Removed setEffectEnabled as my soundpolicy.so doesn't know that func

10 files changed, 4411 insertions, 1 deletions

diff --git a/audio/Android.mk b/audio/Android.mk
index c72f44a..5026c54 100755
--- a/audio/Android.mk
+++ b/audio/Android.mk
@@ -32,6 +32,11 @@ LOCAL_SHARED_LIBRARIES += libdl
 
 LOCAL_SHARED_LIBRARIES += libaudio
 
+ifeq ($(BOARD_FORCE_STATIC_A2DP),true)
+  LOCAL_SHARED_LIBRARIES += liba2dp
+endif
+
+
 LOCAL_STATIC_LIBRARIES := \
     libmedia_helper
 
@@ -40,4 +45,59 @@ LOCAL_WHOLE_STATIC_LIBRARIES := \
 
 include $(BUILD_SHARED_LIBRARY)
 
-endif
\ No newline at end of file
+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
new file mode 100755
index 0000000..08c31c5
--- /dev/null
+++ b/audio/AudioPolicyCompatClient.cpp
@@ -0,0 +1,142 @@
+/*
+ * 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
new file mode 100755
index 0000000..4feee37
--- /dev/null
+++ b/audio/AudioPolicyCompatClient.h
@@ -0,0 +1,79 @@
+/*
+ * 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
new file mode 100755
index 0000000..78f87da
--- /dev/null
+++ b/audio/AudioPolicyInterface.h
@@ -0,0 +1,238 @@
+/*
+ * 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
new file mode 100755
index 0000000..ac31ad4
--- /dev/null
+++ b/audio/AudioPolicyManagerBase.cpp
@@ -0,0 +1,2434 @@
+/*
+ * 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
new file mode 100755
index 0000000..3ab748d
--- /dev/null
+++ b/audio/AudioPolicyManagerBase.h
@@ -0,0 +1,393 @@
+/*
+ * 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
new file mode 100755
index 0000000..9083638
--- /dev/null
+++ b/audio/AudioPolicyManagerDefault.cpp
@@ -0,0 +1,34 @@
+/*
+ * 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
new file mode 100755
index 0000000..b2b2576
--- /dev/null
+++ b/audio/AudioPolicyManagerDefault.h
@@ -0,0 +1,43 @@
+/*
+ * 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
new file mode 100755
index 0000000..bd75a73
--- /dev/null
+++ b/audio/AudioSystem.h
@@ -0,0 +1,562 @@
+/*
+ * 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/audio_policy_hal.cpp b/audio/audio_policy_hal.cpp
new file mode 100755
index 0000000..110a919
--- /dev/null
+++ b/audio/audio_policy_hal.cpp
@@ -0,0 +1,425 @@
+/*
+ * 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