
-                    if (target == "Manager") {
-                        mTestChannels = channels;
-                    } else if (mTestOutputs[mCurOutput] != 0) {
-                        AudioParameter outputParam = AudioParameter();
-                        outputParam.addInt(String8("channels"), channels);
-                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
-                    }
-                }
-            }
-            if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
-                param.remove(String8("test_cmd_policy_sampleRate"));
-                if (valueInt >= 0 && valueInt <= 96000) {
-                    int samplingRate = valueInt;
-                    if (target == "Manager") {
-                        mTestSamplingRate = samplingRate;
-                    } else if (mTestOutputs[mCurOutput] != 0) {
-                        AudioParameter outputParam = AudioParameter();
-                        outputParam.addInt(String8("sampling_rate"), samplingRate);
-                        mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
-                    }
-                }
-            }
-
-            if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
-                param.remove(String8("test_cmd_policy_reopen"));
-
-                mpClientInterface->closeOutput(mHardwareOutput);
-                delete mOutputs.valueFor(mHardwareOutput);
-                mOutputs.removeItem(mHardwareOutput);
-
-                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
-                outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
-                mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
-                                                &outputDesc->mSamplingRate,
-                                                &outputDesc->mFormat,
-                                                &outputDesc->mChannels,
-                                                &outputDesc->mLatency,
-                                                outputDesc->mFlags);
-                if (mHardwareOutput == 0) {
-                    LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
-                            outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
-                } else {
-                    AudioParameter outputCmd = AudioParameter();
-                    outputCmd.addInt(String8("set_id"), 0);
-                    mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
-                    addOutput(mHardwareOutput, outputDesc);
-                }
-            }
-
-
-            mpClientInterface->setParameters(0, String8("test_cmd_policy="));
-        }
-    }
-    return false;
-}
-
-void AudioPolicyManagerBase::exit()
-{
-    {
-        AutoMutex _l(mLock);
-        requestExit();
-        mWaitWorkCV.signal();
-    }
-    requestExitAndWait();
-}
-
-int AudioPolicyManagerBase::testOutputIndex(audio_io_handle_t output)
-{
-    for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
-        if (output == mTestOutputs[i]) return i;
-    }
-    return 0;
-}
-#endif //AUDIO_POLICY_TEST
-
-// ---
-
-void AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
-{
-    outputDesc->mId = id;
-    mOutputs.add(id, outputDesc);
-}
-
-
-#ifdef WITH_A2DP
-status_t AudioPolicyManagerBase::handleA2dpConnection(AudioSystem::audio_devices device,
-                                                 const char *device_address)
-{
-    // when an A2DP device is connected, open an A2DP and a duplicated output
-    LOGV("opening A2DP output for device %s", device_address);
-    AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
-    outputDesc->mDevice = device;
-    mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
-                                            &outputDesc->mSamplingRate,
-                                            &outputDesc->mFormat,
-                                            &outputDesc->mChannels,
-                                            &outputDesc->mLatency,
-                                            outputDesc->mFlags);
-    if (mA2dpOutput) {
-        // add A2DP output descriptor
-        addOutput(mA2dpOutput, outputDesc);
-
-        //TODO: configure audio effect output stage here
-
-        // set initial stream volume for A2DP device
-        applyStreamVolumes(mA2dpOutput, device);
-        if (a2dpUsedForSonification()) {
-            mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput);
-        }
-        if (mDuplicatedOutput != 0 ||
-            !a2dpUsedForSonification()) {
-            // If both A2DP and duplicated outputs are open, send device address to A2DP hardware
-            // interface
-            AudioParameter param;
-            param.add(String8("a2dp_sink_address"), String8(device_address));
-            mpClientInterface->setParameters(mA2dpOutput, param.toString());
-            mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
-
-            if (a2dpUsedForSonification()) {
-                // add duplicated output descriptor
-                AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor();
-                dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput);
-                dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput);
-                dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate;
-                dupOutputDesc->mFormat = outputDesc->mFormat;
-                dupOutputDesc->mChannels = outputDesc->mChannels;
-                dupOutputDesc->mLatency = outputDesc->mLatency;
-                addOutput(mDuplicatedOutput, dupOutputDesc);
-                applyStreamVolumes(mDuplicatedOutput, device);
-            }
-        } else {
-            LOGW("getOutput() could not open duplicated output for %d and %d",
-                    mHardwareOutput, mA2dpOutput);
-            mpClientInterface->closeOutput(mA2dpOutput);
-            mOutputs.removeItem(mA2dpOutput);
-            mA2dpOutput = 0;
-            delete outputDesc;
-            return NO_INIT;
-        }
-    } else {
-        LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device);
-        delete outputDesc;
-        return NO_INIT;
-    }
-    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
-
-    if (!a2dpUsedForSonification()) {
-        // mute music on A2DP output if a notification or ringtone is playing
-        uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION);
-        refCount += hwOutputDesc->strategyRefCount(STRATEGY_ENFORCED_AUDIBLE);
-        for (uint32_t i = 0; i < refCount; i++) {
-            setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
-        }
-    }
-
-    mA2dpSuspended = false;
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::handleA2dpDisconnection(AudioSystem::audio_devices device,
-                                                    const char *device_address)
-{
-    if (mA2dpOutput == 0) {
-        LOGW("setDeviceConnectionState() disconnecting A2DP and no A2DP output!");
-        return INVALID_OPERATION;
-    }
-
-    if (mA2dpDeviceAddress != device_address) {
-        LOGW("setDeviceConnectionState() disconnecting unknow A2DP sink address %s", device_address);
-        return INVALID_OPERATION;
-    }
-
-    // mute media strategy to avoid outputting sound on hardware output while music stream
-    // is switched from A2DP output and before music is paused by music application
-    setStrategyMute(STRATEGY_MEDIA, true, mHardwareOutput);
-    setStrategyMute(STRATEGY_MEDIA, false, mHardwareOutput, MUTE_TIME_MS);
-
-    if (!a2dpUsedForSonification()) {
-        // unmute music on A2DP output if a notification or ringtone is playing
-        uint32_t refCount = mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_SONIFICATION);
-        refCount += mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_ENFORCED_AUDIBLE);
-        for (uint32_t i = 0; i < refCount; i++) {
-            setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput);
-        }
-    }
-    mA2dpDeviceAddress = "";
-    mA2dpSuspended = false;
-    return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::closeA2dpOutputs()
-{
-
-    LOGV("setDeviceConnectionState() closing A2DP and duplicated output!");
-
-    if (mDuplicatedOutput != 0) {
-        AudioOutputDescriptor *dupOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
-        AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
-        // As all active tracks on duplicated output will be deleted,
-        // and as they were also referenced on hardware output, the reference
-        // count for their stream type must be adjusted accordingly on
-        // hardware output.
-        for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
-            int refCount = dupOutputDesc->mRefCount[i];
-            hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
-        }
-
-        mpClientInterface->closeOutput(mDuplicatedOutput);
-        delete mOutputs.valueFor(mDuplicatedOutput);
-        mOutputs.removeItem(mDuplicatedOutput);
-        mDuplicatedOutput = 0;
-    }
-    if (mA2dpOutput != 0) {
-        AudioParameter param;
-        param.add(String8("closing"), String8("true"));
-        mpClientInterface->setParameters(mA2dpOutput, param.toString());
-
-        mpClientInterface->closeOutput(mA2dpOutput);
-        delete mOutputs.valueFor(mA2dpOutput);
-        mOutputs.removeItem(mA2dpOutput);
-        mA2dpOutput = 0;
-    }
-}
-
-void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy)
-{
-    uint32_t prevDevice = getDeviceForStrategy(strategy);
-    uint32_t curDevice = getDeviceForStrategy(strategy, false);
-    bool a2dpWasUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(prevDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
-    bool a2dpIsUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(curDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
-    audio_io_handle_t srcOutput = 0;
-    audio_io_handle_t dstOutput = 0;
-
-    if (a2dpWasUsed && !a2dpIsUsed) {
-        bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2);
-        dstOutput = mHardwareOutput;
-        if (dupUsed) {
-            LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy);
-            srcOutput = mDuplicatedOutput;
-        } else {
-            LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy);
-            srcOutput = mA2dpOutput;
-        }
-    }
-    if (a2dpIsUsed && !a2dpWasUsed) {
-        bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2);
-        srcOutput = mHardwareOutput;
-        if (dupUsed) {
-            LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy);
-            dstOutput = mDuplicatedOutput;
-        } else {
-            LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy);
-            dstOutput = mA2dpOutput;
-        }
-    }
-
-    if (srcOutput != 0 && dstOutput != 0) {
-        // Move effects associated to this strategy from previous output to new output
-        for (size_t i = 0; i < mEffects.size(); i++) {
-            EffectDescriptor *desc = mEffects.valueAt(i);
-            if (desc->mSession != AudioSystem::SESSION_OUTPUT_STAGE &&
-                    desc->mStrategy == strategy &&
-                    desc->mIo == srcOutput) {
-                LOGV("checkOutputForStrategy() moving effect %d to output %d", mEffects.keyAt(i), dstOutput);
-                mpClientInterface->moveEffects(desc->mSession, srcOutput, dstOutput);
-                desc->mIo = dstOutput;
-            }
-        }
-        // Move tracks associated to this strategy from previous output to new output
-        for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
-            if (getStrategy((AudioSystem::stream_type)i) == strategy) {
-                mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, dstOutput);
-            }
-        }
-    }
-}
-
-void AudioPolicyManagerBase::checkOutputForAllStrategies()
-{
-    checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
-    checkOutputForStrategy(STRATEGY_PHONE);
-    checkOutputForStrategy(STRATEGY_SONIFICATION);
-    checkOutputForStrategy(STRATEGY_MEDIA);
-    checkOutputForStrategy(STRATEGY_DTMF);
-}
-
-void AudioPolicyManagerBase::checkA2dpSuspend()
-{
-    // suspend A2DP output if:
-    //      (NOT already suspended) &&
-    //      ((SCO device is connected &&
-    //       (forced usage for communication || for record is SCO))) ||
-    //      (phone state is ringing || in call)
-    //
-    // restore A2DP output if:
-    //      (Already suspended) &&
-    //      ((SCO device is NOT connected ||
-    //       (forced usage NOT for communication && NOT for record is SCO))) &&
-    //      (phone state is NOT ringing && NOT in call)
-    //
-    if (mA2dpOutput == 0) {
-        return;
-    }
-
-    if (mA2dpSuspended) {
-        if (((mScoDeviceAddress == "") ||
-             ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) &&
-              (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) &&
-             ((mPhoneState != AudioSystem::MODE_IN_CALL) &&
-              (mPhoneState != AudioSystem::MODE_RINGTONE))) {
-
-            mpClientInterface->restoreOutput(mA2dpOutput);
-            mA2dpSuspended = false;
-        }
-    } else {
-        if (((mScoDeviceAddress != "") &&
-             ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
-              (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) ||
-             ((mPhoneState == AudioSystem::MODE_IN_CALL) ||
-              (mPhoneState == AudioSystem::MODE_RINGTONE))) {
-
-            mpClientInterface->suspendOutput(mA2dpOutput);
-            mA2dpSuspended = true;
-        }
-    }
-}
-
-
-#endif
-
-uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
-{
-    uint32_t device = 0;
-
-    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-    // check the following by order of priority to request a routing change if necessary:
-    // 1: the strategy enforced audible is active on the output:
-    //      use device for strategy enforced audible
-    // 2: we are in call or the strategy phone is active on the output:
-    //      use device for strategy phone
-    // 3: the strategy sonification is active on the output:
-    //      use device for strategy sonification
-    // 4: the strategy media is active on the output:
-    //      use device for strategy media
-    // 5: the strategy DTMF is active on the output:
-    //      use device for strategy DTMF
-    if (outputDesc->isUsedByStrategy(STRATEGY_ENFORCED_AUDIBLE)) {
-        device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
-    } else if (isInCall() ||
-                    outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
-        device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
-    } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
-        device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
-    } else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) {
-        device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
-    } else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) {
-        device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
-    }
-
-    LOGV("getNewDevice() selected device %x", device);
-    return device;
-}
-
-uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) {
-    return (uint32_t)getStrategy(stream);
-}
-
-uint32_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
-    uint32_t devices;
-    // By checking the range of stream before calling getStrategy, we avoid
-    // getStrategy's behavior for invalid streams.  getStrategy would do a LOGE
-    // and then return STRATEGY_MEDIA, but we want to return the empty set.
-    if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
-        devices = 0;
-    } else {
-        AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
-        devices = getDeviceForStrategy(strategy, true);
-    }
-    return devices;
-}
-
-AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(
-        AudioSystem::stream_type stream) {
-    // stream to strategy mapping
-    switch (stream) {
-    case AudioSystem::VOICE_CALL:
-    case AudioSystem::BLUETOOTH_SCO:
-        return STRATEGY_PHONE;
-    case AudioSystem::RING:
-    case AudioSystem::NOTIFICATION:
-    case AudioSystem::ALARM:
-        return STRATEGY_SONIFICATION;
-    case AudioSystem::DTMF:
-        return STRATEGY_DTMF;
-    default:
-        LOGE("unknown stream type");
-    case AudioSystem::SYSTEM:
-        // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
-        // while key clicks are played produces a poor result
-    case AudioSystem::TTS:
-    case AudioSystem::MUSIC:
-        return STRATEGY_MEDIA;
-    case AudioSystem::ENFORCED_AUDIBLE:
-        return STRATEGY_ENFORCED_AUDIBLE;
-    }
-}
-
-uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, bool fromCache)
-{
-    uint32_t device = 0;
-
-    if (fromCache) {
-        LOGV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]);
-        return mDeviceForStrategy[strategy];
-    }
-
-    switch (strategy) {
-    case STRATEGY_DTMF:
-        if (!isInCall()) {
-            // when off call, DTMF strategy follows the same rules as MEDIA strategy
-            device = getDeviceForStrategy(STRATEGY_MEDIA, false);
-            break;
-        }
-        // when in call, DTMF and PHONE strategies follow the same rules
-        // FALL THROUGH
-
-    case STRATEGY_PHONE:
-        // for phone strategy, we first consider the forced use and then the available devices by order
-        // of priority
-        switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
-        case AudioSystem::FORCE_BT_SCO:
-            if (!isInCall() || strategy != STRATEGY_DTMF) {
-                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
-                if (device) break;
-            }
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO;
-            if (device) break;
-            // if SCO device is requested but no SCO device is available, fall back to default case
-            // FALL THROUGH
-
-        default:    // FORCE_NONE
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
-            if (device) break;
-#ifdef WITH_A2DP
-            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
-            if (!isInCall() && !mA2dpSuspended) {
-                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
-                if (device) break;
-                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-                if (device) break;
-            }
-#endif
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE;
-            if (device == 0) {
-                LOGE("getDeviceForStrategy() earpiece device not found");
-            }
-            break;
-
-        case AudioSystem::FORCE_SPEAKER:
-#ifdef WITH_A2DP
-            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
-            // A2DP speaker when forcing to speaker output
-            if (!isInCall() && !mA2dpSuspended) {
-                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-                if (device) break;
-            }
-#endif
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
-            if (device == 0) {
-                LOGE("getDeviceForStrategy() speaker device not found");
-            }
-            break;
-        }
-    break;
-
-    case STRATEGY_SONIFICATION:
-
-        // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
-        // handleIncallSonification().
-        if (isInCall()) {
-            device = getDeviceForStrategy(STRATEGY_PHONE, false);
-            break;
-        }
-        // FALL THROUGH
-
-    case STRATEGY_ENFORCED_AUDIBLE:
-        // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
-        // except when in call where it doesn't default to STRATEGY_PHONE behavior
-
-        device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
-        if (device == 0) {
-            LOGE("getDeviceForStrategy() speaker device not found");
-        }
-        // The second device used for sonification is the same as the device used by media strategy
-        // FALL THROUGH
-
-    case STRATEGY_MEDIA: {
-        uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
-        if (device2 == 0) {
-            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
-        }
-#ifdef WITH_A2DP
-        if ((mA2dpOutput != 0) && !mA2dpSuspended &&
-                (strategy == STRATEGY_MEDIA || a2dpUsedForSonification())) {
-            if (device2 == 0) {
-                device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
-            }
-            if (device2 == 0) {
-                device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-            }
-            if (device2 == 0) {
-                device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-            }
-        }
-#endif
-        if (device2 == 0) {
-            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
-        }
-        if (device2 == 0) {
-            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
-        }
-        if (device2 == 0) {
-            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
-        }
-        if (device2 == 0) {
-            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
-        }
-
-        // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
-        // STRATEGY_ENFORCED_AUDIBLE, 0 otherwise
-        device |= device2;
-        if (device == 0) {
-            LOGE("getDeviceForStrategy() speaker device not found");
-        }
-        } break;
-
-    default:
-        LOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
-        break;
-    }
-
-    LOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
-    return device;
-}
-
-void AudioPolicyManagerBase::updateDeviceForStrategy()
-{
-    for (int i = 0; i < NUM_STRATEGIES; i++) {
-        mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false);
-    }
-}
-
-void AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs)
-{
-    LOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs);
-    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-
-
-    if (outputDesc->isDuplicated()) {
-        setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
-        setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
-        return;
-    }
-#ifdef WITH_A2DP
-    // filter devices according to output selected
-    if (output == mA2dpOutput) {
-        device &= AudioSystem::DEVICE_OUT_ALL_A2DP;
-    } else {
-        device &= ~AudioSystem::DEVICE_OUT_ALL_A2DP;
-    }
-#endif
-
-    uint32_t prevDevice = (uint32_t)outputDesc->device();
-    // Do not change the routing if:
-    //  - the requestede device is 0
-    //  - the requested device is the same as current device and force is not specified.
-    // Doing this check here allows the caller to call setOutputDevice() without conditions
-    if ((device == 0 || device == prevDevice) && !force) {
-        LOGV("setOutputDevice() setting same device %x or null device for output %d", device, output);
-        return;
-    }
-
-    outputDesc->mDevice = device;
-    // mute media streams if both speaker and headset are selected
-    if (output == mHardwareOutput && AudioSystem::popCount(device) == 2) {
-        setStrategyMute(STRATEGY_MEDIA, true, output);
-        // wait for the PCM output buffers to empty before proceeding with the rest of the command
-        // FIXME: increased delay due to larger buffers used for low power audio mode.
-        // remove when low power audio is controlled by policy manager.
-        usleep(outputDesc->mLatency*8*1000);
-    }
-
-    // do the routing
-    AudioParameter param = AudioParameter();
-    param.addInt(String8(AudioParameter::keyRouting), (int)device);
-    mpClientInterface->setParameters(mHardwareOutput, param.toString(), delayMs);
-    // update stream volumes according to new device
-    applyStreamVolumes(output, device, delayMs);
-
-    // if changing from a combined headset + speaker route, unmute media streams
-    if (output == mHardwareOutput && AudioSystem::popCount(prevDevice) == 2) {
-        setStrategyMute(STRATEGY_MEDIA, false, output, delayMs);
-    }
-}
-
-uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
-{
-    uint32_t device;
-
-    switch(inputSource) {
-    case AUDIO_SOURCE_DEFAULT:
-    case AUDIO_SOURCE_MIC:
-    case AUDIO_SOURCE_VOICE_RECOGNITION:
-    case AUDIO_SOURCE_VOICE_COMMUNICATION:
-        if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
-            mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
-            device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
-        } else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) {
-            device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
-        } else {
-            device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
-        }
-        break;
-    case AUDIO_SOURCE_CAMCORDER:
-        if (hasBackMicrophone()) {
-            device = AudioSystem::DEVICE_IN_BACK_MIC;
-        } else {
-            device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
-        }
-        break;
-    case AUDIO_SOURCE_VOICE_UPLINK:
-    case AUDIO_SOURCE_VOICE_DOWNLINK:
-    case AUDIO_SOURCE_VOICE_CALL:
-        device = AudioSystem::DEVICE_IN_VOICE_CALL;
-        break;
-    default:
-        LOGW("getDeviceForInputSource() invalid input source %d", inputSource);
-        device = 0;
-        break;
-    }
-    LOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
-    return device;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getActiveInput()
-{
-    for (size_t i = 0; i < mInputs.size(); i++) {
-        if (mInputs.valueAt(i)->mRefCount > 0) {
-            return mInputs.keyAt(i);
-        }
-    }
-    return 0;
-}
-
-
-AudioPolicyManagerBase::device_category AudioPolicyManagerBase::getDeviceCategory(uint32_t device)
-{
-    if (device == 0) {
-        // this happens when forcing a route update and no track is active on an output.
-        // In this case the returned category is not important.
-        return DEVICE_CATEGORY_SPEAKER;
-    }
-
-    if (AudioSystem::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.
-        if (device & AUDIO_DEVICE_OUT_SPEAKER)
-            return DEVICE_CATEGORY_SPEAKER;
-
-        device &= AUDIO_DEVICE_OUT_ALL_A2DP;
-    }
-
-    LOGW_IF(AudioSystem::popCount(device) != 1,
-            "getDeviceCategory() invalid device combination: %08x",
-            device);
-
-    switch(device) {
-        case AUDIO_DEVICE_OUT_EARPIECE:
-            return 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 DEVICE_CATEGORY_HEADSET;
-        case AUDIO_DEVICE_OUT_SPEAKER:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
-        default:
-            return DEVICE_CATEGORY_SPEAKER;
-    }
-}
-
-float AudioPolicyManagerBase::volIndexToAmpl(uint32_t device, const StreamDescriptor& streamDesc,
-        int indexInUi)
-{
-    device_category deviceCategory = 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[VOLMAX].mIndex -
-            curve[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[VOLMIN].mIndex) {         // out of bounds
-        return 0.0f;
-    } else if (volIdx < curve[VOLKNEE1].mIndex) {
-        segment = 0;
-    } else if (volIdx < curve[VOLKNEE2].mIndex) {
-        segment = 1;
-    } else if (volIdx <= curve[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 )
-
-    LOGV("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;
-}
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-            *AudioPolicyManagerBase::sVolumeProfiles[AudioPolicyManagerBase::NUM_STRATEGIES]
-                                                   [AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = {
-    { // STRATEGY_MEDIA
-        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // STRATEGY_PHONE
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // STRATEGY_SONIFICATION
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    {  // STRATEGY_DTMF
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // STRATEGY_ENFORCED_AUDIBLE
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-};
-
-void AudioPolicyManagerBase::initializeVolumeCurves()
-{
-    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
-        for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
-            mStreams[i].mVolumeCurve[j] =
-                    sVolumeProfiles[getStrategy((AudioSystem::stream_type)i)][j];
-        }
-    }
-}
-
-float AudioPolicyManagerBase::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device)
-{
-    float volume = 1.0;
-    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-    StreamDescriptor &streamDesc = mStreams[stream];
-
-    if (device == 0) {
-        device = outputDesc->device();
-    }
-
-    // if volume is not 0 (not muted), force media volume to max on digital output
-    if (stream == AudioSystem::MUSIC &&
-        index != mStreams[stream].mIndexMin &&
-        (device == AudioSystem::DEVICE_OUT_AUX_DIGITAL ||
-        device == AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET)) {
-        return 1.0;
-    }
-
-    volume = volIndexToAmpl(device, streamDesc, index);
-
-    // if a headset is connected, apply the following rules to ring tones and notifications
-    // to avoid sound level bursts in user's ears:
-    // - always attenuate ring tones and notifications volume by 6dB
-    // - if music is playing, always limit the volume to current music volume,
-    // with a minimum threshold at -36dB so that notification is always perceived.
-    if ((device &
-        (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
-        AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
-        AudioSystem::DEVICE_OUT_WIRED_HEADSET |
-        AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) &&
-        ((getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) ||
-         (stream == AudioSystem::SYSTEM)) &&
-        streamDesc.mCanBeMuted) {
-        volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
-        // when the phone is ringing we must consider that music could have been paused just before
-        // by the music application and behave as if music was active if the last music track was
-        // just stopped
-        if (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) {
-            float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device);
-            float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
-            if (volume > minVol) {
-                volume = minVol;
-                LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
-            }
-        }
-    }
-
-    return volume;
-}
-
-status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force)
-{
-
-    // do not change actual stream volume if the stream is muted
-    if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
-        LOGV("checkAndSetVolume() stream %d muted count %d", stream, mOutputs.valueFor(output)->mMuteCount[stream]);
-        return NO_ERROR;
-    }
-
-    // do not change in call volume if bluetooth is connected and vice versa
-    if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
-        (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
-        LOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
-             stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
-        return INVALID_OPERATION;
-    }
-
-    float volume = computeVolume(stream, index, output, device);
-    // We actually change the volume if:
-    // - the float value returned by computeVolume() changed
-    // - the force flag is set
-    if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
-            force) {
-        mOutputs.valueFor(output)->mCurVolume[stream] = volume;
-        LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
-        if (stream == AudioSystem::VOICE_CALL ||
-            stream == AudioSystem::DTMF ||
-            stream == AudioSystem::BLUETOOTH_SCO) {
-            // offset value to reflect actual hardware volume that never reaches 0
-            // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
-            volume = 0.01 + 0.99 * volume;
-            // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
-            // enabled
-            if (stream == AudioSystem::BLUETOOTH_SCO) {
-                mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
-            }
-        }
-
-        mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
-    }
-
-    if (stream == AudioSystem::VOICE_CALL ||
-        stream == AudioSystem::BLUETOOTH_SCO) {
-        float voiceVolume;
-        // Force voice volume to max for bluetooth SCO as volume is managed by the headset
-        if (stream == AudioSystem::VOICE_CALL) {
-            voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
-        } else {
-            voiceVolume = 1.0;
-        }
-
-        if (voiceVolume != mLastVoiceVolume && output == mHardwareOutput) {
-            mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
-            mLastVoiceVolume = voiceVolume;
-        }
-    }
-
-    return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs, bool force)
-{
-    LOGV("applyStreamVolumes() for output %d and device %x", output, device);
-
-    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
-        checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, device, delayMs, force);
-    }
-}
-
-void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs)
-{
-    LOGV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
-    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
-        if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
-            setStreamMute(stream, on, output, delayMs);
-        }
-    }
-}
-
-void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs)
-{
-    StreamDescriptor &streamDesc = mStreams[stream];
-    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-
-    LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]);
-
-    if (on) {
-        if (outputDesc->mMuteCount[stream] == 0) {
-            if (streamDesc.mCanBeMuted) {
-                checkAndSetVolume(stream, 0, output, outputDesc->device(), delayMs);
-            }
-        }
-        // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
-        outputDesc->mMuteCount[stream]++;
-    } else {
-        if (outputDesc->mMuteCount[stream] == 0) {
-            LOGW("setStreamMute() unmuting non muted stream!");
-            return;
-        }
-        if (--outputDesc->mMuteCount[stream] == 0) {
-            checkAndSetVolume(stream, streamDesc.mIndexCur, output, outputDesc->device(), delayMs);
-        }
-    }
-}
-
-void AudioPolicyManagerBase::handleIncallSonification(int stream, bool starting, bool stateChange)
-{
-    // if the stream pertains to sonification strategy and we are in call we must
-    // mute the stream if it is low visibility. If it is high visibility, we must play a tone
-    // in the device used for phone strategy and play the tone if the selected device does not
-    // interfere with the device used for phone strategy
-    // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
-    // many times as there are active tracks on the output
-
-    if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
-        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
-        LOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
-                stream, starting, outputDesc->mDevice, stateChange);
-        if (outputDesc->mRefCount[stream]) {
-            int muteCount = 1;
-            if (stateChange) {
-                muteCount = outputDesc->mRefCount[stream];
-            }
-            if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
-                LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
-                for (int i = 0; i < muteCount; i++) {
-                    setStreamMute(stream, starting, mHardwareOutput);
-                }
-            } else {
-                LOGV("handleIncallSonification() high visibility");
-                if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE)) {
-                    LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
-                    for (int i = 0; i < muteCount; i++) {
-                        setStreamMute(stream, starting, mHardwareOutput);
-                    }
-                }
-                if (starting) {
-                    mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
-                } else {
-                    mpClientInterface->stopTone();
-                }
-            }
-        }
-    }
-}
-
-bool AudioPolicyManagerBase::isInCall()
-{
-    return isStateInCall(mPhoneState);
-}
-
-bool AudioPolicyManagerBase::isStateInCall(int state) {
-    return ((state == AudioSystem::MODE_IN_CALL) ||
-            (state == AudioSystem::MODE_IN_COMMUNICATION));
-}
-
-bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
-                                    uint32_t samplingRate,
-                                    uint32_t format,
-                                    uint32_t channels,
-                                    AudioSystem::output_flags flags,
-                                    uint32_t device)
-{
-   return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
-          (format !=0 && !AudioSystem::isLinearPCM(format)));
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
-{
-    return MAX_EFFECTS_CPU_LOAD;
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsMemory()
-{
-    return MAX_EFFECTS_MEMORY;
-}
-
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor()
-    : mId(0), mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
-    mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0)
-{
-    // clear usage count for all stream types
-    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
-        mRefCount[i] = 0;
-        mCurVolume[i] = -1.0;
-        mMuteCount[i] = 0;
-        mStopTime[i] = 0;
-    }
-}
-
-uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::device()
-{
-    uint32_t device = 0;
-    if (isDuplicated()) {
-        device = mOutput1->mDevice | mOutput2->mDevice;
-    } else {
-        device = mDevice;
-    }
-    return device;
-}
-
-void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type 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) {
-        LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
-        mRefCount[stream] = 0;
-        return;
-    }
-    mRefCount[stream] += delta;
-    LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
-{
-    uint32_t refcount = 0;
-    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
-        refcount += mRefCount[i];
-    }
-    return refcount;
-}
-
-uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::strategyRefCount(routing_strategy strategy)
-{
-    uint32_t refCount = 0;
-    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
-        if (getStrategy((AudioSystem::stream_type)i) == strategy) {
-            refCount += mRefCount[i];
-        }
-    }
-    return refCount;
-}
-
-status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    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", mChannels);
-    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 < AudioSystem::NUM_STREAM_TYPES; 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;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor()
-    : mSamplingRate(0), mFormat(0), mChannels(0),
-      mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0),
-      mInputSource(0)
-{
-}
-
-status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    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", mChannels);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-    return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-void AudioPolicyManagerBase::StreamDescriptor::dump(char* buffer, size_t size)
-{
-    snprintf(buffer, size, "      %02d         %02d         %02d         %d\n",
-            mIndexMin,
-            mIndexMax,
-            mIndexCur,
-            mCanBeMuted);
-}
-
-// --- EffectDescriptor class implementation
-
-status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, " I/O: %d\n", mIo);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Session: %d\n", mSession);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Name: %s\n",  mDesc.name);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " %s\n",  mEnabled ? "Enabled" : "Disabled");
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-    return NO_ERROR;
-}
-
-
-
-}; // namespace android
diff --git a/audio/AudioPolicyManagerBase.h b/audio/AudioPolicyManagerBase.h
deleted file mode 100755
index 3ab748d..0000000
--- a/audio/AudioPolicyManagerBase.h
+++ /dev/null
@@ -1,393 +0,0 @@
-/*
- * 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 <utils/Timers.h>
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <hardware_legacy/AudioPolicyInterface.h>
-
-
-namespace android_audio_legacy {
-    using android::KeyedVector;
-
-// ----------------------------------------------------------------------------
-
-#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 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
-
-// ----------------------------------------------------------------------------
-// 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(AudioSystem::audio_devices device,
-                                                          AudioSystem::device_connection_state state,
-                                                          const char *device_address);
-        virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
-                                                                              const char *device_address);
-        virtual void setPhoneState(int state);
-        virtual void setRingerMode(uint32_t mode, uint32_t mask);
-        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 = 0,
-                                            uint32_t format = AudioSystem::FORMAT_DEFAULT,
-                                            uint32_t channels = 0,
-                                            AudioSystem::output_flags flags =
-                                                    AudioSystem::OUTPUT_FLAG_INDIRECT);
-        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,
-                                            uint32_t format,
-                                            uint32_t channels,
-                                            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);
-        virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index);
-
-        // 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 uint32_t getDevicesForStream(AudioSystem::stream_type stream);
-
-        virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
-        virtual status_t registerEffect(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;
-
-        virtual status_t dump(int fd);
-
-protected:
-
-        enum routing_strategy {
-            STRATEGY_MEDIA,
-            STRATEGY_PHONE,
-            STRATEGY_SONIFICATION,
-            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
-        };
-
-        // 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];
-        // default volume curves per strategy and device category. See initializeVolumeCurves()
-        static const VolumeCurvePoint *sVolumeProfiles[NUM_STRATEGIES][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();
-
-            status_t    dump(int fd);
-
-            uint32_t device();
-            void changeRefCount(AudioSystem::stream_type, int delta);
-            uint32_t refCount();
-            uint32_t strategyRefCount(routing_strategy strategy);
-            bool isUsedByStrategy(routing_strategy strategy) { return (strategyRefCount(strategy) != 0);}
-            bool isDuplicated() { return (mOutput1 != NULL && mOutput2 != NULL); }
-
-            audio_io_handle_t mId;              // output handle
-            uint32_t mSamplingRate;             //
-            uint32_t mFormat;                   //
-            uint32_t mChannels;                 // output configuration
-            uint32_t mLatency;                  //
-            AudioSystem::output_flags mFlags;   //
-            uint32_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
-        };
-
-        // 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();
-
-            status_t    dump(int fd);
-
-            uint32_t mSamplingRate;                     //
-            uint32_t mFormat;                           // input configuration
-            uint32_t mChannels;                         //
-            AudioSystem::audio_in_acoustics mAcoustics; //
-            uint32_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)
-        };
-
-        // stream descriptor used for volume control
-        class StreamDescriptor
-        {
-        public:
-            StreamDescriptor()
-            :   mIndexMin(0), mIndexMax(1), mIndexCur(1), mCanBeMuted(true) {}
-
-            void dump(char* buffer, size_t size);
-
-            int mIndexMin;      // min volume index
-            int mIndexMax;      // max volume index
-            int mIndexCur;      // current volume index
-            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 determined
-        // 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 updateDeviceForStrategy() is called.
-        virtual uint32_t getDeviceForStrategy(routing_strategy strategy, bool fromCache = true);
-        // change the route of the specified output
-        void setOutputDevice(audio_io_handle_t output, uint32_t device, bool force = false, int delayMs = 0);
-        // select input device corresponding to requested audio source
-        virtual uint32_t getDeviceForInputSource(int inputSource);
-        // return io handle of active input or 0 if no input is active
-        audio_io_handle_t getActiveInput();
-        // 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, uint32_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, uint32_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, uint32_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);
-        // Mute or unmute the stream on the specified output
-        void setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs = 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 is current platform implements a back microphone
-        virtual bool hasBackMicrophone() const { return false; }
-        // 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);
-
-#ifdef WITH_A2DP
-        // true is current platform supports suplication of notifications and ringtones over A2DP output
-        virtual bool a2dpUsedForSonification() const { return true; }
-        status_t handleA2dpConnection(AudioSystem::audio_devices device,
-                                                            const char *device_address);
-        status_t handleA2dpDisconnection(AudioSystem::audio_devices device,
-                                                            const char *device_address);
-        void closeA2dpOutputs();
-        // checks and if necessary changes output (a2dp, duplicated or hardware) used for all strategies.
-        // must be called every time a condition that affects the output choice for a given strategy is
-        // changed: connected device, phone state, force use...
-        // Must be called before updateDeviceForStrategy()
-        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();
-#endif
-        // 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
-        uint32_t getNewDevice(audio_io_handle_t output, bool fromCache = true);
-        // 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 updateDeviceForStrategy();
-        // true if current platform requires a specific output to be opened for this particular
-        // set of parameters. This function is called by getOutput() and is implemented by platform
-        // specific audio policy manager.
-        virtual bool needsDirectOuput(AudioSystem::stream_type stream,
-                                    uint32_t samplingRate,
-                                    uint32_t format,
-                                    uint32_t channels,
-                                    AudioSystem::output_flags flags,
-                                    uint32_t device);
-        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(uint32_t device);
-
-        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
-        audio_io_handle_t mHardwareOutput;              // hardware output handler
-        audio_io_handle_t mA2dpOutput;                  // A2DP output handler
-        audio_io_handle_t mDuplicatedOutput;            // duplicated output handler: outputs to hardware and A2DP.
-
-        KeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;   // list of output descriptors
-        KeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs;     // list of input descriptors
-        uint32_t mAvailableOutputDevices;                                   // bit field of all available output devices
-        uint32_t mAvailableInputDevices;                                    // bit field of all available input devices
-        int mPhoneState;                                                    // current phone state
-        uint32_t                 mRingerMode;                               // current ringer mode
-        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
-        bool    mLimitRingtoneVolume;                                       // limit ringtone volume to music volume if headset connected
-        uint32_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
-
-#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(uint32_t device, const StreamDescriptor& streamDesc,
-                int indexInUi);
-};
-
-};
diff --git a/audio/AudioPolicyManagerDefault.cpp b/audio/AudioPolicyManagerDefault.cpp
deleted file mode 100755
index 9083638..0000000
--- a/audio/AudioPolicyManagerDefault.cpp
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * 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.
- */
-
-#define LOG_TAG "AudioPolicyManagerDefault"
-//#define LOG_NDEBUG 0
-
-#include "AudioPolicyManagerDefault.h"
-
-namespace android_audio_legacy {
-
-extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
-{
-    return new AudioPolicyManagerDefault(clientInterface);
-}
-
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
-    delete interface;
-}
-
-}; // namespace android
diff --git a/audio/AudioPolicyManagerDefault.h b/audio/AudioPolicyManagerDefault.h
deleted file mode 100755
index b2b2576..0000000
--- a/audio/AudioPolicyManagerDefault.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * 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 <stdbool.h>
-
-#include <hardware_legacy/AudioPolicyManagerBase.h>
-
-namespace android_audio_legacy {
-
-class AudioPolicyManagerDefault: public AudioPolicyManagerBase
-{
-
-public:
-                AudioPolicyManagerDefault(AudioPolicyClientInterface *clientInterface)
-                : AudioPolicyManagerBase(clientInterface) {}
-
-        virtual ~AudioPolicyManagerDefault() {}
-
-protected:
-        // true is current platform implements a back microphone
-        virtual bool hasBackMicrophone() const { return false; }
-#ifdef WITH_A2DP
-        // true is current platform supports suplication of notifications and ringtones over A2DP output
-        virtual bool a2dpUsedForSonification() const { return true; }
-#endif
-
-};
-};
diff --git a/audio/AudioSystem.h b/audio/AudioSystem.h
deleted file mode 100755
index bd75a73..0000000
--- a/audio/AudioSystem.h
+++ /dev/null
@@ -1,562 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_AUDIOSYSTEM_H_
-#define ANDROID_AUDIOSYSTEM_H_
-#define ANDROID_AUDIOPARAMETER_H_
-
-#include <utils/RefBase.h>
-#include <utils/threads.h>
-#include <media/IAudioFlinger.h>
-
-namespace android {
-
-typedef void (*audio_error_callback)(status_t err);
-typedef int audio_io_handle_t;
-
-class IAudioPolicyService;
-class String8;
-
-class AudioSystem
-{
-public:
-
-    enum stream_type {
-        DEFAULT          =-1,
-        VOICE_CALL       = 0,
-        SYSTEM           = 1,
-        RING             = 2,
-        MUSIC            = 3,
-        ALARM            = 4,
-        NOTIFICATION     = 5,
-        BLUETOOTH_SCO    = 6,
-        ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker
-        DTMF             = 8,
-        TTS              = 9,
-#ifdef HAVE_FM_RADIO
-        FM              = 10,
-#endif
-        NUM_STREAM_TYPES
-    };
-
-    // Audio sub formats (see AudioSystem::audio_format).
-    enum pcm_sub_format {
-        PCM_SUB_16_BIT          = 0x1, // must be 1 for backward compatibility
-        PCM_SUB_8_BIT           = 0x2, // must be 2 for backward compatibility
-    };
-
-    // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify
-    // bit rate, stereo mode, version...
-    enum mp3_sub_format {
-        //TODO
-    };
-
-    // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned,
-    // encoding mode for recording...
-    enum amr_sub_format {
-        //TODO
-    };
-
-    // AAC sub format field definition: specify profile or bitrate for recording...
-    enum aac_sub_format {
-        //TODO
-    };
-
-    // VORBIS sub format field definition: specify quality for recording...
-    enum vorbis_sub_format {
-        //TODO
-    };
-
-    // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits).
-    // The main format indicates the main codec type. The sub format field indicates options and parameters
-    // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate
-    // or profile. It can also be used for certain formats to give informations not present in the encoded
-    // audio stream (e.g. octet alignement for AMR).
-    enum audio_format {
-        INVALID_FORMAT      = -1,
-        FORMAT_DEFAULT      = 0,
-        PCM                 = 0x00000000, // must be 0 for backward compatibility
-        MP3                 = 0x01000000,
-        AMR_NB              = 0x02000000,
-        AMR_WB              = 0x03000000,
-        AAC                 = 0x04000000,
-        HE_AAC_V1           = 0x05000000,
-        HE_AAC_V2           = 0x06000000,
-        VORBIS              = 0x07000000,
-        MAIN_FORMAT_MASK    = 0xFF000000,
-        SUB_FORMAT_MASK     = 0x00FFFFFF,
-        // Aliases
-        PCM_16_BIT          = (PCM|PCM_SUB_16_BIT),
-        PCM_8_BIT          = (PCM|PCM_SUB_8_BIT)
-    };
-
-
-    // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java
-    enum audio_channels {
-        // output channels
-        CHANNEL_OUT_FRONT_LEFT = 0x4,
-        CHANNEL_OUT_FRONT_RIGHT = 0x8,
-        CHANNEL_OUT_FRONT_CENTER = 0x10,
-        CHANNEL_OUT_LOW_FREQUENCY = 0x20,
-        CHANNEL_OUT_BACK_LEFT = 0x40,
-        CHANNEL_OUT_BACK_RIGHT = 0x80,
-        CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100,
-        CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,
-        CHANNEL_OUT_BACK_CENTER = 0x400,
-        CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT,
-        CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT),
-        CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
-                CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
-        CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
-                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER),
-        CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
-                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
-        CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
-                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
-                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER),
-        CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
-                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
-                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER),
-
-        // input channels
-        CHANNEL_IN_LEFT = 0x4,
-        CHANNEL_IN_RIGHT = 0x8,
-        CHANNEL_IN_FRONT = 0x10,
-        CHANNEL_IN_BACK = 0x20,
-        CHANNEL_IN_LEFT_PROCESSED = 0x40,
-        CHANNEL_IN_RIGHT_PROCESSED = 0x80,
-        CHANNEL_IN_FRONT_PROCESSED = 0x100,
-        CHANNEL_IN_BACK_PROCESSED = 0x200,
-        CHANNEL_IN_PRESSURE = 0x400,
-        CHANNEL_IN_X_AXIS = 0x800,
-        CHANNEL_IN_Y_AXIS = 0x1000,
-        CHANNEL_IN_Z_AXIS = 0x2000,
-        CHANNEL_IN_VOICE_UPLINK = 0x4000,
-        CHANNEL_IN_VOICE_DNLINK = 0x8000,
-#ifdef OMAP_ENHANCEMENT
-        CHANNEL_IN_VOICE_UPLINK_DNLINK = 0x10000,
-#endif
-        CHANNEL_IN_MONO = CHANNEL_IN_FRONT,
-        CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT),
-        CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK|
-                CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED|
-                CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS |
-#ifdef OMAP_ENHANCEMENT
-                CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK | CHANNEL_IN_VOICE_UPLINK_DNLINK)
-#else
-                CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK )
-#endif
-    };
-
-    enum audio_mode {
-        MODE_INVALID = -2,
-        MODE_CURRENT = -1,
-        MODE_NORMAL = 0,
-        MODE_RINGTONE,
-        MODE_IN_CALL,
-        MODE_IN_COMMUNICATION,
-        NUM_MODES  // not a valid entry, denotes end-of-list
-    };
-
-    enum audio_in_acoustics {
-        AGC_ENABLE    = 0x0001,
-        AGC_DISABLE   = 0,
-        NS_ENABLE     = 0x0002,
-        NS_DISABLE    = 0,
-        TX_IIR_ENABLE = 0x0004,
-        TX_DISABLE    = 0
-    };
-
-    // special audio session values
-    enum audio_sessions {
-        SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream
-                                   // (value must be less than 0)
-        SESSION_OUTPUT_MIX = 0,    // session for effects applied to output mix. These effects can
-                                   // be moved by audio policy manager to another output stream
-                                   // (value must be 0)
-    };
-
-    /* These are static methods to control the system-wide AudioFlinger
-     * only privileged processes can have access to them
-     */
-
-    // mute/unmute microphone
-    static status_t muteMicrophone(bool state);
-    static status_t isMicrophoneMuted(bool *state);
-
-    // set/get master volume
-    static status_t setMasterVolume(float value);
-    static status_t getMasterVolume(float* volume);
-    // mute/unmute audio outputs
-    static status_t setMasterMute(bool mute);
-    static status_t getMasterMute(bool* mute);
-
-    // set/get stream volume on specified output
-    static status_t setStreamVolume(int stream, float value, int output);
-    static status_t getStreamVolume(int stream, float* volume, int output);
-
-    // mute/unmute stream
-    static status_t setStreamMute(int stream, bool mute);
-    static status_t getStreamMute(int stream, bool* mute);
-
-    // set audio mode in audio hardware (see AudioSystem::audio_mode)
-    static status_t setMode(int mode);
-
-    // returns true in *state if tracks are active on the specified stream
-    static status_t isStreamActive(int stream, bool *state);
-
-    // set/get audio hardware parameters. The function accepts a list of parameters
-    // key value pairs in the form: key1=value1;key2=value2;...
-    // Some keys are reserved for standard parameters (See AudioParameter class).
-    static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
-    static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
-
-    static void setErrorCallback(audio_error_callback cb);
-
-    // helper function to obtain AudioFlinger service handle
-    static const sp<IAudioFlinger>& get_audio_flinger();
-
-    static float linearToLog(int volume);
-    static int logToLinear(float volume);
-
-    static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT);
-    static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT);
-    static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT);
-
-    static bool routedToA2dpOutput(int streamType);
-
-    static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
-        size_t* buffSize);
-
-    static status_t setVoiceVolume(float volume);
-#ifdef HAVE_FM_RADIO
-    static status_t setFmVolume(float volume);
-#endif
-
-    // return the number of audio frames written by AudioFlinger to audio HAL and
-    // audio dsp to DAC since the output on which the specificed stream is playing
-    // has exited standby.
-    // returned status (from utils/Errors.h) can be:
-    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
-    // - INVALID_OPERATION: Not supported on current hardware platform
-    // - BAD_VALUE: invalid parameter
-    // NOTE: this feature is not supported on all hardware platforms and it is
-    // necessary to check returned status before using the returned values.
-    static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT);
-
-    static unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);
-
-    static int newAudioSessionId();
-    //
-    // AudioPolicyService interface
-    //
-
-    enum audio_devices {
-        // output devices
-        DEVICE_OUT_EARPIECE = 0x1,
-        DEVICE_OUT_SPEAKER = 0x2,
-        DEVICE_OUT_WIRED_HEADSET = 0x4,
-        DEVICE_OUT_WIRED_HEADPHONE = 0x8,
-        DEVICE_OUT_BLUETOOTH_SCO = 0x10,
-        DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,
-        DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,
-        DEVICE_OUT_BLUETOOTH_A2DP = 0x80,
-        DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
-        DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
-        DEVICE_OUT_AUX_DIGITAL = 0x400,
-#ifdef HAVE_FM_RADIO
-        DEVICE_OUT_FM = 0x800,
-        DEVICE_OUT_FM_SPEAKER = 0x1000,
-        DEVICE_OUT_FM_ALL = (DEVICE_OUT_FM | DEVICE_OUT_FM_SPEAKER),
-#elif defined(OMAP_ENHANCEMENT)
-        DEVICE_OUT_FM_TRANSMIT = 0x800,
-        DEVICE_OUT_LOW_POWER = 0x1000,
-#endif
-        DEVICE_OUT_HDMI = 0x2000,
-        DEVICE_OUT_DEFAULT = 0x8000,
-        DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |
-#ifdef HAVE_FM_RADIO
-                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_FM | DEVICE_OUT_FM_SPEAKER | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
-#else
-                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
-#endif
-                DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
-#if defined(OMAP_ENHANCEMENT) && !defined(HAVE_FM_RADIO)
-                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_LOW_POWER |
-                DEVICE_OUT_FM_TRANSMIT | DEVICE_OUT_DEFAULT),
-#else
-                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_HDMI | DEVICE_OUT_DEFAULT),
-#endif
-        DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
-                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
-
-        // input devices
-        DEVICE_IN_COMMUNICATION = 0x10000,
-        DEVICE_IN_AMBIENT = 0x20000,
-        DEVICE_IN_BUILTIN_MIC = 0x40000,
-        DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,
-        DEVICE_IN_WIRED_HEADSET = 0x100000,
-        DEVICE_IN_AUX_DIGITAL = 0x200000,
-        DEVICE_IN_VOICE_CALL = 0x400000,
-        DEVICE_IN_BACK_MIC = 0x800000,
-#ifdef HAVE_FM_RADIO
-        DEVICE_IN_FM_RX = 0x1000000,
-        DEVICE_IN_FM_RX_A2DP = 0x2000000,
-#endif
-#ifdef OMAP_ENHANCEMENT
-        DEVICE_IN_FM_ANALOG = 0x1000000,
-#endif
-        DEVICE_IN_DEFAULT = 0x80000000,
-
-        DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |
-                DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |
-#ifdef HAVE_FM_RADIO
-                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_FM_RX | DEVICE_IN_FM_RX_A2DP | DEVICE_IN_DEFAULT)
-#elif OMAP_ENHANCEMENT
-                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC  | DEVICE_IN_FM_ANALOG | DEVICE_IN_DEFAULT)
-#else
-                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT)
-#endif
-
-    };
-
-    // device connection states used for setDeviceConnectionState()
-    enum device_connection_state {
-        DEVICE_STATE_UNAVAILABLE,
-        DEVICE_STATE_AVAILABLE,
-        NUM_DEVICE_STATES
-    };
-
-    // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)
-    enum output_flags {
-        OUTPUT_FLAG_INDIRECT = 0x0,
-        OUTPUT_FLAG_DIRECT = 0x1
-    };
-
-    // device categories used for setForceUse()
-    enum forced_config {
-        FORCE_NONE,
-        FORCE_SPEAKER,
-        FORCE_HEADPHONES,
-        FORCE_BT_SCO,
-        FORCE_BT_A2DP,
-        FORCE_WIRED_ACCESSORY,
-        FORCE_BT_CAR_DOCK,
-        FORCE_BT_DESK_DOCK,
-        NUM_FORCE_CONFIG,
-        FORCE_DEFAULT = FORCE_NONE
-    };
-
-    // usages used for setForceUse()
-    enum force_use {
-        FOR_COMMUNICATION,
-        FOR_MEDIA,
-        FOR_RECORD,
-        FOR_DOCK,
-        NUM_FORCE_USE
-    };
-
-    // types of io configuration change events received with ioConfigChanged()
-    enum io_config_event {
-        OUTPUT_OPENED,
-        OUTPUT_CLOSED,
-        OUTPUT_CONFIG_CHANGED,
-        INPUT_OPENED,
-        INPUT_CLOSED,
-        INPUT_CONFIG_CHANGED,
-        STREAM_CONFIG_CHANGED,
-        NUM_CONFIG_EVENTS
-    };
-
-    // audio output descritor used to cache output configurations in client process to avoid frequent calls
-    // through IAudioFlinger
-    class OutputDescriptor {
-    public:
-        OutputDescriptor()
-        : samplingRate(0), format(0), channels(0), frameCount(0), latency(0)  {}
-
-        uint32_t samplingRate;
-        int32_t format;
-        int32_t channels;
-        size_t frameCount;
-        uint32_t latency;
-    };
-
-    //
-    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
-    //
-    static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address);
-    static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address);
-    static status_t setPhoneState(int state);
-    static status_t setRingerMode(uint32_t mode, uint32_t mask);
-    static status_t setForceUse(force_use usage, forced_config config);
-    static forced_config getForceUse(force_use usage);
-    static audio_io_handle_t getOutput(stream_type stream,
-                                        uint32_t samplingRate = 0,
-                                        uint32_t format = FORMAT_DEFAULT,
-                                        uint32_t channels = CHANNEL_OUT_STEREO,
-                                        output_flags flags = OUTPUT_FLAG_INDIRECT);
-    static status_t startOutput(audio_io_handle_t output,
-                                AudioSystem::stream_type stream,
-                                int session = 0);
-    static status_t stopOutput(audio_io_handle_t output,
-                               AudioSystem::stream_type stream,
-                               int session = 0);
-    static void releaseOutput(audio_io_handle_t output);
-    static audio_io_handle_t getInput(int inputSource,
-                                    uint32_t samplingRate = 0,
-                                    uint32_t format = FORMAT_DEFAULT,
-                                    uint32_t channels = CHANNEL_IN_MONO,
-                                    audio_in_acoustics acoustics = (audio_in_acoustics)0);
-    static status_t startInput(audio_io_handle_t input);
-    static status_t stopInput(audio_io_handle_t input);
-    static void releaseInput(audio_io_handle_t input);
-    static status_t initStreamVolume(stream_type stream,
-                                      int indexMin,
-                                      int indexMax);
-    static status_t setStreamVolumeIndex(stream_type stream, int index);
-    static status_t getStreamVolumeIndex(stream_type stream, int *index);
-
-    static uint32_t getStrategyForStream(stream_type stream);
-
-    static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
-    static status_t registerEffect(effect_descriptor_t *desc,
-                                    audio_io_handle_t output,
-                                    uint32_t strategy,
-                                    int session,
-                                    int id);
-    static status_t unregisterEffect(int id);
-
-    static const sp<IAudioPolicyService>& get_audio_policy_service();
-
-    // ----------------------------------------------------------------------------
-
-    static uint32_t popCount(uint32_t u);
-    static bool isOutputDevice(audio_devices device);
-    static bool isInputDevice(audio_devices device);
-    static bool isA2dpDevice(audio_devices device);
-#ifdef HAVE_FM_RADIO
-    static bool isFmDevice(audio_devices device);
-#endif
-    static bool isBluetoothScoDevice(audio_devices device);
-    static bool isLowVisibility(stream_type stream);
-    static bool isOutputChannel(uint32_t channel);
-    static bool isInputChannel(uint32_t channel);
-    static bool isValidFormat(uint32_t format);
-    static bool isLinearPCM(uint32_t format);
-
-private:
-
-    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
-    {
-    public:
-        AudioFlingerClient() {
-        }
-
-        // DeathRecipient
-        virtual void binderDied(const wp<IBinder>& who);
-
-        // IAudioFlingerClient
-
-        // indicate a change in the configuration of an output or input: keeps the cached
-        // values for output/input parameters upto date in client process
-        virtual void ioConfigChanged(int event, int ioHandle, void *param2);
-    };
-
-    class AudioPolicyServiceClient: public IBinder::DeathRecipient
-    {
-    public:
-        AudioPolicyServiceClient() {
-        }
-
-        // DeathRecipient
-        virtual void binderDied(const wp<IBinder>& who);
-    };
-
-    static sp<AudioFlingerClient> gAudioFlingerClient;
-    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
-    friend class AudioFlingerClient;
-    friend class AudioPolicyServiceClient;
-
-    static Mutex gLock;
-    static sp<IAudioFlinger> gAudioFlinger;
-    static audio_error_callback gAudioErrorCallback;
-
-    static size_t gInBuffSize;
-    // previous parameters for recording buffer size queries
-    static uint32_t gPrevInSamplingRate;
-    static int gPrevInFormat;
-    static int gPrevInChannelCount;
-
-    static sp<IAudioPolicyService> gAudioPolicyService;
-
-    // mapping between stream types and outputs
-    static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;
-    // list of output descritor containing cached parameters (sampling rate, framecount, channel count...)
-    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
-};
-
-class AudioParameter {
-
-public:
-    AudioParameter() {}
-    AudioParameter(const String8& keyValuePairs);
-    virtual ~AudioParameter();
-
-    // reserved parameter keys for changing standard parameters with setParameters() function.
-    // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input
-    // configuration changes and act accordingly.
-    //  keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices
-    //  keySamplingRate: to change sampling rate routing, value is an int
-    //  keyFormat: to change audio format, value is an int in AudioSystem::audio_format
-    //  keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels
-    //  keyFrameCount: to change audio output frame count, value is an int
-    //  keyInputSource: to change audio input source, value is an int in audio_source
-    //     (defined in media/mediarecorder.h)
-    static const char *keyRouting;
-    static const char *keySamplingRate;
-    static const char *keyFormat;
-    static const char *keyChannels;
-    static const char *keyFrameCount;
-#ifdef HAVE_FM_RADIO
-    static const char *keyFmOn;
-    static const char *keyFmOff;
-#endif
-    static const char *keyInputSource;
-
-    String8 toString();
-
-    status_t add(const String8& key, const String8& value);
-    status_t addInt(const String8& key, const int value);
-    status_t addFloat(const String8& key, const float value);
-
-    status_t remove(const String8& key);
-
-    status_t get(const String8& key, String8& value);
-    status_t getInt(const String8& key, int& value);
-    status_t getFloat(const String8& key, float& value);
-    status_t getAt(size_t index, String8& key, String8& value);
-
-    size_t size() { return mParameters.size(); }
-
-private:
-    String8 mKeyValuePairs;
-    KeyedVector <String8, String8> mParameters;
-};
-
-};  // namespace android
-
-#endif  /*ANDROID_AUDIOSYSTEM_H_*/
diff --git a/audio/MODULE_LICENSE_APACHE2 b/audio/MODULE_LICENSE_APACHE2
deleted file mode 100755
index e69de29..0000000
--- a/audio/MODULE_LICENSE_APACHE2
+++ /dev/null
diff --git a/audio/NOTICE b/audio/NOTICE
deleted file mode 100755
index 3237da6..0000000
--- a/audio/NOTICE
+++ /dev/null
@@ -1,190 +0,0 @@
-
-   Copyright (c) 2008-2009, 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.
-
-   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.
-
-
-                                 Apache License
-                           Version 2.0, January 2004
-                        http://www.apache.org/licenses/
-
-   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
-
-   1. Definitions.
-
-      "License" shall mean the terms and conditions for use, reproduction,
-      and distribution as defined by Sections 1 through 9 of this document.
-
-      "Licensor" shall mean the copyright owner or entity authorized by
-      the copyright owner that is granting the License.
-
-      "Legal Entity" shall mean the union of the acting entity and all
-      other entities that control, are controlled by, or are under common
-      control with that entity. For the purposes of this definition,
-      "control" means (i) the power, direct or indirect, to cause the
-      direction or management of such entity, whether by contract or
-      otherwise, or (ii) ownership of fifty percent (50%) or more of the
-      outstanding shares, or (iii) beneficial ownership of such entity.
-
-      "You" (or "Your") shall mean an individual or Legal Entity
-      exercising permissions granted by this License.
-
-      "Source" form shall mean the preferred form for making modifications,
-      including but not limited to software source code, documentation
-      source, and configuration files.
-
-      "Object" form shall mean any form resulting from mechanical
-      transformation or translation of a Source form, including but
-      not limited to compiled object code, generated documentation,
-      and conversions to other media types.
-
-      "Work" shall mean the work of authorship, whether in Source or
-      Object form, made available under the License, as indicated by a
-      copyright notice that is included in or attached to the work
-      (an example is provided in the Appendix below).
-
-      "Derivative Works" shall mean any work, whether in Source or Object
-      form, that is based on (or derived from) the Work and for which the
-      editorial revisions, annotations, elaborations, or other modifications
-      represent, as a whole, an original work of authorship. For the purposes
-      of this License, Derivative Works shall not include works that remain
-      separable from, or merely link (or bind by name) to the interfaces of,
-      the Work and Derivative Works thereof.
-
-      "Contribution" shall mean any work of authorship, including
-      the original version of the Work and any modifications or additions
-      to that Work or Derivative Works thereof, that is intentionally
-      submitted to Licensor for inclusion in the Work by the copyright owner
-      or by an individual or Legal Entity authorized to submit on behalf of
-      the copyright owner. For the purposes of this definition, "submitted"
-      means any form of electronic, verbal, or written communication sent
-      to the Licensor or its representatives, including but not limited to
-      communication on electronic mailing lists, source code control systems,
-      and issue tracking systems that are managed by, or on behalf of, the
-      Licensor for the purpose of discussing and improving the Work, but
-      excluding communication that is conspicuously marked or otherwise
-      designated in writing by the copyright owner as "Not a Contribution."
-
-      "Contributor" shall mean Licensor and any individual or Legal Entity
-      on behalf of whom a Contribution has been received by Licensor and
-      subsequently incorporated within the Work.
-
-   2. Grant of Copyright License. Subject to the terms and conditions of
-      this License, each Contributor hereby grants to You a perpetual,
-      worldwide, non-exclusive, no-charge, royalty-free, irrevocable
-      copyright license to reproduce, prepare Derivative Works of,
-      publicly display, publicly perform, sublicense, and distribute the
-      Work and such Derivative Works in Source or Object form.
-
-   3. Grant of Patent License. Subject to the terms and conditions of
-      this License, each Contributor hereby grants to You a perpetual,
-      worldwide, non-exclusive, no-charge, royalty-free, irrevocable
-      (except as stated in this section) patent license to make, have made,
-      use, offer to sell, sell, import, and otherwise transfer the Work,
-      where such license applies only to those patent claims licensable
-      by such Contributor that are necessarily infringed by their
-      Contribution(s) alone or by combination of their Contribution(s)
-      with the Work to which such Contribution(s) was submitted. If You
-      institute patent litigation against any entity (including a
-      cross-claim or counterclaim in a lawsuit) alleging that the Work
-      or a Contribution incorporated within the Work constitutes direct
-      or contributory patent infringement, then any patent licenses
-      granted to You under this License for that Work shall terminate
-      as of the date such litigation is filed.
-
-   4. Redistribution. You may reproduce and distribute copies of the
-      Work or Derivative Works thereof in any medium, with or without
-      modifications, and in Source or Object form, provided that You
-      meet the following conditions:
-
-      (a) You must give any other recipients of the Work or
-          Derivative Works a copy of this License; and
-
-      (b) You must cause any modified files to carry prominent notices
-          stating that You changed the files; and
-
-      (c) You must retain, in the Source form of any Derivative Works
-          that You distribute, all copyright, patent, trademark, and
-          attribution notices from the Source form of the Work,
-          excluding those notices that do not pertain to any part of
-          the Derivative Works; and
-
-      (d) If the Work includes a "NOTICE" text file as part of its
-          distribution, then any Derivative Works that You distribute must
-          include a readable copy of the attribution notices contained
-          within such NOTICE file, excluding those notices that do not
-          pertain to any part of the Derivative Works, in at least one
-          of the following places: within a NOTICE text file distributed
-          as part of the Derivative Works; within the Source form or
-          documentation, if provided along with the Derivative Works; or,
-          within a display generated by the Derivative Works, if and
-          wherever such third-party notices normally appear. The contents
-          of the NOTICE file are for informational purposes only and
-          do not modify the License. You may add Your own attribution
-          notices within Derivative Works that You distribute, alongside
-          or as an addendum to the NOTICE text from the Work, provided
-          that such additional attribution notices cannot be construed
-          as modifying the License.
-
-      You may add Your own copyright statement to Your modifications and
-      may provide additional or different license terms and conditions
-      for use, reproduction, or distribution of Your modifications, or
-      for any such Derivative Works as a whole, provided Your use,
-      reproduction, and distribution of the Work otherwise complies with
-      the conditions stated in this License.
-
-   5. Submission of Contributions. Unless You explicitly state otherwise,
-      any Contribution intentionally submitted for inclusion in the Work
-      by You to the Licensor shall be under the terms and conditions of
-      this License, without any additional terms or conditions.
-      Notwithstanding the above, nothing herein shall supersede or modify
-      the terms of any separate license agreement you may have executed
-      with Licensor regarding such Contributions.
-
-   6. Trademarks. This License does not grant permission to use the trade
-      names, trademarks, service marks, or product names of the Licensor,
-      except as required for reasonable and customary use in describing the
-      origin of the Work and reproducing the content of the NOTICE file.
-
-   7. Disclaimer of Warranty. Unless required by applicable law or
-      agreed to in writing, Licensor provides the Work (and each
-      Contributor provides its Contributions) on an "AS IS" BASIS,
-      WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or
-      implied, including, without limitation, any warranties or conditions
-      of TITLE, NON-INFRINGEMENT, MERCHANTABILITY, or FITNESS FOR A
-      PARTICULAR PURPOSE. You are solely responsible for determining the
-      appropriateness of using or redistributing the Work and assume any
-      risks associated with Your exercise of permissions under this License.
-
-   8. Limitation of Liability. In no event and under no legal theory,
-      whether in tort (including negligence), contract, or otherwise,
-      unless required by applicable law (such as deliberate and grossly
-      negligent acts) or agreed to in writing, shall any Contributor be
-      liable to You for damages, including any direct, indirect, special,
-      incidental, or consequential damages of any character arising as a
-      result of this License or out of the use or inability to use the
-      Work (including but not limited to damages for loss of goodwill,
-      work stoppage, computer failure or malfunction, or any and all
-      other commercial damages or losses), even if such Contributor
-      has been advised of the possibility of such damages.
-
-   9. Accepting Warranty or Additional Liability. While redistributing
-      the Work or Derivative Works thereof, You may choose to offer,
-      and charge a fee for, acceptance of support, warranty, indemnity,
-      or other liability obligations and/or rights consistent with this
-      License. However, in accepting such obligations, You may act only
-      on Your own behalf and on Your sole responsibility, not on behalf
-      of any other Contributor, and only if You agree to indemnify,
-      defend, and hold each Contributor harmless for any liability
-      incurred by, or claims asserted against, such Contributor by reason
-      of your accepting any such warranty or additional liability.
-
-   END OF TERMS AND CONDITIONS
-
diff --git a/audio/audio_policy_hal.cpp b/audio/audio_policy_hal.cpp
deleted file mode 100755
index 110a919..0000000
--- a/audio/audio_policy_hal.cpp
+++ /dev/null
@@ -1,425 +0,0 @@
-/*
- * Copyright (C) 2011 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 "legacy_audio_policy_hal"
-#define LOG_NDEBUG 0
-
-#include <stdint.h>
-
-#include <hardware/hardware.h>
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioPolicyInterface.h>
-#include <hardware_legacy/AudioSystemLegacy.h>
-
-#include "AudioPolicyCompatClient.h"
-
-namespace android_audio_legacy {
-
-extern "C" {
-
-struct legacy_ap_module {
-    struct audio_policy_module module;
-};
-
-struct legacy_ap_device {
-    struct audio_policy_device device;
-};
-
-struct legacy_audio_policy {
-    struct audio_policy policy;
-
-    void *service;
-    struct audio_policy_service_ops *aps_ops;
-    AudioPolicyCompatClient *service_client;
-    AudioPolicyInterface *apm;
-};
-
-static inline struct legacy_audio_policy * to_lap(struct audio_policy *pol)
-{
-    return reinterpret_cast<struct legacy_audio_policy *>(pol);
-}
-
-static inline const struct legacy_audio_policy * to_clap(const struct audio_policy *pol)
-{
-    return reinterpret_cast<const struct legacy_audio_policy *>(pol);
-}
-
-
-static int ap_set_device_connection_state(struct audio_policy *pol,
-                                          audio_devices_t device,
-                                          audio_policy_dev_state_t state,
-                                          const char *device_address)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->setDeviceConnectionState(
-                    (AudioSystem::audio_devices)device,
-                    (AudioSystem::device_connection_state)state,
-                    device_address);
-}
-
-static audio_policy_dev_state_t ap_get_device_connection_state(
-                                            const struct audio_policy *pol,
-                                            audio_devices_t device,
-                                            const char *device_address)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return (audio_policy_dev_state_t)lap->apm->getDeviceConnectionState(
-                    (AudioSystem::audio_devices)device,
-                    device_address);
-}
-
-static void ap_set_phone_state(struct audio_policy *pol, int state)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->setPhoneState(state);
-}
-
-    /* indicate a change in ringer mode */
-static void ap_set_ringer_mode(struct audio_policy *pol, uint32_t mode,
-                               uint32_t mask)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->setRingerMode(mode, mask);
-}
-
-    /* force using a specific device category for the specified usage */
-static void ap_set_force_use(struct audio_policy *pol,
-                          audio_policy_force_use_t usage,
-                          audio_policy_forced_cfg_t config)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->setForceUse((AudioSystem::force_use)usage,
-                          (AudioSystem::forced_config)config);
-}
-
-    /* retreive current device category forced for a given usage */
-static audio_policy_forced_cfg_t ap_get_force_use(
-                                               const struct audio_policy *pol,
-                                               audio_policy_force_use_t usage)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return (audio_policy_forced_cfg_t)lap->apm->getForceUse(
-                          (AudioSystem::force_use)usage);
-}
-
-/* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
- * can still be muted. */
-static void ap_set_can_mute_enforced_audible(struct audio_policy *pol,
-                                             bool can_mute)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->setSystemProperty("ro.camera.sound.forced", can_mute ? "0" : "1");
-}
-
-static int ap_init_check(const struct audio_policy *pol)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->initCheck();
-}
-
-static audio_io_handle_t ap_get_output(struct audio_policy *pol,
-                                       audio_stream_type_t stream,
-                                       uint32_t sampling_rate,
-                                       uint32_t format,
-                                       uint32_t channels,
-                                       audio_policy_output_flags_t flags)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-
-    LOGV("%s: tid %d", __func__, gettid());
-    return lap->apm->getOutput((AudioSystem::stream_type)stream,
-                               sampling_rate, format, channels,
-                               (AudioSystem::output_flags)flags);
-}
-
-static int ap_start_output(struct audio_policy *pol, audio_io_handle_t output,
-                           audio_stream_type_t stream, int session)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->startOutput(output, (AudioSystem::stream_type)stream,
-                                 session);
-}
-
-static int ap_stop_output(struct audio_policy *pol, audio_io_handle_t output,
-                          audio_stream_type_t stream, int session)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->stopOutput(output, (AudioSystem::stream_type)stream,
-                                session);
-}
-
-static void ap_release_output(struct audio_policy *pol,
-                              audio_io_handle_t output)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->releaseOutput(output);
-}
-
-static audio_io_handle_t ap_get_input(struct audio_policy *pol, int inputSource,
-                                      uint32_t sampling_rate,
-                                      uint32_t format,
-                                      uint32_t channels,
-                                      audio_in_acoustics_t acoustics)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->getInput(inputSource, sampling_rate, format, channels,
-                              (AudioSystem::audio_in_acoustics)acoustics);
-}
-
-static int ap_start_input(struct audio_policy *pol, audio_io_handle_t input)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->startInput(input);
-}
-
-static int ap_stop_input(struct audio_policy *pol, audio_io_handle_t input)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->stopInput(input);
-}
-
-static void ap_release_input(struct audio_policy *pol, audio_io_handle_t input)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->releaseInput(input);
-}
-
-static void ap_init_stream_volume(struct audio_policy *pol,
-                                  audio_stream_type_t stream, int index_min,
-                                  int index_max)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->initStreamVolume((AudioSystem::stream_type)stream, index_min,
-                               index_max);
-}
-
-static int ap_set_stream_volume_index(struct audio_policy *pol,
-                                      audio_stream_type_t stream,
-                                      int index)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->setStreamVolumeIndex((AudioSystem::stream_type)stream,
-                                          index);
-}
-
-static int ap_get_stream_volume_index(const struct audio_policy *pol,
-                                      audio_stream_type_t stream,
-                                      int *index)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->getStreamVolumeIndex((AudioSystem::stream_type)stream,
-                                          index);
-}
-
-static uint32_t ap_get_strategy_for_stream(const struct audio_policy *pol,
-                                           audio_stream_type_t stream)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->getStrategyForStream((AudioSystem::stream_type)stream);
-}
-
-static uint32_t ap_get_devices_for_stream(const struct audio_policy *pol,
-                                       audio_stream_type_t stream)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->getDevicesForStream((AudioSystem::stream_type)stream);
-}
-
-static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol,
-                                            struct effect_descriptor_s *desc)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->getOutputForEffect(desc);
-}
-
-static int ap_register_effect(struct audio_policy *pol,
-                              struct effect_descriptor_s *desc,
-                              audio_io_handle_t io,
-                              uint32_t strategy,
-                              int session,
-                              int id)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->registerEffect(desc, io, strategy, session, id);
-}
-
-static int ap_unregister_effect(struct audio_policy *pol, int id)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->unregisterEffect(id);
-}
-
-static int ap_set_effect_enabled(struct audio_policy *pol, int id, bool enabled)
-{
-    return NO_ERROR;
-}
-
-static bool ap_is_stream_active(const struct audio_policy *pol, int stream,
-                                uint32_t in_past_ms)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->isStreamActive(stream, in_past_ms);
-}
-
-static int ap_dump(const struct audio_policy *pol, int fd)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->dump(fd);
-}
-
-static int create_legacy_ap(const struct audio_policy_device *device,
-                            struct audio_policy_service_ops *aps_ops,
-                            void *service,
-                            struct audio_policy **ap)
-{
-    struct legacy_audio_policy *lap;
-    int ret;
-
-    if (!service || !aps_ops)
-        return -EINVAL;
-
-    lap = (struct legacy_audio_policy *)calloc(1, sizeof(*lap));
-    if (!lap)
-        return -ENOMEM;
-
-    lap->policy.set_device_connection_state = ap_set_device_connection_state;
-    lap->policy.get_device_connection_state = ap_get_device_connection_state;
-    lap->policy.set_phone_state = ap_set_phone_state;
-    lap->policy.set_ringer_mode = ap_set_ringer_mode;
-    lap->policy.set_force_use = ap_set_force_use;
-    lap->policy.get_force_use = ap_get_force_use;
-    lap->policy.set_can_mute_enforced_audible =
-        ap_set_can_mute_enforced_audible;
-    lap->policy.init_check = ap_init_check;
-    lap->policy.get_output = ap_get_output;
-    lap->policy.start_output = ap_start_output;
-    lap->policy.stop_output = ap_stop_output;
-    lap->policy.release_output = ap_release_output;
-    lap->policy.get_input = ap_get_input;
-    lap->policy.start_input = ap_start_input;
-    lap->policy.stop_input = ap_stop_input;
-    lap->policy.release_input = ap_release_input;
-    lap->policy.init_stream_volume = ap_init_stream_volume;
-    lap->policy.set_stream_volume_index = ap_set_stream_volume_index;
-    lap->policy.get_stream_volume_index = ap_get_stream_volume_index;
-    lap->policy.get_strategy_for_stream = ap_get_strategy_for_stream;
-    lap->policy.get_devices_for_stream = ap_get_devices_for_stream;
-    lap->policy.get_output_for_effect = ap_get_output_for_effect;
-    lap->policy.register_effect = ap_register_effect;
-    lap->policy.unregister_effect = ap_unregister_effect;
-    lap->policy.set_effect_enabled = ap_set_effect_enabled;
-    lap->policy.is_stream_active = ap_is_stream_active;
-    lap->policy.dump = ap_dump;
-
-    lap->service = service;
-    lap->aps_ops = aps_ops;
-    lap->service_client =
-        new AudioPolicyCompatClient(aps_ops, service);
-    if (!lap->service_client) {
-        ret = -ENOMEM;
-        goto err_new_compat_client;
-    }
-
-    lap->apm = createAudioPolicyManager(lap->service_client);
-    if (!lap->apm) {
-        ret = -ENOMEM;
-        goto err_create_apm;
-    }
-
-    *ap = &lap->policy;
-    return 0;
-
-err_create_apm:
-    delete lap->service_client;
-err_new_compat_client:
-    free(lap);
-    *ap = NULL;
-    return ret;
-}
-
-static int destroy_legacy_ap(const struct audio_policy_device *ap_dev,
-                             struct audio_policy *ap)
-{
-    struct legacy_audio_policy *lap = to_lap(ap);
-
-    if (!lap)
-        return 0;
-
-    if (lap->apm)
-        destroyAudioPolicyManager(lap->apm);
-    if (lap->service_client)
-        delete lap->service_client;
-    free(lap);
-    return 0;
-}
-
-static int legacy_ap_dev_close(hw_device_t* device)
-{
-    if (device)
-        free(device);
-    return 0;
-}
-
-static int legacy_ap_dev_open(const hw_module_t* module, const char* name,
-                                    hw_device_t** device)
-{
-    struct legacy_ap_device *dev;
-
-    if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0)
-        return -EINVAL;
-
-    dev = (struct legacy_ap_device *)calloc(1, sizeof(*dev));
-    if (!dev)
-        return -ENOMEM;
-
-    dev->device.common.tag = HARDWARE_DEVICE_TAG;
-    dev->device.common.version = 0;
-    dev->device.common.module = const_cast<hw_module_t*>(module);
-    dev->device.common.close = legacy_ap_dev_close;
-    dev->device.create_audio_policy = create_legacy_ap;
-    dev->device.destroy_audio_policy = destroy_legacy_ap;
-
-    *device = &dev->device.common;
-
-    return 0;
-}
-
-static struct hw_module_methods_t legacy_ap_module_methods = {
-        open: legacy_ap_dev_open
-};
-
-struct legacy_ap_module HAL_MODULE_INFO_SYM = {
-    module: {
-        common: {
-            tag: HARDWARE_MODULE_TAG,
-            version_major: 1,
-            version_minor: 0,
-            id: AUDIO_POLICY_HARDWARE_MODULE_ID,
-            name: "LEGACY Audio Policy HAL",
-            author: "The Android Open Source Project",
-            methods: &legacy_ap_module_methods,
-            dso : NULL,
-            reserved : {0},
-        },
-    },
-};
-
-}; // extern "C"
-
-}; // namespace android_audio_legacy
diff --git a/configs/melfas_ts.idc b/configs/melfas_ts.idc
new file mode 100755
index 0000000..703e73b
--- /dev/null
+++ b/configs/melfas_ts.idc
@@ -0,0 +1,24 @@
+# Copyright (C) 2010 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.
+
+#
+# Input Device Configuration File for the Atmel Maxtouch touch screen.
+#
+# These calibration values are derived from empirical measurements
+# and may not be appropriate for use with other touch screens.
+# Refer to the input device configuration documentation for more details.
+#
+
+# Basic Parameters
+touch.deviceType = touchScreen
diff --git a/configs/sec_touchscreen.idc b/configs/sec_touchscreen.idc
index d258f04..4bd511a 100755
--- a/configs/sec_touchscreen.idc
+++ b/configs/sec_touchscreen.idc
@@ -13,11 +13,11 @@
 # limitations under the License.
 
 #
-# Input Device Calibration File for the Crespo touch screen.
+# Input Device Configuration File for the Atmel Maxtouch touch screen.
 #
 # These calibration values are derived from empirical measurements
 # and may not be appropriate for use with other touch screens.
-# Refer to the input device calibration documentation for more details.
+# Refer to the input device configuration documentation for more details.
 #
 
 # Basic Parameters
@@ -31,26 +31,26 @@ touch.touchSize.calibration = pressure
 # Driver reports tool size as an area measurement.
 #
 # Based on empirical measurements, we estimate the size of the tool
-# using size = sqrt(22 * rawToolArea + 0) * 9.2 + 0.
+# using size = sqrt(22 * rawToolArea + 0) * 6 + 0.
 touch.toolSize.calibration = area
 touch.toolSize.areaScale = 22
 touch.toolSize.areaBias = 0
-touch.toolSize.linearScale = 9.2
+touch.toolSize.linearScale = 6
 touch.toolSize.linearBias = 0
 touch.toolSize.isSummed = 0
 
 # Pressure
 # Driver reports signal strength as pressure.
 #
-# A normal thumb touch typically registers about 100 signal strength
+# A normal index finger touch typically registers about 80 signal strength
 # units although we don't expect these values to be accurate.
 touch.pressure.calibration = amplitude
 touch.pressure.source = default
-touch.pressure.scale = 0.01
+touch.pressure.scale = 0.0125
 
 # Size
 touch.size.calibration = normalized
 
 # Orientation
-touch.orientation.calibration = none
+touch.orientation.calibration = vector
 
diff --git a/configs/sec_ts_ics_bio.idc b/configs/sec_ts_ics_bio.idc
deleted file mode 100755
index eb46d0a..0000000
--- a/configs/sec_ts_ics_bio.idc
+++ /dev/null
@@ -1,56 +0,0 @@
-# Copyright (C) 2010 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.

-

-#

-# Input Device Calibration File for the Crespo touch screen.

-#

-# These calibration values are derived from empirical measurements

-# and may not be appropriate for use with other touch screens.

-# Refer to the input device calibration documentation for more details.

-#

-

-# Basic Parameters

-touch.deviceType = touchScreen

-touch.orientationAware = 1

-

-# Touch Size

-touch.touchSize.calibration = pressure

-

-# Tool Size

-# Driver reports tool size as an area measurement.

-#

-# Based on empirical measurements, we estimate the size of the tool

-# using size = sqrt(22 * rawToolArea + 0) * 9.2 + 0.

-touch.toolSize.calibration = area

-touch.toolSize.areaScale = 22

-touch.toolSize.areaBias = 0

-touch.toolSize.linearScale = 9.2

-touch.toolSize.linearBias = 0

-touch.toolSize.isSummed = 0

-

-# Pressure

-# Driver reports signal strength as pressure.

-#

-# A normal thumb touch typically registers about 100 signal strength

-# units although we don't expect these values to be accurate.

-touch.pressure.calibration = amplitude

-touch.pressure.source = default

-touch.pressure.scale = 0.01

-

-# Size

-touch.size.calibration = normalized

-

-# Orientation

-touch.orientation.calibration = none

-

diff --git a/galaxynote.mk b/galaxynote.mk
index 1feabf8..679a028 100755
--- a/galaxynote.mk
+++ b/galaxynote.mk
@@ -35,12 +35,19 @@ PRODUCT_COPY_FILES += \
 # soundbooster.txt - needs to be at /data/soundbooster.txt
 PRODUCT_COPY_FILES += \
 	device/samsung/galaxynote/configs/asound.conf:system/etc/asound.conf \
+	device/samsung/galaxynote/configs/audio_effects.conf:system/etc/audio_effects.conf \
 	device/samsung/galaxynote/configs/soundbooster.txt:system/etc/audio/soundbooster.txt
 	
+# omx
+PRODUCT_COPY_FILES += \
+	device/samsung/galaxys2/configs/media_profiles.xml:system/etc/media_profiles.xml \
+	device/samsung/galaxys2/configs/secomxregistry:system/etc/secomxregistry \
+	device/samsung/galaxys2/configs/somxreg.conf:system/etc/somxreg.conf
+	
 # Touchscreen
 PRODUCT_COPY_FILES += \
-	device/samsung/galaxynote/configs/sec_ts_ics_bio.idc:system/usr/idc/sec_ts_ics_bio.idc \
-	device/samsung/galaxynote/configs/sec_ts_ics_bio.idc:system/usr/idc/sec_touchscreen.idc \
+	device/samsung/galaxynote/configs/melfas_ts.idc:system/usr/idc/melfas_ts.idc \
+	device/samsung/galaxynote/configs/sec_touchscreen.idc:system/usr/idc/sec_touchscreen.idc \
 	device/samsung/galaxynote/configs/sec_ts_ics_bio.idc:system/usr/idc/sec_e-pen.idc \
 
 # Keylayout
@@ -78,10 +85,6 @@ PRODUCT_COPY_FILES += \
 
 # Packages
 PRODUCT_PACKAGES := \
-	audio.primary.smdkv310 \
-	audio_policy.smdkv310 \
-	gps.smdkv310 \
-    smdkv310_hdcp_keys \
     com.android.future.usb.accessory
 
 # Charger
@@ -94,15 +97,12 @@ PRODUCT_PACKAGES += \
 	Camera
 
 # Sensors
-PRODUCT_PACKAGES += \
-	lights.smdkv310 \
-	sensors.smdkv310
+# PRODUCT_PACKAGES += \
+#	sensors.smdkv310
 
 # Ril
 PRODUCT_PROPERTY_OVERRIDES += \
     ro.telephony.ril_class=samsung \
-    ro.telephony.ril.v3=icccardstatus,datacall,signalstrength,facilitylock \
-	ro.telephony.sends_barcount=1 \
     mobiledata.interfaces=pdp0,wlan0,gprs,ppp0
 
 # Filesystem management tools
diff --git a/gpswrapper/Android.mk b/gpswrapper/Android.mk
index c13be1b..e0b4e2b 100755
--- a/gpswrapper/Android.mk
+++ b/gpswrapper/Android.mk
@@ -3,7 +3,7 @@ include $(CLEAR_VARS)
 

 LOCAL_MODULE_TAGS := optional

 

-LOCAL_MODULE := gps.smdkv310

+LOCAL_MODULE := gps.$(TARGET_BOARD_PLATFORM)

 

 LOCAL_SHARED_LIBRARIES:= \

 	liblog \

diff --git a/gpswrapper/gps.c b/gpswrapper/gps.c
index d921e27..a4e8f7f 100755
--- a/gpswrapper/gps.c
+++ b/gpswrapper/gps.c
@@ -31,7 +31,7 @@
 #define LOG_TAG "gps-wrapper"

 #include <utils/Log.h>

 

-#define ORIGINAL_HAL_PATH "/system/lib/hw/vendor-gps.smdkv310.so"

+#define ORIGINAL_HAL_PATH "/system/lib/hw/vendor-gps.s5pc210.so"

 

 static const AGpsRilInterface* oldAGPSRIL = NULL;

 static AGpsRilInterface newAGPSRIL;

diff --git a/init.smdkv310.rc b/init.smdk4210.rc
index a2cacaa..1bac747 100755
--- a/init.smdkv310.rc
+++ b/init.smdk4210.rc
@@ -126,21 +126,11 @@ on fs
     chown system system /sys/devices/platform/android_usb/tethering

 

     # setup for alsa snd device

-    symlink /dev/snd/pcmC0D0c /dev/pcmC0D0c

-    symlink /dev/snd/pcmC0D0p /dev/pcmC0D0p

-    symlink /dev/snd/controlC0 /dev/controlC0

-    symlink /dev/snd/timer /dev/timer

-    symlink /dev/snd/hwC0D0 /dev/hwC0D0

-    chmod 0777 /dev/pcmC0D0c

-    chmod 0777 /dev/pcmC0D0p

-    chmod 0777 /dev/controlC0

-    chmod 0777 /dev/timer

-    chmod 0777 /dev/hwC0D0

-    chmod 0777 /dev/snd/pcmC0D0c

-    chmod 0777 /dev/snd/pcmC0D0p

-    chmod 0777 /dev/snd/controlC0

-    chmod 0777 /dev/snd/timer

-    chmod 0777 /dev/snd/hwC0D0

+	chmod 0770 /dev/snd/pcmC0D0c

+    chmod 0770 /dev/snd/pcmC0D0p

+    chmod 0770 /dev/snd/controlC0

+    chmod 0770 /dev/snd/timer

+    chmod 0770 /dev/snd/hwC0D0

 

 	# Permissions for gpio_keys

     chown radio system /sys/devices/platform/sec_key.0/wakeup_keys 

diff --git a/init.smdkv310.usb.rc b/init.smdk4210.usb.rc
index 9307c56..9307c56 100755
--- a/init.smdkv310.usb.rc
+++ b/init.smdk4210.usb.rc
diff --git a/kernel b/kernel
index 062e6e1..4d712a6 100755
--- a/kernel
+++ b/kernel
diff --git a/modules/Si4709_driver.ko b/modules/Si4709_driver.ko
index a94df4d..3de78b2 100755
--- a/modules/Si4709_driver.ko
+++ b/modules/Si4709_driver.ko
diff --git a/modules/cifs.ko b/modules/cifs.ko
index d597f57..424aa09 100755
--- a/modules/cifs.ko
+++ b/modules/cifs.ko
diff --git a/modules/gspca_main.ko b/modules/gspca_main.ko
index 58fef5b..24381a0 100755
--- a/modules/gspca_main.ko
+++ b/modules/gspca_main.ko
diff --git a/modules/j4fs.ko b/modules/j4fs.ko
index 65cf1cc..ca2fefa 100755
--- a/modules/j4fs.ko
+++ b/modules/j4fs.ko
diff --git a/modules/scsi_wait_scan.ko b/modules/scsi_wait_scan.ko
index b7479d8..46724ab 100755
--- a/modules/scsi_wait_scan.ko
+++ b/modules/scsi_wait_scan.ko
diff --git a/modules/vibrator.ko b/modules/vibrator.ko
index b41e9f7..9ebf564 100755
--- a/modules/vibrator.ko
+++ b/modules/vibrator.ko
diff --git a/ueventd.smdkv310.rc b/ueventd.smdk4210.rc
index 14c1481..14c1481 100755
--- a/ueventd.smdkv310.rc
+++ b/ueventd.smdk4210.rc