Newer audio shim.

11 files changed, 2075 insertions, 1 deletions

diff --git a/audio/Android.mk b/audio/Android.mk
index 71b841b..9951ba3 100755
--- a/audio/Android.mk
+++ b/audio/Android.mk
@@ -40,4 +40,59 @@ LOCAL_WHOLE_STATIC_LIBRARIES := \
 
 include $(BUILD_SHARED_LIBRARY)
 
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+    AudioPolicyManagerBase.cpp \
+    AudioPolicyCompatClient.cpp \
+    audio_policy_hal.cpp
+
+ifeq ($(AUDIO_POLICY_TEST),true)
+  LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
+endif
+
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+  LOCAL_CFLAGS += -DWITH_A2DP
+endif
+
+LOCAL_STATIC_LIBRARIES := libmedia_helper
+LOCAL_MODULE := libaudiopolicy_legacy2
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_STATIC_LIBRARY)
+
+
+# The default audio policy, for now still implemented on top of legacy
+# policy code
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+    AudioPolicyManagerDefault.cpp
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libutils \
+    libmedia
+
+LOCAL_STATIC_LIBRARIES := \
+    libmedia_helper
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+    libaudiopolicy_legacy2
+
+ifeq ($(BOARD_USES_AUDIO_LEGACY),true)
+LOCAL_SHARED_LIBRARIES += libaudiopolicy
+endif
+
+LOCAL_C_INCLUDES := $(LOCAL_PATH)
+LOCAL_MODULE := audio_policy.$(TARGET_BOARD_PLATFORM)
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+LOCAL_MODULE_TAGS := optional
+
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+  LOCAL_CFLAGS += -DWITH_A2DP
+endif
+
+include $(BUILD_SHARED_LIBRARY)
+
 endif
diff --git a/audio/AudioPolicyCompatClient.cpp b/audio/AudioPolicyCompatClient.cpp
new file mode 100644
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 100644
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 100644
index 0000000..bce7a2f
--- /dev/null
+++ b/audio/AudioPolicyInterface.h
@@ -0,0 +1,239 @@
+/*
+ * 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 {
+    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 status_t setEffectEnabled(int id, bool enabled) = 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 100644
index 0000000..db4487a
--- /dev/null
+++ b/audio/AudioPolicyManagerBase.cpp
@@ -0,0 +1,101 @@
+/*
+ * 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 <AudioSystem.h>
+#define LOG_TAG "AudioPolicyManagerBase"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+#include <AudioPolicyManagerBase.h>
+#include <hardware/audio_effect.h>
+#include <math.h>
+
+namespace android {
+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;
+}
+
+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);
+}
+
+
+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::initCheck()
+{
+    return (mHardwareOutput == 0) ? NO_INIT : NO_ERROR;
+}
+
+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;
+}
+
+
+
+}; // namespace android
diff --git a/audio/AudioPolicyManagerBase.h b/audio/AudioPolicyManagerBase.h
new file mode 100644
index 0000000..79af563
--- /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 <AudioPolicyInterface.h>
+#include <AudioSystem.h>
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+#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 100644
index 0000000..cc56047
--- /dev/null
+++ b/audio/AudioPolicyManagerDefault.cpp
@@ -0,0 +1,35 @@
+/*
+ * 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 <AudioSystem.h>
+#define LOG_TAG "AudioPolicyManagerDefault"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManagerDefault.h"
+
+namespace android {
+
+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 100644
index 0000000..86627f0
--- /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 <AudioPolicyManagerBase.h>
+
+namespace android {
+
+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 100644
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/NOTICE b/audio/NOTICE
index a0cab01..3237da6 100755
--- a/audio/NOTICE
+++ b/audio/NOTICE
@@ -188,4 +188,3 @@
 
    END OF TERMS AND CONDITIONS
 
-
diff --git a/audio/audio_policy_hal.cpp b/audio/audio_policy_hal.cpp
new file mode 100644
index 0000000..d5f2816
--- /dev/null
+++ b/audio/audio_policy_hal.cpp
@@ -0,0 +1,426 @@
+/*
+ * 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)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->setEffectEnabled(id, enabled);
+}
+
+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