diff --git a/api/current.xml b/api/current.xml
index 9248ccb..fcebc73 100644
--- a/api/current.xml
+++ b/api/current.xml
@@ -73186,7 +73186,7 @@
  value="1"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -73197,7 +73197,7 @@
  value="0"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -73208,7 +73208,7 @@
  value="2"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -73219,6 +73219,358 @@
  value="3"
  static="true"
  final="true"
+ deprecated="deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_INVALID"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="-1"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_BACK"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="524288"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_BACK_PROCESSED"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="8388608"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_DEFAULT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="0"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_FRONT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="262144"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_FRONT_PROCESSED"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="4194304"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_LEFT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="65536"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_LEFT_PROCESSED"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="1048576"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_MONO"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="262144"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_PRESSURE"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="16777216"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_RIGHT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="131072"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_RIGHT_PROCESSED"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="2097152"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_STEREO"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="196608"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_X_AXIS"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="33554432"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_Y_AXIS"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="67108864"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_IN_Z_AXIS"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="134217728"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_5POINT1"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="63"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_7POINT1"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="255"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_BACK_CENTER"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="256"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_BACK_LEFT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="16"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_BACK_RIGHT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="32"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_DEFAULT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="0"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_FRONT_CENTER"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="4"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_FRONT_LEFT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="1"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_FRONT_LEFT_OF_CENTER"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="64"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_FRONT_RIGHT"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="2"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_FRONT_RIGHT_OF_CENTER"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="128"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_LOW_FREQUENCY"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="8"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_MONO"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="1"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_QUAD"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="51"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_STEREO"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="3"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
+<field name="CHANNEL_OUT_SURROUND"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="263"
+ static="true"
+ final="true"
  deprecated="not deprecated"
  visibility="public"
 >
@@ -73336,6 +73688,19 @@
  visibility="public"
 >
 </method>
+<method name="getParameters"
+ return="java.lang.String"
+ abstract="false"
+ native="false"
+ synchronized="false"
+ static="false"
+ final="false"
+ deprecated="not deprecated"
+ visibility="public"
+>
+<parameter name="keys" type="java.lang.String">
+</parameter>
+</method>
 <method name="getRingerMode"
  return="int"
  abstract="false"
@@ -73454,6 +73819,17 @@
  visibility="public"
 >
 </method>
+<method name="isWiredHeadsetOn"
+ return="boolean"
+ abstract="false"
+ native="false"
+ synchronized="false"
+ static="false"
+ final="false"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</method>
 <method name="loadSoundEffects"
  return="void"
  abstract="false"
@@ -73500,7 +73876,7 @@
  synchronized="false"
  static="false"
  final="false"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 <parameter name="on" type="boolean">
@@ -73545,6 +73921,19 @@
 <parameter name="mode" type="int">
 </parameter>
 </method>
+<method name="setParameters"
+ return="void"
+ abstract="false"
+ native="false"
+ synchronized="false"
+ static="false"
+ final="false"
+ deprecated="not deprecated"
+ visibility="public"
+>
+<parameter name="keyValuePairs" type="java.lang.String">
+</parameter>
+</method>
 <method name="setRingerMode"
  return="void"
  abstract="false"
@@ -73650,6 +74039,19 @@
 <parameter name="vibrateSetting" type="int">
 </parameter>
 </method>
+<method name="setWiredHeadsetOn"
+ return="void"
+ abstract="false"
+ native="false"
+ synchronized="false"
+ static="false"
+ final="false"
+ deprecated="deprecated"
+ visibility="public"
+>
+<parameter name="on" type="boolean">
+</parameter>
+</method>
 <method name="shouldVibrate"
  return="boolean"
  abstract="false"
@@ -74022,7 +74424,7 @@
  value="-1"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -74044,7 +74446,7 @@
  value="16"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -74055,7 +74457,7 @@
  value="4"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -74066,7 +74468,7 @@
  value="1"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -74077,7 +74479,7 @@
  value="8"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -74088,7 +74490,7 @@
  value="2"
  static="true"
  final="true"
- deprecated="not deprecated"
+ deprecated="deprecated"
  visibility="public"
 >
 </field>
@@ -74103,6 +74505,17 @@
  visibility="public"
 >
 </field>
+<field name="STREAM_DTMF"
+ type="int"
+ transient="false"
+ volatile="false"
+ value="8"
+ static="true"
+ final="true"
+ deprecated="not deprecated"
+ visibility="public"
+>
+</field>
 <field name="STREAM_MUSIC"
  type="int"
  transient="false"
diff --git a/camera/libcameraservice/CameraService.cpp b/camera/libcameraservice/CameraService.cpp
index 97b43a4..e7c1fbb 100644
--- a/camera/libcameraservice/CameraService.cpp
+++ b/camera/libcameraservice/CameraService.cpp
@@ -33,7 +33,6 @@
 #include "CameraService.h"
 
 #include <cutils/atomic.h>
-#include <cutils/properties.h>
 
 namespace android {
 
@@ -199,13 +198,7 @@
 {
     sp<MediaPlayer> mp = new MediaPlayer();
     if (mp->setDataSource(file) == NO_ERROR) {
-        char value[PROPERTY_VALUE_MAX];
-        property_get("ro.camera.sound.forced", value, "0");
-        if (atoi(value)) {
-            mp->setAudioStreamType(AudioSystem::ENFORCED_AUDIBLE);
-        } else {
-            mp->setAudioStreamType(AudioSystem::SYSTEM);            
-        }
+        mp->setAudioStreamType(AudioSystem::ENFORCED_AUDIBLE);
         mp->prepare();
     } else {
         mp.clear();
diff --git a/cmds/servicemanager/service_manager.c b/cmds/servicemanager/service_manager.c
index e4aa8b5..f3a4713 100644
--- a/cmds/servicemanager/service_manager.c
+++ b/cmds/servicemanager/service_manager.c
@@ -30,6 +30,7 @@
     { AID_MEDIA, "media.audio_flinger" },
     { AID_MEDIA, "media.player" },
     { AID_MEDIA, "media.camera" },
+    { AID_MEDIA, "media.audio_policy" },
     { AID_RADIO, "radio.phone" },
     { AID_RADIO, "radio.sms" },
     { AID_RADIO, "radio.phonesubinfo" },
diff --git a/cmds/system_server/library/system_init.cpp b/cmds/system_server/library/system_init.cpp
index ea78461..1d57fdc 100644
--- a/cmds/system_server/library/system_init.cpp
+++ b/cmds/system_server/library/system_init.cpp
@@ -17,6 +17,7 @@
 #include <SurfaceFlinger.h>
 #include <AudioFlinger.h>
 #include <CameraService.h>
+#include <AudioPolicyService.h>
 #include <MediaPlayerService.h>
 
 #include <android_runtime/AndroidRuntime.h>
@@ -80,6 +81,9 @@
 
         // Start the camera service
         CameraService::instantiate();
+
+        // Start the audio policy service
+        AudioPolicyService::instantiate();
     }
 
     // And now start the Android runtime.  We have to do this bit
diff --git a/core/java/android/server/BluetoothA2dpService.java b/core/java/android/server/BluetoothA2dpService.java
index 722a7cc..fb436e5 100644
--- a/core/java/android/server/BluetoothA2dpService.java
+++ b/core/java/android/server/BluetoothA2dpService.java
@@ -54,7 +54,6 @@
     private static final String BLUETOOTH_ADMIN_PERM = android.Manifest.permission.BLUETOOTH_ADMIN;
     private static final String BLUETOOTH_PERM = android.Manifest.permission.BLUETOOTH;
 
-    private static final String A2DP_SINK_ADDRESS = "a2dp_sink_address";
     private static final String BLUETOOTH_ENABLED = "bluetooth_enabled";
 
     private static final int MESSAGE_CONNECT_TO = 1;
@@ -238,7 +237,7 @@
                     }
             }
         }
-        mAudioManager.setParameter(BLUETOOTH_ENABLED, "true");
+        mAudioManager.setParameters(BLUETOOTH_ENABLED+"=true");
     }
 
     private synchronized void onBluetoothDisable() {
@@ -262,8 +261,8 @@
             }
             mAudioDevices.clear();
         }
-        mAudioManager.setBluetoothA2dpOn(false);
-        mAudioManager.setParameter(BLUETOOTH_ENABLED, "false");
+
+        mAudioManager.setParameters(BLUETOOTH_ENABLED+"=false");
     }
 
     public synchronized int connectSink(String address) {
@@ -403,8 +402,7 @@
                    Intent intent = new Intent(AudioManager.ACTION_AUDIO_BECOMING_NOISY);
                    mContext.sendBroadcast(intent);
                 }
-                if (--mSinkCount == 0)
-                    mAudioManager.setBluetoothA2dpOn(false);
+                mSinkCount--;
             } else if (state == BluetoothA2dp.STATE_CONNECTED) {
                 mSinkCount ++;
             }
@@ -417,11 +415,6 @@
             mContext.sendBroadcast(intent, BLUETOOTH_PERM);
 
             if (DBG) log("A2DP state : address: " + address + " State:" + prevState + "->" + state);
-
-            if (state == BluetoothA2dp.STATE_CONNECTED) {
-                mAudioManager.setParameter(A2DP_SINK_ADDRESS, address);
-                mAudioManager.setBluetoothA2dpOn(true);
-            }
         }
     }
 
diff --git a/core/jni/android_media_AudioRecord.cpp b/core/jni/android_media_AudioRecord.cpp
index 44a9e8c..0be996d 100644
--- a/core/jni/android_media_AudioRecord.cpp
+++ b/core/jni/android_media_AudioRecord.cpp
@@ -28,8 +28,8 @@
 #include "android_runtime/AndroidRuntime.h"
 
 #include "utils/Log.h"
-#include "media/AudioSystem.h"
 #include "media/AudioRecord.h"
+#include "media/mediarecorder.h"
 
 
 // ----------------------------------------------------------------------------
@@ -62,7 +62,7 @@
 #define AUDIORECORD_ERROR_BAD_VALUE                 -2
 #define AUDIORECORD_ERROR_INVALID_OPERATION         -3
 #define AUDIORECORD_ERROR_SETUP_ZEROFRAMECOUNT      -16
-#define AUDIORECORD_ERROR_SETUP_INVALIDCHANNELCOUNT -17
+#define AUDIORECORD_ERROR_SETUP_INVALIDCHANNELMASK -17
 #define AUDIORECORD_ERROR_SETUP_INVALIDFORMAT       -18
 #define AUDIORECORD_ERROR_SETUP_INVALIDSOURCE       -19
 #define AUDIORECORD_ERROR_SETUP_NATIVEINITFAILED    -20
@@ -122,17 +122,18 @@
 // ----------------------------------------------------------------------------
 static int
 android_media_AudioRecord_setup(JNIEnv *env, jobject thiz, jobject weak_this,
-        jint source, jint sampleRateInHertz, jint nbChannels, 
+        jint source, jint sampleRateInHertz, jint channels,
         jint audioFormat, jint buffSizeInBytes)
 {
     //LOGV(">> Entering android_media_AudioRecord_setup");
-    //LOGV("sampleRate=%d, audioFormat=%d, nbChannels=%d, buffSizeInBytes=%d",
-    //     sampleRateInHertz, audioFormat, nbChannels,     buffSizeInBytes);
+    //LOGV("sampleRate=%d, audioFormat=%d, channels=%x, buffSizeInBytes=%d",
+    //     sampleRateInHertz, audioFormat, channels,     buffSizeInBytes);
 
-    if ((nbChannels == 0) || (nbChannels > 2)) {
+    if (!AudioSystem::isInputChannel(channels)) {
         LOGE("Error creating AudioRecord: channel count is not 1 or 2.");
-        return AUDIORECORD_ERROR_SETUP_INVALIDCHANNELCOUNT;
+        return AUDIORECORD_ERROR_SETUP_INVALIDCHANNELMASK;
     }
+    uint32_t nbChannels = AudioSystem::popCount(channels);
 
     // compare the format against the Java constants
     if ((audioFormat != javaAudioRecordFields.PCM16) 
@@ -152,12 +153,7 @@
     int frameSize = nbChannels * bytesPerSample;
     size_t frameCount = buffSizeInBytes / frameSize;
     
-    // convert and check input source value
-    // input_source values defined in AudioRecord.h are equal to
-    // JAVA MediaRecord.AudioSource values minus 1.
-    AudioRecord::input_source arSource = (AudioRecord::input_source)(source - 1);
-    if (arSource < AudioRecord::DEFAULT_INPUT ||
-        arSource >= AudioRecord::NUM_INPUT_SOURCES) {
+    if (source >= AUDIO_SOURCE_LIST_END) {
         LOGE("Error creating AudioRecord: unknown source.");
         return AUDIORECORD_ERROR_SETUP_INVALIDSOURCE;
     }
@@ -184,10 +180,10 @@
     // we use a weak reference so the AudioRecord object can be garbage collected.
     lpCallbackData->audioRecord_ref = env->NewGlobalRef(weak_this);
     
-    lpRecorder->set(arSource,
+    lpRecorder->set(source,
         sampleRateInHertz,
         format,        // word length, PCM
-        nbChannels,
+        channels,
         frameCount,
         0,             // flags
         recorderCallback,// callback_t
diff --git a/core/jni/android_media_AudioSystem.cpp b/core/jni/android_media_AudioSystem.cpp
index 692610e..3d8d296 100644
--- a/core/jni/android_media_AudioSystem.cpp
+++ b/core/jni/android_media_AudioSystem.cpp
@@ -50,25 +50,6 @@
 }
 
 static int
-android_media_AudioSystem_setVolume(JNIEnv *env, jobject clazz, jint type, jint volume)
-{
-    LOGV("setVolume(%d)", int(volume));
-    
-    return check_AudioSystem_Command(AudioSystem::setStreamVolume(type, AudioSystem::linearToLog(volume)));
-}
-
-static int
-android_media_AudioSystem_getVolume(JNIEnv *env, jobject clazz, jint type)
-{
-    float v;
-    int v_int = -1;
-    if (AudioSystem::getStreamVolume(int(type), &v) == NO_ERROR) {
-        v_int = AudioSystem::logToLinear(v);
-    }
-    return v_int;
-}
-
-static int
 android_media_AudioSystem_muteMicrophone(JNIEnv *env, jobject thiz, jboolean on)
 {
     return check_AudioSystem_Command(AudioSystem::muteMicrophone(on));
@@ -82,34 +63,6 @@
     return state;
 }
 
-static int
-android_media_AudioSystem_setRouting(JNIEnv *env, jobject clazz, jint mode, jint routes, jint mask)
-{
-    return check_AudioSystem_Command(AudioSystem::setRouting(mode, uint32_t(routes), uint32_t(mask)));
-}
-
-static jint
-android_media_AudioSystem_getRouting(JNIEnv *env, jobject clazz, jint mode)
-{
-    uint32_t routes = -1;
-    AudioSystem::getRouting(mode, &routes);
-    return jint(routes);
-}
-
-static int
-android_media_AudioSystem_setMode(JNIEnv *env, jobject clazz, jint mode)
-{
-    return check_AudioSystem_Command(AudioSystem::setMode(mode));
-}
-
-static jint
-android_media_AudioSystem_getMode(JNIEnv *env, jobject clazz)
-{
-    int mode = AudioSystem::MODE_INVALID;
-    AudioSystem::getMode(&mode);
-    return jint(mode);
-}
-
 static jboolean
 android_media_AudioSystem_isMusicActive(JNIEnv *env, jobject thiz)
 {
@@ -118,16 +71,29 @@
     return state;
 }
 
-// Temporary interface, do not use
-// TODO: Replace with a more generic key:value get/set mechanism
-static void
-android_media_AudioSystem_setParameter(JNIEnv *env, jobject thiz, jstring key, jstring value)
+static int
+android_media_AudioSystem_setParameters(JNIEnv *env, jobject thiz, jstring keyValuePairs)
 {
-    const char *c_key = env->GetStringUTFChars(key, NULL);
-    const char *c_value = env->GetStringUTFChars(value, NULL);
-    AudioSystem::setParameter(c_key, c_value);
-    env->ReleaseStringUTFChars(key, c_key);
-    env->ReleaseStringUTFChars(value, c_value);
+    const jchar* c_keyValuePairs = env->GetStringCritical(keyValuePairs, 0);
+    String8 c_keyValuePairs8;
+    if (keyValuePairs) {
+        c_keyValuePairs8 = String8(c_keyValuePairs, env->GetStringLength(keyValuePairs));
+        env->ReleaseStringCritical(keyValuePairs, c_keyValuePairs);
+    }
+    int status = check_AudioSystem_Command(AudioSystem::setParameters(0, c_keyValuePairs8));
+    return status;
+}
+
+static jstring
+android_media_AudioSystem_getParameters(JNIEnv *env, jobject thiz, jstring keys)
+{
+    const jchar* c_keys = env->GetStringCritical(keys, 0);
+    String8 c_keys8;
+    if (keys) {
+        c_keys8 = String8(c_keys, env->GetStringLength(keys));
+        env->ReleaseStringCritical(keys, c_keys);
+    }
+    return env->NewStringUTF(AudioSystem::getParameters(0, c_keys8).string());
 }
 
 void android_media_AudioSystem_error_callback(status_t err)
@@ -152,19 +118,93 @@
     env->CallStaticVoidMethod(clazz, env->GetStaticMethodID(clazz, "errorCallbackFromNative","(I)V"), error);
 }
 
+static int
+android_media_AudioSystem_setDeviceConnectionState(JNIEnv *env, jobject thiz, jint device, jint state, jstring device_address)
+{
+    const char *c_address = env->GetStringUTFChars(device_address, NULL);
+    int status = check_AudioSystem_Command(AudioSystem::setDeviceConnectionState(static_cast <AudioSystem::audio_devices>(device),
+                                          static_cast <AudioSystem::device_connection_state>(state),
+                                          c_address));
+    env->ReleaseStringUTFChars(device_address, c_address);
+    return status;
+}
+
+static int
+android_media_AudioSystem_getDeviceConnectionState(JNIEnv *env, jobject thiz, jint device, jstring device_address)
+{
+    const char *c_address = env->GetStringUTFChars(device_address, NULL);
+    int state = static_cast <int>(AudioSystem::getDeviceConnectionState(static_cast <AudioSystem::audio_devices>(device),
+                                          c_address));
+    env->ReleaseStringUTFChars(device_address, c_address);
+    return state;
+}
+
+static int
+android_media_AudioSystem_setPhoneState(JNIEnv *env, jobject thiz, jint state)
+{
+    return check_AudioSystem_Command(AudioSystem::setPhoneState(state));
+}
+
+static int
+android_media_AudioSystem_setRingerMode(JNIEnv *env, jobject thiz, jint mode, jint mask)
+{
+    return check_AudioSystem_Command(AudioSystem::setRingerMode(mode, mask));
+}
+
+static int
+android_media_AudioSystem_setForceUse(JNIEnv *env, jobject thiz, jint usage, jint config)
+{
+    return check_AudioSystem_Command(AudioSystem::setForceUse(static_cast <AudioSystem::force_use>(usage),
+                                                           static_cast <AudioSystem::forced_config>(config)));
+}
+
+static int
+android_media_AudioSystem_getForceUse(JNIEnv *env, jobject thiz, jint usage)
+{
+    return static_cast <int>(AudioSystem::getForceUse(static_cast <AudioSystem::force_use>(usage)));
+}
+
+static int
+android_media_AudioSystem_initStreamVolume(JNIEnv *env, jobject thiz, jint stream, jint indexMin, jint indexMax)
+{
+    return check_AudioSystem_Command(AudioSystem::initStreamVolume(static_cast <AudioSystem::stream_type>(stream),
+                                                                   indexMin,
+                                                                   indexMax));
+}
+
+static int
+android_media_AudioSystem_setStreamVolumeIndex(JNIEnv *env, jobject thiz, jint stream, jint index)
+{
+    return check_AudioSystem_Command(AudioSystem::setStreamVolumeIndex(static_cast <AudioSystem::stream_type>(stream), index));
+}
+
+static int
+android_media_AudioSystem_getStreamVolumeIndex(JNIEnv *env, jobject thiz, jint stream)
+{
+    int index;
+    if (AudioSystem::getStreamVolumeIndex(static_cast <AudioSystem::stream_type>(stream), &index) != NO_ERROR) {
+        index = -1;
+    }
+    return index;
+}
+
 // ----------------------------------------------------------------------------
 
 static JNINativeMethod gMethods[] = {
-    {"setVolume",           "(II)I",    (void *)android_media_AudioSystem_setVolume},
-    {"getVolume",           "(I)I",     (void *)android_media_AudioSystem_getVolume},
-    {"setParameter",        "(Ljava/lang/String;Ljava/lang/String;)V", (void *)android_media_AudioSystem_setParameter},
+    {"setParameters",        "(Ljava/lang/String;)I", (void *)android_media_AudioSystem_setParameters},
+    {"getParameters",        "(Ljava/lang/String;)Ljava/lang/String;", (void *)android_media_AudioSystem_getParameters},
     {"muteMicrophone",      "(Z)I",     (void *)android_media_AudioSystem_muteMicrophone},
     {"isMicrophoneMuted",   "()Z",      (void *)android_media_AudioSystem_isMicrophoneMuted},
-    {"setRouting",          "(III)I",   (void *)android_media_AudioSystem_setRouting},
-    {"getRouting",          "(I)I",     (void *)android_media_AudioSystem_getRouting},
-    {"setMode",             "(I)I",     (void *)android_media_AudioSystem_setMode},
-    {"getMode",             "()I",      (void *)android_media_AudioSystem_getMode},
     {"isMusicActive",       "()Z",      (void *)android_media_AudioSystem_isMusicActive},
+    {"setDeviceConnectionState", "(IILjava/lang/String;)I", (void *)android_media_AudioSystem_setDeviceConnectionState},
+    {"getDeviceConnectionState", "(ILjava/lang/String;)I",  (void *)android_media_AudioSystem_getDeviceConnectionState},
+    {"setPhoneState",       "(I)I",     (void *)android_media_AudioSystem_setPhoneState},
+    {"setRingerMode",       "(II)I",    (void *)android_media_AudioSystem_setRingerMode},
+    {"setForceUse",         "(II)I",    (void *)android_media_AudioSystem_setForceUse},
+    {"getForceUse",         "(I)I",     (void *)android_media_AudioSystem_getForceUse},
+    {"initStreamVolume",    "(III)I",   (void *)android_media_AudioSystem_initStreamVolume},
+    {"setStreamVolumeIndex","(II)I",    (void *)android_media_AudioSystem_setStreamVolumeIndex},
+    {"getStreamVolumeIndex","(I)I",     (void *)android_media_AudioSystem_getStreamVolumeIndex}
 };
 
 const char* const kClassPathName = "android/media/AudioSystem";
diff --git a/core/jni/android_media_AudioTrack.cpp b/core/jni/android_media_AudioTrack.cpp
index bc7f3f5..6449147 100644
--- a/core/jni/android_media_AudioTrack.cpp
+++ b/core/jni/android_media_AudioTrack.cpp
@@ -103,7 +103,7 @@
 #define AUDIOTRACK_ERROR_BAD_VALUE                 -2
 #define AUDIOTRACK_ERROR_INVALID_OPERATION         -3
 #define AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM         -16
-#define AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELCOUNT -17
+#define AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK -17
 #define AUDIOTRACK_ERROR_SETUP_INVALIDFORMAT       -18
 #define AUDIOTRACK_ERROR_SETUP_INVALIDSTREAMTYPE   -19
 #define AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED    -20
@@ -164,11 +164,11 @@
 // ----------------------------------------------------------------------------
 static int
 android_media_AudioTrack_native_setup(JNIEnv *env, jobject thiz, jobject weak_this,
-        jint streamType, jint sampleRateInHertz, jint nbChannels, 
+        jint streamType, jint sampleRateInHertz, jint channels,
         jint audioFormat, jint buffSizeInBytes, jint memoryMode)
 {
-    LOGV("sampleRate=%d, audioFormat(from Java)=%d, nbChannels=%d, buffSize=%d", 
-        sampleRateInHertz, audioFormat, nbChannels, buffSizeInBytes);
+    LOGV("sampleRate=%d, audioFormat(from Java)=%d, channels=%x, buffSize=%d",
+        sampleRateInHertz, audioFormat, channels, buffSizeInBytes);
     int afSampleRate;
     int afFrameCount;
 
@@ -181,10 +181,11 @@
         return AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM;
     }
 
-    if ((nbChannels == 0) || (nbChannels > 2)) {
-        LOGE("Error creating AudioTrack: channel count is not 1 or 2.");
-        return AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELCOUNT;
+    if (!AudioSystem::isOutputChannel(channels)) {
+        LOGE("Error creating AudioTrack: invalid channel mask.");
+        return AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK;
     }
+    int nbChannels = AudioSystem::popCount(channels);
     
     // check the stream type
     AudioSystem::stream_type atStreamType;
@@ -231,15 +232,7 @@
     int bytesPerSample = audioFormat == javaAudioTrackFields.PCM16 ? 2 : 1;
     int format = audioFormat == javaAudioTrackFields.PCM16 ? 
             AudioSystem::PCM_16_BIT : AudioSystem::PCM_8_BIT;
-    int frameCount;
-    if (buffSizeInBytes == -1) {
-        // compute the frame count based on the system's output frame count 
-        // and the native sample rate
-        frameCount = (sampleRateInHertz*afFrameCount)/afSampleRate;
-    } else {
-        // compute the frame count based on the specified buffer size
-        frameCount = buffSizeInBytes / (nbChannels * bytesPerSample);
-    }
+    int frameCount = buffSizeInBytes / (nbChannels * bytesPerSample);
     
     AudioTrackJniStorage* lpJniStorage = new AudioTrackJniStorage();
     
@@ -271,7 +264,7 @@
             atStreamType,// stream type
             sampleRateInHertz,
             format,// word length, PCM
-            nbChannels,
+            channels,
             frameCount,
             0,// flags
             audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user)
@@ -291,7 +284,7 @@
             atStreamType,// stream type
             sampleRateInHertz,
             format,// word length, PCM
-            nbChannels,
+            channels,
             frameCount,
             0,// flags
             audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user));
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 83ff508..503cb31 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -39,21 +39,10 @@
 {
 public:
 
-    // input sources values must always be defined in the range
-    // [AudioRecord::DEFAULT_INPUT, AudioRecord::NUM_INPUT_SOURCES[
-    enum input_source {
-        DEFAULT_INPUT   =-1,
-        MIC_INPUT       = 0,
-        VOICE_UPLINK_INPUT = 1,
-        VOICE_DOWNLINK_INPUT = 2,
-        VOICE_CALL_INPUT = 3,
-        NUM_INPUT_SOURCES
-    };
-
     static const int DEFAULT_SAMPLE_RATE = 8000;
 
     /* Events used by AudioRecord callback function (callback_t).
-     * 
+     *
      * to keep in sync with frameworks/base/media/java/android/media/AudioRecord.java
      */
     enum event_type {
@@ -61,7 +50,7 @@
         EVENT_OVERRUN = 1,          // PCM buffer overrun occured.
         EVENT_MARKER = 2,           // Record head is at the specified marker position
                                     // (See setMarkerPosition()).
-        EVENT_NEW_POS = 3,          // Record head is at a new position 
+        EVENT_NEW_POS = 3,          // Record head is at a new position
                                     // (See setPositionUpdatePeriod()).
     };
 
@@ -123,11 +112,11 @@
      *
      * Parameters:
      *
-     * inputSource:        Select the audio input to record to (e.g. AudioRecord::MIC_INPUT).
+     * inputSource:        Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT).
      * sampleRate:         Track sampling rate in Hz.
-     * format:             PCM sample format (e.g AudioSystem::PCM_16_BIT for signed
+     * format:             Audio format (e.g AudioSystem::PCM_16_BIT for signed
      *                     16 bits per sample).
-     * channelCount:       Number of PCM channels (e.g 2 for stereo).
+     * channels:           Channel mask: see AudioSystem::audio_channels.
      * frameCount:         Total size of track PCM buffer in frames. This defines the
      *                     latency of the track.
      * flags:              A bitmask of acoustic values from enum record_flags.  It enables
@@ -148,7 +137,7 @@
                         AudioRecord(int inputSource,
                                     uint32_t sampleRate = 0,
                                     int format          = 0,
-                                    int channelCount    = 0,
+                                    uint32_t channels = AudioSystem::CHANNEL_IN_MONO,
                                     int frameCount      = 0,
                                     uint32_t flags      = 0,
                                     callback_t cbf = 0,
@@ -166,14 +155,14 @@
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful intialization
      *  - INVALID_OPERATION: AudioRecord is already intitialized or record device is already in use
-     *  - BAD_VALUE: invalid parameter (channelCount, format, sampleRate...)
+     *  - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - PERMISSION_DENIED: recording is not allowed for the requesting process
      * */
             status_t    set(int inputSource     = 0,
                             uint32_t sampleRate = 0,
                             int format          = 0,
-                            int channelCount    = 0,
+                            uint32_t channels = AudioSystem::CHANNEL_IN_MONO,
                             int frameCount      = 0,
                             uint32_t flags      = 0,
                             callback_t cbf = 0,
@@ -199,6 +188,7 @@
 
             int         format() const;
             int         channelCount() const;
+            int         channels() const;
             uint32_t    frameCount() const;
             int         frameSize() const;
             int         inputSource() const;
@@ -222,8 +212,8 @@
 
     /* Sets marker position. When record reaches the number of frames specified,
      * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
-     * with marker == 0 cancels marker notification callback. 
-     * If the AudioRecord has been opened with no callback function associated, 
+     * with marker == 0 cancels marker notification callback.
+     * If the AudioRecord has been opened with no callback function associated,
      * the operation will fail.
      *
      * Parameters:
@@ -238,10 +228,10 @@
             status_t    getMarkerPosition(uint32_t *marker);
 
 
-    /* Sets position update period. Every time the number of frames specified has been recorded, 
-     * a callback with event type EVENT_NEW_POS is called. 
-     * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 
-     * callback. 
+    /* Sets position update period. Every time the number of frames specified has been recorded,
+     * a callback with event type EVENT_NEW_POS is called.
+     * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
+     * callback.
      * If the AudioRecord has been opened with no callback function associated,
      * the operation will fail.
      *
@@ -257,8 +247,8 @@
             status_t    getPositionUpdatePeriod(uint32_t *updatePeriod);
 
 
-    /* Gets record head position. The position is the  total number of frames 
-     * recorded since record start. 
+    /* Gets record head position. The position is the  total number of frames
+     * recorded since record start.
      *
      * Parameters:
      *
@@ -270,8 +260,16 @@
      */
             status_t    getPosition(uint32_t *position);
 
-            
-            
+    /* returns a handle on the audio input used by this AudioRecord.
+     *
+     * Parameters:
+     *  none.
+     *
+     * Returned value:
+     *  handle on audio hardware input
+     */
+            audio_io_handle_t    getInput() { return mInput; }
+
     /* obtains a buffer of "frameCount" frames. The buffer must be
      * filled entirely. If the track is stopped, obtainBuffer() returns
      * STOPPED instead of NO_ERROR as long as there are buffers availlable,
@@ -342,6 +340,7 @@
     bool                    mMarkerReached;
     uint32_t                mNewPosition;
     uint32_t                mUpdatePeriod;
+    audio_io_handle_t       mInput;
 };
 
 }; // namespace android
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 3a3a714..0ea04a4 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -24,36 +24,130 @@
 namespace android {
 
 typedef void (*audio_error_callback)(status_t err);
+typedef void * 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,
+        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,
         NUM_STREAM_TYPES
     };
 
-    enum audio_output_type {
-        AUDIO_OUTPUT_DEFAULT      =-1,
-        AUDIO_OUTPUT_HARDWARE     = 0,
-        AUDIO_OUTPUT_A2DP         = 1,
-        NUM_AUDIO_OUTPUT_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 {
-        FORMAT_DEFAULT = 0,
-        PCM_16_BIT,
-        PCM_8_BIT,
-        INVALID_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 = 0x1,
+        CHANNEL_OUT_FRONT_RIGHT = 0x2,
+        CHANNEL_OUT_FRONT_CENTER = 0x4,
+        CHANNEL_OUT_LOW_FREQUENCY = 0x8,
+        CHANNEL_OUT_BACK_LEFT = 0x10,
+        CHANNEL_OUT_BACK_RIGHT = 0x20,
+        CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40,
+        CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80,
+        CHANNEL_OUT_BACK_CENTER = 0x100,
+        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 = 0x10000,
+        CHANNEL_IN_RIGHT = 0x20000,
+        CHANNEL_IN_FRONT = 0x40000,
+        CHANNEL_IN_BACK = 0x80000,
+        CHANNEL_IN_LEFT_PROCESSED = 0x100000,
+        CHANNEL_IN_RIGHT_PROCESSED = 0x200000,
+        CHANNEL_IN_FRONT_PROCESSED = 0x400000,
+        CHANNEL_IN_BACK_PROCESSED = 0x800000,
+        CHANNEL_IN_PRESSURE = 0x1000000,
+        CHANNEL_IN_X_AXIS = 0x2000000,
+        CHANNEL_IN_Y_AXIS = 0x4000000,
+        CHANNEL_IN_Z_AXIS = 0x8000000,
+        CHANNEL_IN_VOICE_UPLINK = 0x10000000,
+        CHANNEL_IN_VOICE_DNLINK = 0x20000000,
+        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 |
+                CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)
     };
 
     enum audio_mode {
@@ -65,15 +159,6 @@
         NUM_MODES  // not a valid entry, denotes end-of-list
     };
 
-    enum audio_routes {
-        ROUTE_EARPIECE       = (1 << 0),
-        ROUTE_SPEAKER        = (1 << 1),
-        ROUTE_BLUETOOTH_SCO  = (1 << 2),
-        ROUTE_HEADSET        = (1 << 3),
-        ROUTE_BLUETOOTH_A2DP = (1 << 4),
-        ROUTE_ALL            = -1UL,
-    };
-
     enum audio_in_acoustics {
         AGC_ENABLE    = 0x0001,
         AGC_DISABLE   = 0,
@@ -87,36 +172,37 @@
      * only privileged processes can have access to them
      */
 
-    // routing helper functions
-    static status_t speakerphone(bool state);
-    static status_t isSpeakerphoneOn(bool* state);
-    static status_t bluetoothSco(bool state);
-    static status_t isBluetoothScoOn(bool* state);
+    // 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 setMasterMute(bool mute);
     static status_t getMasterVolume(float* volume);
+    // mute/unmute audio outputs
+    static status_t setMasterMute(bool mute);
     static status_t getMasterMute(bool* mute);
 
-    static status_t setStreamVolume(int stream, float value);
+    // set/get stream volume on specified output
+    static status_t setStreamVolume(int stream, float value, void *output);
+    static status_t getStreamVolume(int stream, float* volume, void *output);
+
+    // mute/unmute stream
     static status_t setStreamMute(int stream, bool mute);
-    static status_t getStreamVolume(int stream, float* volume);
     static status_t getStreamMute(int stream, bool* mute);
 
+    // set audio mode in audio hardware (see AudioSystem::audio_mode)
     static status_t setMode(int mode);
-    static status_t getMode(int* mode);
 
-    static status_t setRouting(int mode, uint32_t routes, uint32_t mask);
-    static status_t getRouting(int mode, uint32_t* routes);
-
+    // returns true if tracks are active on AudioSystem::MUSIC stream
     static status_t isMusicActive(bool *state);
 
-    // Temporary interface, do not use
-    // TODO: Replace with a more generic key:value get/set mechanism
-    static status_t setParameter(const char* key, const char* value);
-    
+    // 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
@@ -130,47 +216,247 @@
     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, 
+
+    static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
         size_t* buffSize);
 
+
+    //
+    // 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,
+        DEVICE_OUT_FM_HEADPHONE = 0x800,
+        DEVICE_OUT_FM_SPEAKER = 0x1000,
+        DEVICE_OUT_TTY = 0x2000,
+        DEVICE_OUT_DEFAULT = 0x8000,
+        DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |
+                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
+                DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_FM_HEADPHONE |
+                DEVICE_OUT_FM_SPEAKER | DEVICE_OUT_TTY | DEVICE_OUT_DEFAULT),
+
+        // 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_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 |
+                DEVICE_IN_VOICE_CALL| DEVICE_IN_DEFAULT)
+    };
+
+    // 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,
+        NUM_FORCE_CONFIG,
+        FORCE_DEFAULT = FORCE_NONE
+    };
+
+    // usages used for setForceUse()
+    enum force_use {
+        FOR_COMMUNICATION,
+        FOR_MEDIA,
+        FOR_RECORD,
+        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);
+    static status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+    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 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);
+    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() {      
+        AudioFlingerClient() {
         }
-        
+
         // DeathRecipient
         virtual void binderDied(const wp<IBinder>& who);
-        
+
         // IAudioFlingerClient
-        virtual void a2dpEnabledChanged(bool enabled);
-        
+
+        // 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, void *param1, void *param2);
     };
-    static int getOutput(int streamType);
+
+    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 int gOutSamplingRate[NUM_AUDIO_OUTPUT_TYPES];
-    static int gOutFrameCount[NUM_AUDIO_OUTPUT_TYPES];
-    static uint32_t gOutLatency[NUM_AUDIO_OUTPUT_TYPES];
-    static bool gA2dpEnabled;
-    
+
     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 changeing 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
+    static const char *keyRouting;
+    static const char *keySamplingRate;
+    static const char *keyFormat;
+    static const char *keyChannels;
+    static const char *keyFrameCount;
+
+    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);
+
+    size_t size() { return mParameters.size(); }
+
+private:
+    String8 mKeyValuePairs;
+    KeyedVector <String8, String8> mParameters;
 };
 
 };  // namespace android
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 2e1fbda..981c2f6 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -117,9 +117,9 @@
      * streamType:         Select the type of audio stream this track is attached to
      *                     (e.g. AudioSystem::MUSIC).
      * sampleRate:         Track sampling rate in Hz.
-     * format:             PCM sample format (e.g AudioSystem::PCM_16_BIT for signed
+     * format:             Audio format (e.g AudioSystem::PCM_16_BIT for signed
      *                     16 bits per sample).
-     * channelCount:       Number of PCM channels (e.g 2 for stereo).
+     * channels:           Channel mask: see AudioSystem::audio_channels.
      * frameCount:         Total size of track PCM buffer in frames. This defines the
      *                     latency of the track.
      * flags:              Reserved for future use.
@@ -133,7 +133,7 @@
                         AudioTrack( int streamType,
                                     uint32_t sampleRate  = 0,
                                     int format           = 0,
-                                    int channelCount     = 0,
+                                    int channels         = 0,
                                     int frameCount       = 0,
                                     uint32_t flags       = 0,
                                     callback_t cbf       = 0,
@@ -152,7 +152,7 @@
                         AudioTrack( int streamType,
                                     uint32_t sampleRate = 0,
                                     int format          = 0,
-                                    int channelCount    = 0,
+                                    int channels        = 0,
                                     const sp<IMemory>& sharedBuffer = 0,
                                     uint32_t flags      = 0,
                                     callback_t cbf      = 0,
@@ -169,13 +169,13 @@
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful intialization
      *  - INVALID_OPERATION: AudioTrack is already intitialized
-     *  - BAD_VALUE: invalid parameter (channelCount, format, sampleRate...)
+     *  - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
      * */
             status_t    set(int streamType      =-1,
                             uint32_t sampleRate = 0,
                             int format          = 0,
-                            int channelCount    = 0,
+                            int channels        = 0,
                             int frameCount      = 0,
                             uint32_t flags      = 0,
                             callback_t cbf      = 0,
@@ -330,6 +330,16 @@
      */
             status_t    reload();
 
+    /* returns a handle on the audio output used by this AudioTrack.
+     *
+     * Parameters:
+     *  none.
+     *
+     * Returned value:
+     *  handle on audio hardware output
+     */
+            audio_io_handle_t    getOutput();
+
     /* obtains a buffer of "frameCount" frames. The buffer must be
      * filled entirely. If the track is stopped, obtainBuffer() returns
      * STOPPED instead of NO_ERROR as long as there are buffers availlable,
@@ -387,7 +397,6 @@
     sp<AudioTrackThread>    mAudioTrackThread;
 
     float                   mVolume[2];
-    uint32_t                mSampleRate;
     uint32_t                mFrameCount;
 
     audio_track_cblk_t*     mCblk;
@@ -395,6 +404,7 @@
     uint8_t                 mFormat;
     uint8_t                 mChannelCount;
     uint8_t                 mMuted;
+    uint32_t                mChannels;
     status_t                mStatus;
     uint32_t                mLatency;
 
@@ -410,6 +420,7 @@
     bool                    mMarkerReached;
     uint32_t                mNewPosition;
     uint32_t                mUpdatePeriod;
+    uint32_t                mFlags;
 };
 
 
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index bac3d29..26e6972 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -27,7 +27,7 @@
 #include <media/IAudioTrack.h>
 #include <media/IAudioRecord.h>
 #include <media/IAudioFlingerClient.h>
-
+#include <utils/String8.h>
 
 namespace android {
 
@@ -50,11 +50,12 @@
                                 int frameCount,
                                 uint32_t flags,
                                 const sp<IMemory>& sharedBuffer,
+                                void *output,
                                 status_t *status) = 0;
 
     virtual sp<IAudioRecord> openRecord(
                                 pid_t pid,
-                                int inputSource,
+                                void *input,
                                 uint32_t sampleRate,
                                 int format,
                                 int channelCount,
@@ -65,11 +66,11 @@
     /* query the audio hardware state. This state never changes,
      * and therefore can be cached.
      */
-    virtual     uint32_t    sampleRate(int output) const = 0;
-    virtual     int         channelCount(int output) const = 0;
-    virtual     int         format(int output) const = 0;
-    virtual     size_t      frameCount(int output) const = 0;
-    virtual     uint32_t    latency(int output) const = 0;
+    virtual     uint32_t    sampleRate(void *output) const = 0;
+    virtual     int         channelCount(void *output) const = 0;
+    virtual     int         format(void *output) const = 0;
+    virtual     size_t      frameCount(void *output) const = 0;
+    virtual     uint32_t    latency(void *output) const = 0;
 
     /* set/get the audio hardware state. This will probably be used by
      * the preference panel, mostly.
@@ -83,19 +84,14 @@
     /* set/get stream type state. This will probably be used by
      * the preference panel, mostly.
      */
-    virtual     status_t    setStreamVolume(int stream, float value) = 0;
+    virtual     status_t    setStreamVolume(int stream, float value, void *output) = 0;
     virtual     status_t    setStreamMute(int stream, bool muted) = 0;
 
-    virtual     float       streamVolume(int stream) const = 0;
+    virtual     float       streamVolume(int stream, void *output) const = 0;
     virtual     bool        streamMute(int stream) const = 0;
 
-    // set/get audio routing
-    virtual     status_t    setRouting(int mode, uint32_t routes, uint32_t mask) = 0;
-    virtual     uint32_t    getRouting(int mode) const = 0;
-
-    // set/get audio mode
+    // set audio mode
     virtual     status_t    setMode(int mode) = 0;
-    virtual     int         getMode() const = 0;
 
     // mic mute/state
     virtual     status_t    setMicMute(bool state) = 0;
@@ -104,22 +100,34 @@
     // is a music stream active?
     virtual     bool        isMusicActive() const = 0;
 
-    // pass a generic configuration parameter to libaudio
-    // Temporary interface, do not use
-    // TODO: Replace with a more generic key:value get/set mechanism
-    virtual     status_t  setParameter(const char* key, const char* value) = 0;
+    virtual     status_t    setParameters(void *ioHandle, const String8& keyValuePairs) = 0;
+    virtual     String8     getParameters(void *ioHandle, const String8& keys) = 0;
     
     // register a current process for audio output change notifications
     virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
     
     // retrieve the audio recording buffer size
     virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
-    
-    // force AudioFlinger thread out of standby
-    virtual     void        wakeUp() = 0;
 
-    // is A2DP output enabled
-    virtual     bool        isA2dpEnabled() const = 0;
+    virtual void *openOutput(uint32_t *pDevices,
+                                    uint32_t *pSamplingRate,
+                                    uint32_t *pFormat,
+                                    uint32_t *pChannels,
+                                    uint32_t *pLatencyMs,
+                                    uint32_t flags) = 0;
+    virtual void *openDuplicateOutput(void *output1, void *output2) = 0;
+    virtual status_t closeOutput(void *output) = 0;
+    virtual status_t suspendOutput(void *output) = 0;
+    virtual status_t restoreOutput(void *output) = 0;
+
+    virtual void *openInput(uint32_t *pDevices,
+                                    uint32_t *pSamplingRate,
+                                    uint32_t *pFormat,
+                                    uint32_t *pChannels,
+                                    uint32_t acoustics) = 0;
+    virtual status_t closeInput(void *input) = 0;
+
+    virtual status_t setStreamOutput(uint32_t stream, void *output) = 0;
 };
 
 
diff --git a/include/media/IAudioFlingerClient.h b/include/media/IAudioFlingerClient.h
index 383ec0c..78142ce3 100644
--- a/include/media/IAudioFlingerClient.h
+++ b/include/media/IAudioFlingerClient.h
@@ -20,7 +20,7 @@
 
 #include <utils/RefBase.h>
 #include <binder/IInterface.h>
-
+#include <utils/KeyedVector.h>
 
 namespace android {
 
@@ -31,8 +31,8 @@
 public:
     DECLARE_META_INTERFACE(AudioFlingerClient);
 
-    // Notifies a change of audio output from/to hardware to/from A2DP.
-    virtual void a2dpEnabledChanged(bool enabled) = 0;
+    // Notifies a change of audio input/output configuration.
+    virtual void ioConfigChanged(int event, void *param1, void *param2) = 0;
 
 };
 
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
new file mode 100644
index 0000000..4804bbd
--- /dev/null
+++ b/include/media/IAudioPolicyService.h
@@ -0,0 +1,90 @@
+/*
+ * 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_IAUDIOPOLICYSERVICE_H
+#define ANDROID_IAUDIOPOLICYSERVICE_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <media/AudioSystem.h>
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class IAudioPolicyService : public IInterface
+{
+public:
+    DECLARE_META_INTERFACE(AudioPolicyService);
+
+    //
+    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
+    //
+    virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
+                                              AudioSystem::device_connection_state state,
+                                              const char *device_address) = 0;
+    virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
+                                                                          const char *device_address) = 0;
+    virtual status_t setPhoneState(int state) = 0;
+    virtual status_t setRingerMode(uint32_t mode, uint32_t mask) = 0;
+    virtual status_t setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0;
+    virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0;
+    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;
+    virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream) = 0;
+    virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream) = 0;
+    virtual void releaseOutput(audio_io_handle_t output) = 0;
+    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;
+    virtual status_t startInput(audio_io_handle_t input) = 0;
+    virtual status_t stopInput(audio_io_handle_t input) = 0;
+    virtual void releaseInput(audio_io_handle_t input) = 0;
+    virtual status_t initStreamVolume(AudioSystem::stream_type stream,
+                                      int indexMin,
+                                      int indexMax) = 0;
+    virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index) = 0;
+    virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) = 0;
+};
+
+
+// ----------------------------------------------------------------------------
+
+class BnAudioPolicyService : public BnInterface<IAudioPolicyService>
+{
+public:
+    virtual status_t    onTransact( uint32_t code,
+                                    const Parcel& data,
+                                    Parcel* reply,
+                                    uint32_t flags = 0);
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_IAUDIOPOLICYSERVICE_H
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 496a739..8e2db20 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -55,17 +55,18 @@
                     uint32_t    volumeLR;
                 };
                 uint32_t    sampleRate;
+                // NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
+                // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
+                // 16 bit because data is converted to 16 bit before being stored in buffer
+                uint32_t    frameSize;
                 uint8_t     channels;
                 uint8_t     flowControlFlag; // underrun (out) or overrrun (in) indication
                 uint8_t     out;        // out equals 1 for AudioTrack and 0 for AudioRecord
-                uint8_t     forceReady; 
+                uint8_t     forceReady;
                 uint16_t    bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
                 uint16_t    waitTimeMs;      // Cumulated wait time
-                // Padding ensuring that data buffer starts on a cache line boundary (32 bytes). 
-                // See AudioFlinger::TrackBase constructor
-                int32_t     Padding[1];
-                // Cache line boundary
-                
+                // Cache line boundary (32 bytes)
+
                             audio_track_cblk_t();
                 uint32_t    stepUser(uint32_t frameCount);
                 bool        stepServer(uint32_t frameCount);
diff --git a/include/tts/TtsEngine.h b/include/tts/TtsEngine.h
index ed084ca..28b0d2f 100644
--- a/include/tts/TtsEngine.h
+++ b/include/tts/TtsEngine.h
@@ -43,7 +43,7 @@
 // @param [inout] void *&       - The userdata pointer set in the original
 //                                 synth call
 // @param [in]    uint32_t      - Track sampling rate in Hz
-// @param [in]    audio_format  - The AudioSystem::audio_format enum
+// @param [in]    uint32_t      - The audio format
 // @param [in]    int           - The number of channels
 // @param [inout] int8_t *&     - A buffer of audio data only valid during the
 //                                execution of the callback
@@ -54,7 +54,7 @@
 //         TTS_CALLBACK_CONTINUE to indicate the synthesis must continue if
 //            there is more data to produce.
 typedef tts_callback_status (synthDoneCB_t)(void *&, uint32_t,
-        AudioSystem::audio_format, int, int8_t *&, size_t&, tts_synth_status);
+        uint32_t, int, int8_t *&, size_t&, tts_synth_status);
 
 class TtsEngine;
 extern "C" TtsEngine* getTtsEngine();
diff --git a/libs/audioflinger/A2dpAudioInterface.cpp b/libs/audioflinger/A2dpAudioInterface.cpp
index 16a4f2d..6f9e934 100644
--- a/libs/audioflinger/A2dpAudioInterface.cpp
+++ b/libs/audioflinger/A2dpAudioInterface.cpp
@@ -29,25 +29,41 @@
 
 // ----------------------------------------------------------------------------
 
-A2dpAudioInterface::A2dpAudioInterface() :
-    mOutput(0)
+//AudioHardwareInterface* A2dpAudioInterface::createA2dpInterface()
+//{
+//    AudioHardwareInterface* hw = 0;
+//
+//    hw = AudioHardwareInterface::create();
+//    LOGD("new A2dpAudioInterface(hw: %p)", hw);
+//    hw = new A2dpAudioInterface(hw);
+//    return hw;
+//}
+
+A2dpAudioInterface::A2dpAudioInterface(AudioHardwareInterface* hw) :
+    mOutput(0), mHardwareInterface(hw), mBluetoothEnabled(true)
 {
 }
 
 A2dpAudioInterface::~A2dpAudioInterface()
 {
-    delete mOutput;
+    closeOutputStream((AudioStreamOut *)mOutput);
+    delete mHardwareInterface;
 }
 
 status_t A2dpAudioInterface::initCheck()
 {
-    return 0;
+    if (mHardwareInterface == 0) return NO_INIT;
+    return mHardwareInterface->initCheck();
 }
 
 AudioStreamOut* A2dpAudioInterface::openOutputStream(
-        int format, int channelCount, uint32_t sampleRate, status_t *status)
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
 {
-    LOGD("A2dpAudioInterface::openOutputStream %d, %d, %d\n", format, channelCount, sampleRate);
+    if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices)) {
+        LOGV("A2dpAudioInterface::openOutputStream() open HW device: %x", devices);
+        return mHardwareInterface->openOutputStream(devices, format, channels, sampleRate, status);
+    }
+
     status_t err = 0;
 
     // only one output stream allowed
@@ -59,71 +75,127 @@
 
     // create new output stream
     A2dpAudioStreamOut* out = new A2dpAudioStreamOut();
-    if ((err = out->set(format, channelCount, sampleRate)) == NO_ERROR) {
+    if ((err = out->set(devices, format, channels, sampleRate)) == NO_ERROR) {
         mOutput = out;
+        mOutput->setBluetoothEnabled(mBluetoothEnabled);
     } else {
         delete out;
     }
-    
+
     if (status)
         *status = err;
     return mOutput;
 }
 
+void A2dpAudioInterface::closeOutputStream(AudioStreamOut* out) {
+    if (mOutput == 0 || mOutput != out) {
+        LOGW("Attempt to close invalid output stream");
+    }
+    else {
+        delete mOutput;
+        mOutput = 0;
+    }
+}
+
+
 AudioStreamIn* A2dpAudioInterface::openInputStream(
-        int inputSource, int format, int channelCount, uint32_t sampleRate,
-        status_t *status, AudioSystem::audio_in_acoustics acoustics)
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
+        AudioSystem::audio_in_acoustics acoustics)
 {
-    if (status)
-        *status = -1;
-    return NULL;
+    return mHardwareInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
+}
+
+void A2dpAudioInterface::closeInputStream(AudioStreamIn* in)
+{
+    return mHardwareInterface->closeInputStream(in);
+}
+
+status_t A2dpAudioInterface::setMode(int mode)
+{
+    return mHardwareInterface->setMode(mode);
 }
 
 status_t A2dpAudioInterface::setMicMute(bool state)
 {
-    return 0;
+    return mHardwareInterface->setMicMute(state);
 }
 
 status_t A2dpAudioInterface::getMicMute(bool* state)
 {
-    return 0;
+    return mHardwareInterface->getMicMute(state);
 }
 
-status_t A2dpAudioInterface::setParameter(const char *key, const char *value)
+status_t A2dpAudioInterface::setParameters(const String8& keyValuePairs)
 {
-    LOGD("setParameter %s,%s\n", key, value);
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 value;
+    String8 key;
+    status_t status = NO_ERROR;
 
-    if (!key || !value)
-        return -EINVAL;
+    LOGV("setParameters() %s", keyValuePairs.string());
 
-    if (strcmp(key, "a2dp_sink_address") == 0) {
-        return mOutput->setAddress(value);
-    }
-    if (strcmp(key, "bluetooth_enabled") == 0) {
-        mOutput->setBluetoothEnabled(strcmp(value, "true") == 0);
+    key = "bluetooth_enabled";
+    if (param.get(key, value) == NO_ERROR) {
+        mBluetoothEnabled = (value == "true");
+        if (mOutput) {
+            mOutput->setBluetoothEnabled(mBluetoothEnabled);
+        }
+        param.remove(key);
     }
 
-    return 0;
+    if (param.size()) {
+        status_t hwStatus = mHardwareInterface->setParameters(param.toString());
+        if (status == NO_ERROR) {
+            status = hwStatus;
+        }
+    }
+
+    return status;
+}
+
+String8 A2dpAudioInterface::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    AudioParameter a2dpParam = AudioParameter();
+    String8 value;
+    String8 key;
+
+    key = "bluetooth_enabled";
+    if (param.get(key, value) == NO_ERROR) {
+        value = mBluetoothEnabled ? "true" : "false";
+        a2dpParam.add(key, value);
+        param.remove(key);
+    }
+
+    String8 keyValuePairs  = a2dpParam.toString();
+
+    if (param.size()) {
+        keyValuePairs += ";";
+        keyValuePairs += mHardwareInterface->getParameters(param.toString());
+    }
+
+    LOGV("getParameters() %s", keyValuePairs.string());
+    return keyValuePairs;
+}
+
+size_t A2dpAudioInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+{
+    return mHardwareInterface->getInputBufferSize(sampleRate, format, channelCount);
 }
 
 status_t A2dpAudioInterface::setVoiceVolume(float v)
 {
-    return 0;
+    return mHardwareInterface->setVoiceVolume(v);
 }
 
 status_t A2dpAudioInterface::setMasterVolume(float v)
 {
-    return 0;
-}
-
-status_t A2dpAudioInterface::doRouting()
-{
-    return 0;
+    return mHardwareInterface->setMasterVolume(v);
 }
 
 status_t A2dpAudioInterface::dump(int fd, const Vector<String16>& args)
 {
-    return 0;
+    return mHardwareInterface->dumpState(fd, args);
 }
 
 // ----------------------------------------------------------------------------
@@ -132,7 +204,7 @@
     mFd(-1), mStandby(true), mStartCount(0), mRetryCount(0), mData(NULL),
     // assume BT enabled to start, this is safe because its only the
     // enabled->disabled transition we are worried about
-    mBluetoothEnabled(true)
+    mBluetoothEnabled(true), mDevice(0)
 {
     // use any address by default
     strcpy(mA2dpAddress, "00:00:00:00:00:00");
@@ -140,27 +212,43 @@
 }
 
 status_t A2dpAudioInterface::A2dpAudioStreamOut::set(
-        int format, int channels, uint32_t rate)
+        uint32_t device, int *pFormat, uint32_t *pChannels, uint32_t *pRate)
 {
-    LOGD("A2dpAudioStreamOut::set %d, %d, %d\n", format, channels, rate);
+    int lFormat = pFormat ? *pFormat : 0;
+    uint32_t lChannels = pChannels ? *pChannels : 0;
+    uint32_t lRate = pRate ? *pRate : 0;
+
+    LOGD("A2dpAudioStreamOut::set %x, %d, %d, %d\n", device, lFormat, lChannels, lRate);
 
     // fix up defaults
-    if (format == 0) format = AudioSystem::PCM_16_BIT;
-    if (channels == 0) channels = channelCount();
-    if (rate == 0) rate = sampleRate();
+    if (lFormat == 0) lFormat = format();
+    if (lChannels == 0) lChannels = channels();
+    if (lRate == 0) lRate = sampleRate();
 
     // check values
-    if ((format != AudioSystem::PCM_16_BIT) ||
-            (channels != channelCount()) ||
-            (rate != sampleRate()))
+    if ((lFormat != format()) ||
+            (lChannels != channels()) ||
+            (lRate != sampleRate())){
+        if (pFormat) *pFormat = format();
+        if (pChannels) *pChannels = channels();
+        if (pRate) *pRate = sampleRate();
         return BAD_VALUE;
+    }
 
+    if (pFormat) *pFormat = lFormat;
+    if (pChannels) *pChannels = lChannels;
+    if (pRate) *pRate = lRate;
+
+    mDevice = device;
     return NO_ERROR;
 }
 
 A2dpAudioInterface::A2dpAudioStreamOut::~A2dpAudioStreamOut()
 {
+    LOGV("A2dpAudioStreamOut destructor");
+    standby();
     close();
+    LOGV("A2dpAudioStreamOut destructor returning from close()");
 }
 
 ssize_t A2dpAudioInterface::A2dpAudioStreamOut::write(const void* buffer, size_t bytes)
@@ -230,6 +318,59 @@
     return result;
 }
 
+status_t A2dpAudioInterface::A2dpAudioStreamOut::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 value;
+    String8 key = String8("a2dp_sink_address");
+    status_t status = NO_ERROR;
+    int device;
+    LOGV("A2dpAudioStreamOut::setParameters() %s", keyValuePairs.string());
+
+    if (param.get(key, value) == NO_ERROR) {
+        if (value.length() != strlen("00:00:00:00:00:00")) {
+            status = BAD_VALUE;
+        } else {
+            setAddress(value.string());
+        }
+        param.remove(key);
+    }
+    key = AudioParameter::keyRouting;
+    if (param.getInt(key, device) == NO_ERROR) {
+        if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device)) {
+            mDevice = device;
+            status = NO_ERROR;
+        } else {
+            status = BAD_VALUE;
+        }
+        param.remove(key);
+    }
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+    return status;
+}
+
+String8 A2dpAudioInterface::A2dpAudioStreamOut::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8("a2dp_sink_address");
+
+    if (param.get(key, value) == NO_ERROR) {
+        value = mA2dpAddress;
+        param.add(key, value);
+    }
+    key = AudioParameter::keyRouting;
+    if (param.get(key, value) == NO_ERROR) {
+        param.addInt(key, (int)mDevice);
+    }
+
+    LOGV("A2dpAudioStreamOut::getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
 status_t A2dpAudioInterface::A2dpAudioStreamOut::setAddress(const char* address)
 {
     Mutex::Autolock lock(mLock);
@@ -260,12 +401,14 @@
 status_t A2dpAudioInterface::A2dpAudioStreamOut::close()
 {
     Mutex::Autolock lock(mLock);
+    LOGV("A2dpAudioStreamOut::close() calling close_l()");
     return close_l();
 }
 
 status_t A2dpAudioInterface::A2dpAudioStreamOut::close_l()
 {
     if (mData) {
+        LOGV("A2dpAudioStreamOut::close_l() calling a2dp_cleanup(mData)");
         a2dp_cleanup(mData);
         mData = NULL;
     }
@@ -277,5 +420,4 @@
     return NO_ERROR;
 }
 
-
 }; // namespace android
diff --git a/libs/audioflinger/A2dpAudioInterface.h b/libs/audioflinger/A2dpAudioInterface.h
index 091e775..d6709e2 100644
--- a/libs/audioflinger/A2dpAudioInterface.h
+++ b/libs/audioflinger/A2dpAudioInterface.h
@@ -32,38 +32,44 @@
     class A2dpAudioStreamOut;
 
 public:
-                        A2dpAudioInterface();
+                        A2dpAudioInterface(AudioHardwareInterface* hw);
     virtual             ~A2dpAudioInterface();
     virtual status_t    initCheck();
 
     virtual status_t    setVoiceVolume(float volume);
     virtual status_t    setMasterVolume(float volume);
 
+    virtual status_t    setMode(int mode);
+
     // mic mute
     virtual status_t    setMicMute(bool state);
     virtual status_t    getMicMute(bool* state);
 
-    // Temporary interface, do not use
-    // TODO: Replace with a more generic key:value get/set mechanism
-    virtual status_t    setParameter(const char *key, const char *value);
+    virtual status_t    setParameters(const String8& keyValuePairs);
+    virtual String8     getParameters(const String8& keys);
+
+    virtual size_t      getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
 
     // create I/O streams
     virtual AudioStreamOut* openOutputStream(
-                                int format=0,
-                                int channelCount=0,
-                                uint32_t sampleRate=0,
+                                uint32_t devices,
+                                int *format=0,
+                                uint32_t *channels=0,
+                                uint32_t *sampleRate=0,
                                 status_t *status=0);
+    virtual    void        closeOutputStream(AudioStreamOut* out);
 
     virtual AudioStreamIn* openInputStream(
-                                int inputSource,
-                                int format,
-                                int channelCount,
-                                uint32_t sampleRate,
+                                uint32_t devices,
+                                int *format,
+                                uint32_t *channels,
+                                uint32_t *sampleRate,
                                 status_t *status,
                                 AudioSystem::audio_in_acoustics acoustics);
+    virtual    void        closeInputStream(AudioStreamIn* in);
+//    static AudioHardwareInterface* createA2dpInterface();
 
 protected:
-    virtual status_t    doRouting();
     virtual status_t    dump(int fd, const Vector<String16>& args);
 
 private:
@@ -71,19 +77,22 @@
     public:
                             A2dpAudioStreamOut();
         virtual             ~A2dpAudioStreamOut();
-                status_t    set(int format,
-                                int channelCount,
-                                uint32_t sampleRate);
+                status_t    set(uint32_t device,
+                                int *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t *pRate);
         virtual uint32_t    sampleRate() const { return 44100; }
         // SBC codec wants a multiple of 512
         virtual size_t      bufferSize() const { return 512 * 20; }
-        virtual int         channelCount() const { return 2; }
+        virtual uint32_t    channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
         virtual int         format() const { return AudioSystem::PCM_16_BIT; }
         virtual uint32_t    latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; }
-        virtual status_t    setVolume(float volume) { return INVALID_OPERATION; }
+        virtual status_t    setVolume(float left, float right) { return INVALID_OPERATION; }
         virtual ssize_t     write(const void* buffer, size_t bytes);
                 status_t    standby();
         virtual status_t    dump(int fd, const Vector<String16>& args);
+        virtual status_t    setParameters(const String8& keyValuePairs);
+        virtual String8     getParameters(const String8& keys);
 
     private:
         friend class A2dpAudioInterface;
@@ -102,11 +111,18 @@
                 void*       mData;
                 Mutex       mLock;
                 bool        mBluetoothEnabled;
+                uint32_t    mDevice;
     };
 
+    friend class A2dpAudioStreamOut;
+
     A2dpAudioStreamOut*     mOutput;
+    AudioHardwareInterface  *mHardwareInterface;
+    char        mA2dpAddress[20];
+    bool        mBluetoothEnabled;
 };
 
+
 // ----------------------------------------------------------------------------
 
 }; // namespace android
diff --git a/libs/audioflinger/Android.mk b/libs/audioflinger/Android.mk
index bb224be..7ed6a5f 100644
--- a/libs/audioflinger/Android.mk
+++ b/libs/audioflinger/Android.mk
@@ -1,13 +1,26 @@
 LOCAL_PATH:= $(call my-dir)
 
+#AUDIO_POLICY_TEST := true
+#ENABLE_AUDIO_DUMP := true
+
 include $(CLEAR_VARS)
 
+
+ifeq ($(AUDIO_POLICY_TEST),true)
+  ENABLE_AUDIO_DUMP := true
+endif
+
+
 LOCAL_SRC_FILES:= \
     AudioHardwareGeneric.cpp \
     AudioHardwareStub.cpp \
-    AudioDumpInterface.cpp \
     AudioHardwareInterface.cpp
 
+ifeq ($(ENABLE_AUDIO_DUMP),true)
+  LOCAL_SRC_FILES += AudioDumpInterface.cpp
+  LOCAL_CFLAGS += -DENABLE_AUDIO_DUMP
+endif
+
 LOCAL_SHARED_LIBRARIES := \
     libcutils \
     libutils \
@@ -21,8 +34,40 @@
 
 LOCAL_MODULE:= libaudiointerface
 
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+  LOCAL_SRC_FILES += A2dpAudioInterface.cpp
+  LOCAL_SHARED_LIBRARIES += liba2dp
+  LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP
+  LOCAL_C_INCLUDES += $(call include-path-for, bluez)
+endif
+
 include $(BUILD_STATIC_LIBRARY)
 
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:=               \
+    AudioPolicyManagerGeneric.cpp
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libutils \
+    libmedia
+
+LOCAL_MODULE:= libaudiopolicygeneric
+
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+  LOCAL_CFLAGS += -DWITH_A2DP
+endif
+
+ifeq ($(AUDIO_POLICY_TEST),true)
+  LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
+endif
+
+LOCAL_PRELINK_MODULE := false
+
+include $(BUILD_SHARED_LIBRARY)
+
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:=               \
@@ -30,7 +75,8 @@
     AudioMixer.cpp.arm          \
     AudioResampler.cpp.arm      \
     AudioResamplerSinc.cpp.arm  \
-    AudioResamplerCubic.cpp.arm
+    AudioResamplerCubic.cpp.arm \
+    AudioPolicyService.cpp
 
 LOCAL_SHARED_LIBRARIES := \
     libcutils \
@@ -41,17 +87,25 @@
 
 ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
   LOCAL_STATIC_LIBRARIES += libaudiointerface
+  LOCAL_CFLAGS += -DGENERIC_AUDIO
 else
   LOCAL_SHARED_LIBRARIES += libaudio
 endif
 
+ifeq ($(TARGET_SIMULATOR),true)
+ LOCAL_LDLIBS += -ldl
+else
+ LOCAL_SHARED_LIBRARIES += libdl
+endif
+
 LOCAL_MODULE:= libaudioflinger
 
 ifeq ($(BOARD_HAVE_BLUETOOTH),true)
-  LOCAL_SRC_FILES += A2dpAudioInterface.cpp
-  LOCAL_SHARED_LIBRARIES += liba2dp
   LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP
-  LOCAL_C_INCLUDES += $(call include-path-for, bluez)
+endif
+
+ifeq ($(AUDIO_POLICY_TEST),true)
+  LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
 endif
 
 ifeq ($(TARGET_SIMULATOR),true)
diff --git a/libs/audioflinger/AudioDumpInterface.cpp b/libs/audioflinger/AudioDumpInterface.cpp
index b4940cb..87bb014 100644
--- a/libs/audioflinger/AudioDumpInterface.cpp
+++ b/libs/audioflinger/AudioDumpInterface.cpp
@@ -16,6 +16,7 @@
 */
 
 #define LOG_TAG "AudioFlingerDump"
+//#define LOG_NDEBUG 0
 
 #include <stdint.h>
 #include <sys/types.h>
@@ -28,68 +29,209 @@
 
 namespace android {
 
-bool gFirst = true;       // true if first write after a standby
-
 // ----------------------------------------------------------------------------
 
 AudioDumpInterface::AudioDumpInterface(AudioHardwareInterface* hw)
+    : mFirstHwOutput(true), mPolicyCommands(String8("")), mFileName(String8(""))
 {
     if(hw == 0) {
         LOGE("Dump construct hw = 0");
     }
     mFinalInterface = hw;
-    mStreamOut = 0;
+    LOGV("Constructor %p, mFinalInterface %p", this, mFinalInterface);
 }
 
 
 AudioDumpInterface::~AudioDumpInterface()
 {
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        closeOutputStream((AudioStreamOut *)mOutputs[i]);
+    }
     if(mFinalInterface) delete mFinalInterface;
-    if(mStreamOut) delete mStreamOut;
 }
 
 
 AudioStreamOut* AudioDumpInterface::openOutputStream(
-        int format, int channelCount, uint32_t sampleRate, status_t *status)
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
 {
-    AudioStreamOut* outFinal = mFinalInterface->openOutputStream(format, channelCount, sampleRate, status);
+    AudioStreamOut* outFinal = NULL;
+    int lFormat = AudioSystem::PCM_16_BIT;
+    uint32_t lChannels = AudioSystem::CHANNEL_OUT_STEREO;
+    uint32_t lRate = 44100;
 
-    if(outFinal) {
-        mStreamOut =  new AudioStreamOutDump(outFinal);
-        return mStreamOut;
+
+    if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices) || mFirstHwOutput) {
+        outFinal = mFinalInterface->openOutputStream(devices, format, channels, sampleRate, status);
+        if (outFinal != 0) {
+            lFormat = outFinal->format();
+            lChannels = outFinal->channels();
+            lRate = outFinal->sampleRate();
+            if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices)) {
+                mFirstHwOutput = false;
+            }
+        }
     } else {
-        LOGE("Dump outFinal=0");
-        return 0;
+        if (format != 0 && *format != 0) lFormat = *format;
+        if (channels != 0 && *channels != 0) lChannels = *channels;
+        if (sampleRate != 0 && *sampleRate != 0) lRate = *sampleRate;
+        if (status) *status = NO_ERROR;
     }
+    LOGV("openOutputStream(), outFinal %p", outFinal);
+
+    AudioStreamOutDump *dumOutput = new AudioStreamOutDump(this, mOutputs.size(), outFinal,
+            devices, lFormat, lChannels, lRate);
+    mOutputs.add(dumOutput);
+
+    return dumOutput;
 }
 
+void AudioDumpInterface::closeOutputStream(AudioStreamOut* out)
+{
+    AudioStreamOutDump *dumpOut = (AudioStreamOutDump *)out;
+
+    if (mOutputs.indexOf(dumpOut) < 0) {
+        LOGW("Attempt to close invalid output stream");
+        return;
+    }
+    dumpOut->standby();
+    if (dumpOut->finalStream() != NULL) {
+        mFinalInterface->closeOutputStream(dumpOut->finalStream());
+    }
+
+    mOutputs.remove(dumpOut);
+    delete dumpOut;
+}
+
+AudioStreamIn* AudioDumpInterface::openInputStream(uint32_t devices, int *format, uint32_t *channels,
+        uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics)
+{
+    AudioStreamIn* inFinal = NULL;
+    int lFormat = AudioSystem::PCM_16_BIT;
+    uint32_t lChannels = AudioSystem::CHANNEL_IN_MONO;
+    uint32_t lRate = 8000;
+
+
+    if (mInputs.size() == 0) {
+        inFinal = mFinalInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
+        if (inFinal == 0) return 0;
+
+        lFormat = inFinal->format();
+        lChannels = inFinal->channels();
+        lRate = inFinal->sampleRate();
+    } else {
+        if (format != 0 && *format != 0) lFormat = *format;
+        if (channels != 0 && *channels != 0) lChannels = *channels;
+        if (sampleRate != 0 && *sampleRate != 0) lRate = *sampleRate;
+        if (status) *status = NO_ERROR;
+    }
+    LOGV("openInputStream(), inFinal %p", inFinal);
+
+    AudioStreamInDump *dumInput = new AudioStreamInDump(this, mInputs.size(), inFinal,
+            devices, lFormat, lChannels, lRate);
+    mInputs.add(dumInput);
+
+    return dumInput;
+}
+void AudioDumpInterface::closeInputStream(AudioStreamIn* in)
+{
+    AudioStreamInDump *dumpIn = (AudioStreamInDump *)in;
+
+    if (mInputs.indexOf(dumpIn) < 0) {
+        LOGW("Attempt to close invalid input stream");
+        return;
+    }
+    dumpIn->standby();
+    if (dumpIn->finalStream() != NULL) {
+        mFinalInterface->closeInputStream(dumpIn->finalStream());
+    }
+
+    mInputs.remove(dumpIn);
+    delete dumpIn;
+}
+
+
+status_t AudioDumpInterface::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 value;
+    int valueInt;
+    LOGV("setParameters %s", keyValuePairs.string());
+
+    if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) {
+        mFileName = value;
+        return NO_ERROR;
+    }
+    if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) {
+        Mutex::Autolock _l(mLock);
+        param.remove(String8("test_cmd_policy"));
+        mPolicyCommands = param.toString();
+        LOGV("test_cmd_policy command %s written", mPolicyCommands.string());
+        return NO_ERROR;
+    }
+
+    if (mFinalInterface != 0 ) return mFinalInterface->setParameters(keyValuePairs);
+    return NO_ERROR;
+}
+
+String8 AudioDumpInterface::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+
+//    LOGV("getParameters %s", keys.string());
+
+    if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) {
+        return mFileName;
+    }
+    if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) {
+        Mutex::Autolock _l(mLock);
+//        LOGV("test_cmd_policy command %s read", mPolicyCommands.string());
+        return mPolicyCommands;
+    }
+
+    if (mFinalInterface != 0 ) return mFinalInterface->getParameters(keys);
+    return String8("");
+}
+
+
 // ----------------------------------------------------------------------------
 
-AudioStreamOutDump::AudioStreamOutDump( AudioStreamOut* finalStream)
+AudioStreamOutDump::AudioStreamOutDump(AudioDumpInterface *interface,
+                                        int id,
+                                        AudioStreamOut* finalStream,
+                                        uint32_t devices,
+                                        int format,
+                                        uint32_t channels,
+                                        uint32_t sampleRate)
+    : mInterface(interface), mId(id),
+      mSampleRate(sampleRate), mFormat(format), mChannels(channels), mLatency(0), mDevice(devices),
+      mBufferSize(1024), mFinalStream(finalStream), mOutFile(0), mFileCount(0)
 {
-    mFinalStream = finalStream;
-    mOutFile = 0;
+    LOGV("AudioStreamOutDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream);
 }
 
 
 AudioStreamOutDump::~AudioStreamOutDump()
 {
     Close();
-    delete mFinalStream;
 }
 
 ssize_t AudioStreamOutDump::write(const void* buffer, size_t bytes)
 {
     ssize_t ret;
 
-    ret = mFinalStream->write(buffer, bytes);
-    if(!mOutFile && gFirst) {
-        gFirst = false;
-        // check if dump file exist
-        mOutFile = fopen(FLINGER_DUMP_NAME, "r");
-        if(mOutFile) {
-            fclose(mOutFile);
-            mOutFile = fopen(FLINGER_DUMP_NAME, "ab");
+    if (mFinalStream) {
+        ret = mFinalStream->write(buffer, bytes);
+    } else {
+        usleep((bytes * 1000000) / frameSize() / sampleRate());
+        ret = bytes;
+    }
+    if(!mOutFile) {
+        if (mInterface->fileName() != "") {
+            char name[255];
+            sprintf(name, "%s_%d_%d.pcm", mInterface->fileName().string(), mId, ++mFileCount);
+            mOutFile = fopen(name, "wb");
+            LOGV("Opening dump file %s, fh %p", name, mOutFile);
         }
     }
     if (mOutFile) {
@@ -100,13 +242,65 @@
 
 status_t AudioStreamOutDump::standby()
 {
+    LOGV("AudioStreamOutDump standby(), mOutFile %p, mFinalStream %p", mOutFile, mFinalStream);
+
     Close();
-    gFirst = true;
-    return mFinalStream->standby();
+    if (mFinalStream != 0 ) return mFinalStream->standby();
+    return NO_ERROR;
 }
 
+uint32_t AudioStreamOutDump::sampleRate() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->sampleRate();
+    return mSampleRate;
+}
 
-void AudioStreamOutDump::Close(void)
+size_t AudioStreamOutDump::bufferSize() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->bufferSize();
+    return mBufferSize;
+}
+
+uint32_t AudioStreamOutDump::channels() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->channels();
+    return mChannels;
+}
+int AudioStreamOutDump::format() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->format();
+    return mFormat;
+}
+uint32_t AudioStreamOutDump::latency() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->latency();
+    return 0;
+}
+status_t AudioStreamOutDump::setVolume(float left, float right)
+{
+    if (mFinalStream != 0 ) return mFinalStream->setVolume(left, right);
+    return NO_ERROR;
+}
+status_t AudioStreamOutDump::setParameters(const String8& keyValuePairs)
+{
+    LOGV("AudioStreamOutDump::setParameters()");
+    if (mFinalStream != 0 ) return mFinalStream->setParameters(keyValuePairs);
+    return NO_ERROR;
+}
+String8 AudioStreamOutDump::getParameters(const String8& keys)
+{
+    String8 result = String8("");
+    if (mFinalStream != 0 ) result = mFinalStream->getParameters(keys);
+    return result;
+}
+
+status_t AudioStreamOutDump::dump(int fd, const Vector<String16>& args)
+{
+    if (mFinalStream != 0 ) return mFinalStream->dump(fd, args);
+    return NO_ERROR;
+}
+
+void AudioStreamOutDump::Close()
 {
     if(mOutFile) {
         fclose(mOutFile);
@@ -114,4 +308,140 @@
     }
 }
 
+// ----------------------------------------------------------------------------
+
+AudioStreamInDump::AudioStreamInDump(AudioDumpInterface *interface,
+                                        int id,
+                                        AudioStreamIn* finalStream,
+                                        uint32_t devices,
+                                        int format,
+                                        uint32_t channels,
+                                        uint32_t sampleRate)
+    : mInterface(interface), mId(id),
+      mSampleRate(sampleRate), mFormat(format), mChannels(channels), mDevice(devices),
+      mBufferSize(1024), mFinalStream(finalStream), mInFile(0)
+{
+    LOGV("AudioStreamInDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream);
+}
+
+
+AudioStreamInDump::~AudioStreamInDump()
+{
+    Close();
+}
+
+ssize_t AudioStreamInDump::read(void* buffer, ssize_t bytes)
+{
+    if (mFinalStream) {
+        return mFinalStream->read(buffer, bytes);
+    }
+
+    usleep((bytes * 1000000) / frameSize() / sampleRate());
+
+    if(!mInFile) {
+        char name[255];
+        strcpy(name, "/sdcard/music/sine440");
+        if (channels() == AudioSystem::CHANNEL_IN_MONO) {
+            strcat(name, "_mo");
+        } else {
+            strcat(name, "_st");
+        }
+        if (format() == AudioSystem::PCM_16_BIT) {
+            strcat(name, "_16b");
+        } else {
+            strcat(name, "_8b");
+        }
+        if (sampleRate() < 16000) {
+            strcat(name, "_8k");
+        } else if (sampleRate() < 32000) {
+            strcat(name, "_22k");
+        } else if (sampleRate() < 48000) {
+            strcat(name, "_44k");
+        } else {
+            strcat(name, "_48k");
+        }
+        strcat(name, ".wav");
+        mInFile = fopen(name, "rb");
+        LOGV("Opening dump file %s, fh %p", name, mInFile);
+        if (mInFile) {
+            fseek(mInFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET);
+        }
+
+    }
+    if (mInFile) {
+        ssize_t bytesRead = fread(buffer, bytes, 1, mInFile);
+        if (bytesRead != bytes) {
+            fseek(mInFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET);
+            fread((uint8_t *)buffer+bytesRead, bytes-bytesRead, 1, mInFile);
+        }
+    }
+    return bytes;
+}
+
+status_t AudioStreamInDump::standby()
+{
+    LOGV("AudioStreamInDump standby(), mInFile %p, mFinalStream %p", mInFile, mFinalStream);
+
+    Close();
+    if (mFinalStream != 0 ) return mFinalStream->standby();
+    return NO_ERROR;
+}
+
+status_t AudioStreamInDump::setGain(float gain)
+{
+    if (mFinalStream != 0 ) return mFinalStream->setGain(gain);
+    return NO_ERROR;
+}
+
+uint32_t AudioStreamInDump::sampleRate() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->sampleRate();
+    return mSampleRate;
+}
+
+size_t AudioStreamInDump::bufferSize() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->bufferSize();
+    return mBufferSize;
+}
+
+uint32_t AudioStreamInDump::channels() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->channels();
+    return mChannels;
+}
+
+int AudioStreamInDump::format() const
+{
+    if (mFinalStream != 0 ) return mFinalStream->format();
+    return mFormat;
+}
+
+status_t AudioStreamInDump::setParameters(const String8& keyValuePairs)
+{
+    LOGV("AudioStreamInDump::setParameters()");
+    if (mFinalStream != 0 ) return mFinalStream->setParameters(keyValuePairs);
+    return NO_ERROR;
+}
+
+String8 AudioStreamInDump::getParameters(const String8& keys)
+{
+    String8 result = String8("");
+    if (mFinalStream != 0 ) result = mFinalStream->getParameters(keys);
+    return result;
+}
+
+status_t AudioStreamInDump::dump(int fd, const Vector<String16>& args)
+{
+    if (mFinalStream != 0 ) return mFinalStream->dump(fd, args);
+    return NO_ERROR;
+}
+
+void AudioStreamInDump::Close()
+{
+    if(mInFile) {
+        fclose(mInFile);
+        mInFile = 0;
+    }
+}
 }; // namespace android
diff --git a/libs/audioflinger/AudioDumpInterface.h b/libs/audioflinger/AudioDumpInterface.h
index b72c94e..4de4a16 100644
--- a/libs/audioflinger/AudioDumpInterface.h
+++ b/libs/audioflinger/AudioDumpInterface.h
@@ -20,35 +20,94 @@
 
 #include <stdint.h>
 #include <sys/types.h>
+#include <utils/String8.h>
+#include <utils/SortedVector.h>
 
 #include <hardware_legacy/AudioHardwareBase.h>
 
 namespace android {
 
-#define FLINGER_DUMP_NAME "/data/FlingerOut.pcm" // name of file used for dump
+#define AUDIO_DUMP_WAVE_HDR_SIZE 44
+
+class AudioDumpInterface;
 
 class AudioStreamOutDump : public AudioStreamOut {
 public:
-                        AudioStreamOutDump( AudioStreamOut* FinalStream);
+                        AudioStreamOutDump(AudioDumpInterface *interface,
+                                            int id,
+                                            AudioStreamOut* finalStream,
+                                            uint32_t devices,
+                                            int format,
+                                            uint32_t channels,
+                                            uint32_t sampleRate);
                         ~AudioStreamOutDump();
-                        virtual ssize_t     write(const void* buffer, size_t bytes);
 
-    virtual uint32_t    sampleRate() const { return mFinalStream->sampleRate(); }
-    virtual size_t      bufferSize() const { return mFinalStream->bufferSize(); }
-    virtual int         channelCount() const { return mFinalStream->channelCount(); }
-    virtual int         format() const { return mFinalStream->format(); }
-    virtual uint32_t    latency() const { return mFinalStream->latency(); }
-    virtual status_t    setVolume(float volume)
-                            { return mFinalStream->setVolume(volume); }
+    virtual ssize_t     write(const void* buffer, size_t bytes);
+    virtual uint32_t    sampleRate() const;
+    virtual size_t      bufferSize() const;
+    virtual uint32_t    channels() const;
+    virtual int         format() const;
+    virtual uint32_t    latency() const;
+    virtual status_t    setVolume(float left, float right);
     virtual status_t    standby();
-    virtual status_t    dump(int fd, const Vector<String16>& args) { return mFinalStream->dump(fd, args); }
+    virtual status_t    setParameters(const String8& keyValuePairs);
+    virtual String8     getParameters(const String8& keys);
+    virtual status_t    dump(int fd, const Vector<String16>& args);
     void                Close(void);
+    AudioStreamOut*     finalStream() { return mFinalStream; }
+    uint32_t            device() { return mDevice; }
 
 private:
+    AudioDumpInterface *mInterface;
+    int                  mId;
+    uint32_t mSampleRate;               //
+    uint32_t mFormat;                   //
+    uint32_t mChannels;                 // output configuration
+    uint32_t mLatency;                  //
+    uint32_t mDevice;                   // current device this output is routed to
+    size_t  mBufferSize;
     AudioStreamOut      *mFinalStream;
-    FILE                *mOutFile;     // output file
+    FILE                *mOutFile;      // output file
+    int                 mFileCount;
 };
 
+class AudioStreamInDump : public AudioStreamIn {
+public:
+                        AudioStreamInDump(AudioDumpInterface *interface,
+                                            int id,
+                                            AudioStreamIn* finalStream,
+                                            uint32_t devices,
+                                            int format,
+                                            uint32_t channels,
+                                            uint32_t sampleRate);
+                        ~AudioStreamInDump();
+
+    virtual uint32_t    sampleRate() const;
+    virtual size_t      bufferSize() const;
+    virtual uint32_t    channels() const;
+    virtual int         format() const;
+
+    virtual status_t    setGain(float gain);
+    virtual ssize_t     read(void* buffer, ssize_t bytes);
+    virtual status_t    standby();
+    virtual status_t    setParameters(const String8& keyValuePairs);
+    virtual String8     getParameters(const String8& keys);
+    virtual status_t    dump(int fd, const Vector<String16>& args);
+    void                Close(void);
+    AudioStreamIn*     finalStream() { return mFinalStream; }
+    uint32_t            device() { return mDevice; }
+
+private:
+    AudioDumpInterface *mInterface;
+    int                  mId;
+    uint32_t mSampleRate;               //
+    uint32_t mFormat;                   //
+    uint32_t mChannels;                 // output configuration
+    uint32_t mDevice;                   // current device this output is routed to
+    size_t  mBufferSize;
+    AudioStreamIn      *mFinalStream;
+    FILE                *mInFile;      // output file
+};
 
 class AudioDumpInterface : public AudioHardwareBase
 {
@@ -56,10 +115,13 @@
 public:
                         AudioDumpInterface(AudioHardwareInterface* hw);
     virtual AudioStreamOut* openOutputStream(
-                                int format=0,
-                                int channelCount=0,
-                                uint32_t sampleRate=0,
+                                uint32_t devices,
+                                int *format=0,
+                                uint32_t *channels=0,
+                                uint32_t *sampleRate=0,
                                 status_t *status=0);
+    virtual    void        closeOutputStream(AudioStreamOut* out);
+
     virtual             ~AudioDumpInterface();
 
     virtual status_t    initCheck()
@@ -75,21 +137,25 @@
     virtual status_t    getMicMute(bool* state)
                             {return mFinalInterface->getMicMute(state);}
 
-    virtual status_t    setParameter(const char* key, const char* value)
-                            {return mFinalInterface->setParameter(key, value);}
+    virtual status_t    setParameters(const String8& keyValuePairs);
+    virtual String8     getParameters(const String8& keys);
 
-    virtual AudioStreamIn* openInputStream(int inputSource, int format, int channelCount,
-            uint32_t sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics)
-        { return mFinalInterface->openInputStream(inputSource, format, channelCount, sampleRate, status, acoustics); }
+    virtual AudioStreamIn* openInputStream(uint32_t devices, int *format, uint32_t *channels,
+            uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics);
+    virtual    void        closeInputStream(AudioStreamIn* in);
 
     virtual status_t    dump(int fd, const Vector<String16>& args) { return mFinalInterface->dumpState(fd, args); }
 
+            String8     fileName() const { return mFileName; }
 protected:
-    virtual status_t    doRouting() {return mFinalInterface->setRouting(mMode, mRoutes[mMode]);}
 
-    AudioHardwareInterface  *mFinalInterface;
-    AudioStreamOutDump      *mStreamOut;
-
+    AudioHardwareInterface          *mFinalInterface;
+    SortedVector<AudioStreamOutDump *>    mOutputs;
+    bool                            mFirstHwOutput;
+    SortedVector<AudioStreamInDump *>    mInputs;
+    Mutex                           mLock;
+    String8                         mPolicyCommands;
+    String8                         mFileName;
 };
 
 }; // namespace android
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index ffc0278..c05ab77 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -71,15 +71,9 @@
 static const int8_t kMaxTrackRetries = 50;
 static const int8_t kMaxTrackStartupRetries = 50;
 
-static const int kStartSleepTime = 30000;
-static const int kStopSleepTime = 30000;
-
 static const int kDumpLockRetries = 50;
 static const int kDumpLockSleep = 20000;
 
-// Maximum number of pending buffers allocated by OutputTrack::write()
-static const uint8_t kMaxOutputTrackBuffers = 5;
-
 
 #define AUDIOFLINGER_SECURITY_ENABLED 1
 
@@ -121,132 +115,32 @@
 
 AudioFlinger::AudioFlinger()
     : BnAudioFlinger(),
-        mAudioHardware(0), mA2dpAudioInterface(0), mA2dpEnabled(false), mNotifyA2dpChange(false),
-        mForcedSpeakerCount(0), mA2dpDisableCount(0), mA2dpSuppressed(false), mForcedRoute(0),
-        mRouteRestoreTime(0), mMusicMuteSaved(false)
+        mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false)
 {
     mHardwareStatus = AUDIO_HW_IDLE;
+
     mAudioHardware = AudioHardwareInterface::create();
+
     mHardwareStatus = AUDIO_HW_INIT;
     if (mAudioHardware->initCheck() == NO_ERROR) {
         // open 16-bit output stream for s/w mixer
-        mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
-        status_t status;
-        AudioStreamOut *hwOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
-        mHardwareStatus = AUDIO_HW_IDLE;
-        if (hwOutput) {
-            mHardwareMixerThread = new MixerThread(this, hwOutput, AudioSystem::AUDIO_OUTPUT_HARDWARE);
-        } else {
-            LOGE("Failed to initialize hardware output stream, status: %d", status);
-        }
-        
-#ifdef WITH_A2DP
-        // Create A2DP interface
-        mA2dpAudioInterface = new A2dpAudioInterface();
-        AudioStreamOut *a2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
-        if (a2dpOutput) {
-            mA2dpMixerThread = new MixerThread(this, a2dpOutput, AudioSystem::AUDIO_OUTPUT_A2DP);
-            if (hwOutput) {  
-                uint32_t frameCount = ((a2dpOutput->bufferSize()/a2dpOutput->frameSize()) * hwOutput->sampleRate()) / a2dpOutput->sampleRate();
-                MixerThread::OutputTrack *a2dpOutTrack = new MixerThread::OutputTrack(mA2dpMixerThread,
-                                                            hwOutput->sampleRate(),
-                                                            AudioSystem::PCM_16_BIT,
-                                                            hwOutput->channelCount(),
-                                                            frameCount);
-                mHardwareMixerThread->setOuputTrack(a2dpOutTrack);                
-            }
-        } else {
-            LOGE("Failed to initialize A2DP output stream, status: %d", status);
-        }
-#endif
- 
-        // FIXME - this should come from settings
-        setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
-        setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
-        setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
+
         setMode(AudioSystem::MODE_NORMAL);
 
         setMasterVolume(1.0f);
         setMasterMute(false);
-
-        // Start record thread
-        mAudioRecordThread = new AudioRecordThread(mAudioHardware, this);
-        if (mAudioRecordThread != 0) {
-            mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);            
-        }
-     } else {
+    } else {
         LOGE("Couldn't even initialize the stubbed audio hardware!");
     }
 }
 
 AudioFlinger::~AudioFlinger()
 {
-    if (mAudioRecordThread != 0) {
-        mAudioRecordThread->exit();
-        mAudioRecordThread.clear();        
-    }
-    mHardwareMixerThread.clear();
-    delete mAudioHardware;
-    // deleting mA2dpAudioInterface also deletes mA2dpOutput;
-#ifdef WITH_A2DP
-    mA2dpMixerThread.clear();
-    delete mA2dpAudioInterface;
-#endif
+    mRecordThreads.clear();
+    mPlaybackThreads.clear();
 }
 
 
-#ifdef WITH_A2DP
-// setA2dpEnabled_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::setA2dpEnabled_l(bool enable)
-{    
-    SortedVector < sp<MixerThread::Track> > tracks;
-    SortedVector < wp<MixerThread::Track> > activeTracks;
-    
-    LOGV_IF(enable, "set output to A2DP\n");
-    LOGV_IF(!enable, "set output to hardware audio\n");
-
-    // Transfer tracks playing on MUSIC stream from one mixer to the other
-    if (enable) {
-        mHardwareMixerThread->getTracks_l(tracks, activeTracks);
-        mA2dpMixerThread->putTracks_l(tracks, activeTracks);
-    } else {
-        mA2dpMixerThread->getTracks_l(tracks, activeTracks);
-        mHardwareMixerThread->putTracks_l(tracks, activeTracks);
-        mA2dpMixerThread->mOutput->standby();
-    }
-    mA2dpEnabled = enable;
-    mNotifyA2dpChange = true;
-    mWaitWorkCV.broadcast();
-}
-
-// checkA2dpEnabledChange_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::checkA2dpEnabledChange_l()
-{
-    if (mNotifyA2dpChange) {
-        // Notify AudioSystem of the A2DP activation/deactivation
-        size_t size = mNotificationClients.size();
-        for (size_t i = 0; i < size; i++) {
-            sp<IBinder> binder = mNotificationClients.itemAt(i).promote();
-            if (binder != NULL) {
-                LOGV("Notifying output change to client %p", binder.get());
-                sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
-                client->a2dpEnabledChanged(mA2dpEnabled);
-            }
-        }
-        mNotifyA2dpChange = false;
-    }
-}
-#endif // WITH_A2DP
-
-bool AudioFlinger::streamForcedToSpeaker(int streamType)
-{
-    // NOTE that streams listed here must not be routed to A2DP by default:
-    // AudioSystem::routedToA2dpOutput(streamType) == false
-    return (streamType == AudioSystem::RING ||
-            streamType == AudioSystem::ALARM ||
-            streamType == AudioSystem::NOTIFICATION ||
-            streamType == AudioSystem::ENFORCED_AUDIBLE);
-}
 
 status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
 {
@@ -276,10 +170,7 @@
     char buffer[SIZE];
     String8 result;
     int hardwareStatus = mHardwareStatus;
-    
-    if (hardwareStatus == AUDIO_HW_IDLE && mHardwareMixerThread->mStandby) {
-        hardwareStatus = AUDIO_HW_STANDBY;
-    }
+
     snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
     result.append(buffer);
     write(fd, result.string(), result.size());
@@ -337,13 +228,16 @@
 
         dumpClients(fd, args);
         dumpInternals(fd, args);
-        mHardwareMixerThread->dump(fd, args);
-#ifdef WITH_A2DP
-        mA2dpMixerThread->dump(fd, args);
-#endif
 
-        // dump record client
-        if (mAudioRecordThread != 0) mAudioRecordThread->dump(fd, args);
+        // dump playback threads
+        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+            mPlaybackThreads[i]->dump(fd, args);
+        }
+
+        // dump record threads
+        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+            mRecordThreads[i]->dump(fd, args);
+        }
 
         if (mAudioHardware) {
             mAudioHardware->dumpState(fd, args);
@@ -353,6 +247,7 @@
     return NO_ERROR;
 }
 
+
 // IAudioFlinger interface
 
 
@@ -365,9 +260,10 @@
         int frameCount,
         uint32_t flags,
         const sp<IMemory>& sharedBuffer,
+        void *output,
         status_t *status)
 {
-    sp<MixerThread::Track> track;
+    sp<PlaybackThread::Track> track;
     sp<TrackHandle> trackHandle;
     sp<Client> client;
     wp<Client> wclient;
@@ -381,6 +277,12 @@
 
     {
         Mutex::Autolock _l(mLock);
+        PlaybackThread *thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            LOGE("unknown output thread");
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
 
         wclient = mClients.valueFor(pid);
 
@@ -390,16 +292,8 @@
             client = new Client(this, pid);
             mClients.add(pid, client);
         }
-#ifdef WITH_A2DP
-        if (isA2dpEnabled() && AudioSystem::routedToA2dpOutput(streamType)) {
-            track = mA2dpMixerThread->createTrack_l(client, streamType, sampleRate, format,
-                    channelCount, frameCount, sharedBuffer, &lStatus);            
-        } else 
-#endif
-        {
-            track = mHardwareMixerThread->createTrack_l(client, streamType, sampleRate, format,
-                    channelCount, frameCount, sharedBuffer, &lStatus);            
-        }
+        track = thread->createTrack_l(client, streamType, sampleRate, format,
+                channelCount, frameCount, sharedBuffer, &lStatus);
     }
     if (lStatus == NO_ERROR) {
         trackHandle = new TrackHandle(track);
@@ -414,54 +308,59 @@
     return trackHandle;
 }
 
-uint32_t AudioFlinger::sampleRate(int output) const
+uint32_t AudioFlinger::sampleRate(void *output) const
 {
-#ifdef WITH_A2DP
-     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
-         return mA2dpMixerThread->sampleRate();
-     }
-#endif
-     return mHardwareMixerThread->sampleRate();
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("sampleRate() unknown thread %p", output);
+        return 0;
+    }
+    return thread->sampleRate();
 }
 
-int AudioFlinger::channelCount(int output) const
+int AudioFlinger::channelCount(void *output) const
 {
-#ifdef WITH_A2DP
-     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
-         return mA2dpMixerThread->channelCount();
-     }
-#endif
-     return mHardwareMixerThread->channelCount();
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("channelCount() unknown thread %p", output);
+        return 0;
+    }
+    return thread->channelCount();
 }
 
-int AudioFlinger::format(int output) const
+int AudioFlinger::format(void *output) const
 {
-#ifdef WITH_A2DP
-     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
-         return mA2dpMixerThread->format();
-     }
-#endif
-     return mHardwareMixerThread->format();
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("format() unknown thread %p", output);
+        return 0;
+    }
+    return thread->format();
 }
 
-size_t AudioFlinger::frameCount(int output) const
+size_t AudioFlinger::frameCount(void *output) const
 {
-#ifdef WITH_A2DP
-     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
-         return mA2dpMixerThread->frameCount();
-     }
-#endif
-     return mHardwareMixerThread->frameCount();
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("frameCount() unknown thread %p", output);
+        return 0;
+    }
+    return thread->frameCount();
 }
 
-uint32_t AudioFlinger::latency(int output) const
+uint32_t AudioFlinger::latency(void *output) const
 {
-#ifdef WITH_A2DP
-     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
-         return mA2dpMixerThread->latency();
-     }
-#endif
-     return mHardwareMixerThread->latency();
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("latency() unknown thread %p", output);
+        return 0;
+    }
+    return thread->latency();
 }
 
 status_t AudioFlinger::setMasterVolume(float value)
@@ -478,96 +377,14 @@
         value = 1.0f;
     }
     mHardwareStatus = AUDIO_HW_IDLE;
-    mHardwareMixerThread->setMasterVolume(value);
-#ifdef WITH_A2DP
-    mA2dpMixerThread->setMasterVolume(value);
-#endif
+
+    mMasterVolume = value;
+    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+       mPlaybackThreads[i]->setMasterVolume(value);
 
     return NO_ERROR;
 }
 
-status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
-{
-    status_t err = NO_ERROR;
-
-    // check calling permissions
-    if (!settingsAllowed()) {
-        return PERMISSION_DENIED;
-    }
-    if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
-        LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
-        return BAD_VALUE;
-    }
-
-#ifdef WITH_A2DP
-    LOGV("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(),
-            IPCThreadState::self()->getCallingPid());
-    if (mode == AudioSystem::MODE_NORMAL && 
-            (mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
-        AutoMutex lock(&mLock);
-
-        bool enableA2dp = false;
-        if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) {
-            enableA2dp = true;
-        }
-        if (mA2dpDisableCount > 0) {
-            mA2dpSuppressed = enableA2dp;
-        } else {
-            setA2dpEnabled_l(enableA2dp);
-        }
-        LOGV("setOutput done\n");
-    }
-    // setRouting() is always called at least for mode == AudioSystem::MODE_IN_CALL when 
-    // SCO is enabled, whatever current mode is so we can safely handle A2DP disabling only
-    // in this case to avoid doing it several times.
-    if (mode == AudioSystem::MODE_IN_CALL &&
-        (mask & AudioSystem::ROUTE_BLUETOOTH_SCO)) {
-        AutoMutex lock(&mLock);
-        handleRouteDisablesA2dp_l(routes);
-    }
-#endif
-
-    // do nothing if only A2DP routing is affected
-    mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP;
-    if (mask) {
-        AutoMutex lock(mHardwareLock);
-        mHardwareStatus = AUDIO_HW_GET_ROUTING;
-        uint32_t r;
-        err = mAudioHardware->getRouting(mode, &r);
-        if (err == NO_ERROR) {
-            r = (r & ~mask) | (routes & mask);
-            if (mode == AudioSystem::MODE_NORMAL || 
-                (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
-                mSavedRoute = r;
-                r |= mForcedRoute;
-                LOGV("setRouting mSavedRoute %08x mForcedRoute %08x\n", mSavedRoute, mForcedRoute);
-            }
-            mHardwareStatus = AUDIO_HW_SET_ROUTING;
-            err = mAudioHardware->setRouting(mode, r);
-        }
-        mHardwareStatus = AUDIO_HW_IDLE;
-    }
-    return err;
-}
-
-uint32_t AudioFlinger::getRouting(int mode) const
-{
-    uint32_t routes = 0;
-    if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
-        if (mode == AudioSystem::MODE_NORMAL || 
-            (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
-            routes = mSavedRoute;                
-        } else {
-            mHardwareStatus = AUDIO_HW_GET_ROUTING;
-            mAudioHardware->getRouting(mode, &routes);
-            mHardwareStatus = AUDIO_HW_IDLE;
-        }
-    } else {
-        LOGW("Illegal value: getRouting(%d)", mode);
-    }
-    return routes;
-}
-
 status_t AudioFlinger::setMode(int mode)
 {
     // check calling permissions
@@ -586,15 +403,6 @@
     return ret;
 }
 
-int AudioFlinger::getMode() const
-{
-    int mode = AudioSystem::MODE_INVALID;
-    mHardwareStatus = AUDIO_HW_SET_MODE;
-    mAudioHardware->getMode(&mode);
-    mHardwareStatus = AUDIO_HW_IDLE;
-    return mode;
-}
-
 status_t AudioFlinger::setMicMute(bool state)
 {
     // check calling permissions
@@ -624,37 +432,46 @@
     if (!settingsAllowed()) {
         return PERMISSION_DENIED;
     }
-    mHardwareMixerThread->setMasterMute(muted);
-#ifdef WITH_A2DP
-    mA2dpMixerThread->setMasterMute(muted);
-#endif
+
+    mMasterMute = muted;
+    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+       mPlaybackThreads[i]->setMasterMute(muted);
+
     return NO_ERROR;
 }
 
 float AudioFlinger::masterVolume() const
 {
-    return mHardwareMixerThread->masterVolume();
+    return mMasterVolume;
 }
 
 bool AudioFlinger::masterMute() const
 {
-    return mHardwareMixerThread->masterMute();
+    return mMasterMute;
 }
 
-status_t AudioFlinger::setStreamVolume(int stream, float value)
+status_t AudioFlinger::setStreamVolume(int stream, float value, void *output)
 {
     // check calling permissions
     if (!settingsAllowed()) {
         return PERMISSION_DENIED;
     }
 
-    if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
-        uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
+    if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
         return BAD_VALUE;
     }
 
+    AutoMutex lock(mLock);
+    PlaybackThread *thread = NULL;
+    if (output) {
+        thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            return BAD_VALUE;
+        }
+    }
+
     status_t ret = NO_ERROR;
-    
+
     if (stream == AudioSystem::VOICE_CALL ||
         stream == AudioSystem::BLUETOOTH_SCO) {
         float hwValue;
@@ -671,18 +488,18 @@
         mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
         ret = mAudioHardware->setVoiceVolume(hwValue);
         mHardwareStatus = AUDIO_HW_IDLE;
-        
-    }
-    
-    mHardwareMixerThread->setStreamVolume(stream, value);
-#ifdef WITH_A2DP
-    mA2dpMixerThread->setStreamVolume(stream, value);
-#endif
 
-    mHardwareMixerThread->setStreamVolume(stream, value);
-#ifdef WITH_A2DP
-    mA2dpMixerThread->setStreamVolume(stream, value);
-#endif
+    }
+
+    mStreamTypes[stream].volume = value;
+
+    if (thread == NULL) {
+        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+           mPlaybackThreads[i]->setStreamVolume(stream, value);
+
+    } else {
+        thread->setStreamVolume(stream, value);
+    }
 
     return ret;
 }
@@ -694,82 +511,116 @@
         return PERMISSION_DENIED;
     }
 
-    if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
+    if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
         uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
         return BAD_VALUE;
     }
 
-#ifdef WITH_A2DP
-    mA2dpMixerThread->setStreamMute(stream, muted);
-#endif
-    if (stream == AudioSystem::MUSIC) 
-    {
-        AutoMutex lock(&mHardwareLock);
-        if (mForcedRoute != 0)
-            mMusicMuteSaved = muted;
-        else
-            mHardwareMixerThread->setStreamMute(stream, muted);
-    } else {
-        mHardwareMixerThread->setStreamMute(stream, muted);
-    }
+    mStreamTypes[stream].mute = muted;
+    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+       mPlaybackThreads[i]->setStreamMute(stream, muted);
 
     return NO_ERROR;
 }
 
-float AudioFlinger::streamVolume(int stream) const
+float AudioFlinger::streamVolume(int stream, void *output) const
 {
-    if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+    if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
         return 0.0f;
     }
-    
-    float volume = mHardwareMixerThread->streamVolume(stream); 
+
+    AutoMutex lock(mLock);
+    float volume;
+    if (output) {
+        PlaybackThread *thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            return 0.0f;
+        }
+        volume = thread->streamVolume(stream);
+    } else {
+        volume = mStreamTypes[stream].volume;
+    }
+
     // remove correction applied by setStreamVolume()
     if (stream == AudioSystem::VOICE_CALL) {
         volume = (volume - 0.01) / 0.99 ;
     }
-    
+
     return volume;
 }
 
 bool AudioFlinger::streamMute(int stream) const
 {
-    if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+    if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) {
         return true;
     }
-    
-    if (stream == AudioSystem::MUSIC && mForcedRoute != 0) 
-    {
-        return mMusicMuteSaved;
-    }
-    return mHardwareMixerThread->streamMute(stream);
+
+    return mStreamTypes[stream].mute;
 }
 
 bool AudioFlinger::isMusicActive() const
 {
     Mutex::Autolock _l(mLock);
- #ifdef WITH_A2DP
-     if (isA2dpEnabled()) {
-         return mA2dpMixerThread->isMusicActive_l();
-     }
- #endif
-    return mHardwareMixerThread->isMusicActive_l();
+    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
+        if (mPlaybackThreads[i]->isMusicActive()) {
+            return true;
+        }
+    }
+    return false;
 }
 
-status_t AudioFlinger::setParameter(const char* key, const char* value)
+status_t AudioFlinger::setParameters(void *ioHandle, const String8& keyValuePairs)
 {
-    status_t result, result2;
-    AutoMutex lock(mHardwareLock);
-    mHardwareStatus = AUDIO_SET_PARAMETER;
-    
-    LOGV("setParameter() key %s, value %s, tid %d, calling tid %d", key, value, gettid(), IPCThreadState::self()->getCallingPid());
-    result = mAudioHardware->setParameter(key, value);
-    if (mA2dpAudioInterface) {
-        result2 = mA2dpAudioInterface->setParameter(key, value);
-        if (result2)
-            result = result2;
+    status_t result;
+
+    LOGV("setParameters(): io %p, keyvalue %s, tid %d, calling tid %d",
+            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
     }
-    mHardwareStatus = AUDIO_HW_IDLE;
-    return result;
+
+    // ioHandle == 0 means the parameters are global to the audio hardware interface
+    if (ioHandle == 0) {
+        AutoMutex lock(mHardwareLock);
+        mHardwareStatus = AUDIO_SET_PARAMETER;
+        result = mAudioHardware->setParameters(keyValuePairs);
+        mHardwareStatus = AUDIO_HW_IDLE;
+        return result;
+    }
+
+    // Check if parameters are for an output
+    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
+    if (playbackThread != NULL) {
+        return playbackThread->setParameters(keyValuePairs);
+    }
+
+    // Check if parameters are for an input
+    RecordThread *recordThread = checkRecordThread_l(ioHandle);
+    if (recordThread != NULL) {
+        return recordThread->setParameters(keyValuePairs);
+    }
+
+    return BAD_VALUE;
+}
+
+String8 AudioFlinger::getParameters(void *ioHandle, const String8& keys)
+{
+//    LOGV("getParameters() io %p, keys %s, tid %d, calling tid %d",
+//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
+
+    if (ioHandle == 0) {
+        return mAudioHardware->getParameters(keys);
+    }
+    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
+    if (playbackThread != NULL) {
+        return playbackThread->getParameters(keys);
+    }
+    RecordThread *recordThread = checkRecordThread_l(ioHandle);
+    if (recordThread != NULL) {
+        return recordThread->getParameters(keys);
+    }
+    return String8("");
 }
 
 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
@@ -779,7 +630,7 @@
 
 void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
 {
-    
+
     LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
     Mutex::Autolock _l(mLock);
 
@@ -788,12 +639,21 @@
         LOGV("Adding notification client %p", binder.get());
         binder->linkToDeath(this);
         mNotificationClients.add(binder);
-        client->a2dpEnabledChanged(isA2dpEnabled());
+    }
+
+    // the config change is always sent from playback or record threads to avoid deadlock
+    // with AudioSystem::gLock
+    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+        mPlaybackThreads[i]->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
+    }
+
+    for (size_t i = 0; i < mRecordThreads.size(); i++) {
+        mRecordThreads[i]->sendConfigEvent(AudioSystem::INPUT_OPENED);
     }
 }
 
 void AudioFlinger::binderDied(const wp<IBinder>& who) {
-    
+
     LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
     Mutex::Autolock _l(mLock);
 
@@ -808,6 +668,17 @@
     }
 }
 
+void AudioFlinger::audioConfigChanged(int event, void *param1, void *param2) {
+    Mutex::Autolock _l(mLock);
+    size_t size = mNotificationClients.size();
+    for (size_t i = 0; i < size; i++) {
+        sp<IBinder> binder = mNotificationClients.itemAt(i);
+        LOGV("audioConfigChanged() Notifying change to client %p", binder.get());
+        sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
+        client->ioConfigChanged(event, param1, param2);
+    }
+}
+
 void AudioFlinger::removeClient(pid_t pid)
 {
     LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
@@ -815,147 +686,140 @@
     mClients.removeItem(pid);
 }
 
-bool AudioFlinger::isA2dpEnabled() const
+// ----------------------------------------------------------------------------
+
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger)
+    :   Thread(false),
+        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
+        mFormat(0), mFrameSize(1), mNewParameters(String8("")), mStandby(false)
 {
-    return mA2dpEnabled;
 }
 
-void AudioFlinger::handleForcedSpeakerRoute(int command)
+AudioFlinger::ThreadBase::~ThreadBase()
 {
-    switch(command) {
-    case ACTIVE_TRACK_ADDED:
-        {
-            AutoMutex lock(mHardwareLock);
-            if (mForcedSpeakerCount++ == 0) {
-                if (mForcedRoute == 0) {
-                    mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
-                    LOGV("++mForcedSpeakerCount == 0, mMusicMuteSaved = %d, mRouteRestoreTime = %d", mMusicMuteSaved, mRouteRestoreTime);
-                    if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
-                        LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
-                        mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
-                        usleep(mHardwareMixerThread->latency()*1000);
-                        mHardwareStatus = AUDIO_HW_SET_ROUTING;
-                        mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
-                        mHardwareStatus = AUDIO_HW_IDLE;
-                        // delay track start so that audio hardware has time to siwtch routes
-                        usleep(kStartSleepTime);
-                    }
-                }
-                mForcedRoute = AudioSystem::ROUTE_SPEAKER;
-                mRouteRestoreTime = 0;
-            }
-            LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount);
-        }
-        break;
-    case ACTIVE_TRACK_REMOVED:
-        {
-            AutoMutex lock(mHardwareLock);
-            if (mForcedSpeakerCount > 0){
-                if (--mForcedSpeakerCount == 0) {
-                    mRouteRestoreTime = systemTime() + milliseconds(kStopSleepTime/1000);
-                }
-                LOGV("mForcedSpeakerCount decremented to %d", mForcedSpeakerCount);
-            } else {
-                LOGE("mForcedSpeakerCount is already zero");
-            }
-        }
-        break;
-    case CHECK_ROUTE_RESTORE_TIME:
-    case FORCE_ROUTE_RESTORE:
-        if (mRouteRestoreTime) {
-            AutoMutex lock(mHardwareLock);
-            if (mRouteRestoreTime && 
-               (systemTime() > mRouteRestoreTime || command == FORCE_ROUTE_RESTORE)) {
-                mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, mMusicMuteSaved);
-                mForcedRoute = 0;
-                if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
-                    mHardwareStatus = AUDIO_HW_SET_ROUTING;
-                    mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute);
-                    mHardwareStatus = AUDIO_HW_IDLE;
-                    LOGV("Route forced to Speaker OFF %08x", mSavedRoute);
-                }
-                mRouteRestoreTime = 0;
-            }
-        }
-        break;
-    }
 }
 
-#ifdef WITH_A2DP
-// handleRouteDisablesA2dp_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::handleRouteDisablesA2dp_l(int routes)
+void AudioFlinger::ThreadBase::exit()
 {
-   if (routes & AudioSystem::ROUTE_BLUETOOTH_SCO) {
-        if (mA2dpDisableCount++ == 0) {
-            if (mA2dpEnabled) {
-                setA2dpEnabled_l(false);
-                mA2dpSuppressed = true;
-            }
-        }
-        LOGV("mA2dpDisableCount incremented to %d", mA2dpDisableCount);
-   } else {
-        if (mA2dpDisableCount > 0) {
-            if (--mA2dpDisableCount == 0) {
-                if (mA2dpSuppressed) {
-                    setA2dpEnabled_l(true);
-                    mA2dpSuppressed = false;
-                }
-            }
-            LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount);
-        } else {
-            LOGV("mA2dpDisableCount is already zero");
-        }
+    // keep a strong ref on ourself so that we want get
+    // destroyed in the middle of requestExitAndWait()
+    sp <ThreadBase> strongMe = this;
+
+    LOGV("ThreadBase::exit");
+    {
+        AutoMutex lock(&mLock);
+        requestExit();
+        mWaitWorkCV.signal();
     }
+    requestExitAndWait();
 }
-#endif
+
+uint32_t AudioFlinger::ThreadBase::sampleRate() const
+{
+    return mSampleRate;
+}
+
+int AudioFlinger::ThreadBase::channelCount() const
+{
+    return mChannelCount;
+}
+
+int AudioFlinger::ThreadBase::format() const
+{
+    return mFormat;
+}
+
+size_t AudioFlinger::ThreadBase::frameCount() const
+{
+    return mFrameCount;
+}
+
+status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+{
+    status_t result;
+
+    Mutex::Autolock _l(mLock);
+    mNewParameters = keyValuePairs;
+
+    mWaitWorkCV.signal();
+    mParamCond.wait(mLock);
+
+    return mParamStatus;
+}
+
+void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
+{
+    Mutex::Autolock _l(mLock);
+    ConfigEvent *configEvent = new ConfigEvent();
+    configEvent->mEvent = event;
+    configEvent->mParam = param;
+    mConfigEvents.add(configEvent);
+    LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
+    mWaitWorkCV.signal();
+}
+
+void AudioFlinger::ThreadBase::processConfigEvents()
+{
+    mLock.lock();
+    while(!mConfigEvents.isEmpty()) {
+        LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
+        ConfigEvent *configEvent = mConfigEvents[0];
+        mConfigEvents.removeAt(0);
+        // release mLock because audioConfigChanged() will call
+        // Audioflinger::audioConfigChanged() which locks AudioFlinger mLock thus creating
+        // potential cross deadlock between AudioFlinger::mLock and mLock
+        mLock.unlock();
+        audioConfigChanged(configEvent->mEvent, configEvent->mParam);
+        delete configEvent;
+        mLock.lock();
+    }
+    mLock.unlock();
+}
+
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int outputType)
-    :   Thread(false),
-        mAudioFlinger(audioFlinger), mAudioMixer(0), mOutput(output), mOutputType(outputType), 
-        mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0),
-        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false),
-        mInWrite(false)
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+    :   ThreadBase(audioFlinger),
+        mOutput(output),
+        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0),
+        mInWrite(false), mMixBuffer(0), mSuspended(false), mBytesWritten(0)
 {
-    mSampleRate = output->sampleRate();
-    mChannelCount = output->channelCount();
+    readOutputParameters();
 
-    // FIXME - Current mixer implementation only supports stereo output
-    if (mChannelCount == 1) {
-        LOGE("Invalid audio hardware channel count");
+    mMasterVolume = mAudioFlinger->masterVolume();
+    mMasterMute = mAudioFlinger->masterMute();
+
+    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
+        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
     }
-
-    mFormat = output->format();
-    mFrameCount = output->bufferSize() / output->channelCount() / sizeof(int16_t);
-    mAudioMixer = new AudioMixer(mFrameCount, output->sampleRate());
-
-    // FIXME - Current mixer implementation only supports stereo output: Always
-    // Allocate a stereo buffer even if HW output is mono.
-    mMixBuffer = new int16_t[mFrameCount * 2];
-    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
+    // notify client processes that a new input has been opened
+    sendConfigEvent(AudioSystem::OUTPUT_OPENED);
 }
 
-AudioFlinger::MixerThread::~MixerThread()
+AudioFlinger::PlaybackThread::~PlaybackThread()
 {
     delete [] mMixBuffer;
-    delete mAudioMixer;
+    if (mType != DUPLICATING) {
+        mAudioFlinger->mAudioHardware->closeOutputStream(mOutput);
+    }
 }
 
-status_t AudioFlinger::MixerThread::dump(int fd, const Vector<String16>& args)
+status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
 {
     dumpInternals(fd, args);
     dumpTracks(fd, args);
     return NO_ERROR;
 }
 
-status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector<String16>& args)
+status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, "Output %d mixer thread tracks\n", mOutputType);
+    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
     result.append(buffer);
     result.append("   Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
     for (size_t i = 0; i < mTracks.size(); ++i) {
@@ -966,7 +830,7 @@
         }
     }
 
-    snprintf(buffer, SIZE, "Output %d mixer thread active tracks\n", mOutputType);
+    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
     result.append(buffer);
     result.append("   Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
     for (size_t i = 0; i < mActiveTracks.size(); ++i) {
@@ -983,15 +847,13 @@
     return NO_ERROR;
 }
 
-status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
 
-    snprintf(buffer, SIZE, "Output %d mixer thread internals\n", mOutputType);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+    snprintf(buffer, SIZE, "Output thread %p internals\n", this);
     result.append(buffer);
     snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
     result.append(buffer);
@@ -1008,275 +870,28 @@
 }
 
 // Thread virtuals
-bool AudioFlinger::MixerThread::threadLoop()
-{
-    unsigned long sleepTime = kBufferRecoveryInUsecs;
-    int16_t* curBuf = mMixBuffer;
-    Vector< sp<Track> > tracksToRemove;
-    size_t enabledTracks = 0;
-    nsecs_t standbyTime = systemTime();   
-    size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
-    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
-
-#ifdef WITH_A2DP
-    bool outputTrackActive = false;
-#endif
-
-    do {
-        enabledTracks = 0;
-        { // scope for the AudioFlinger::mLock
-        
-            Mutex::Autolock _l(mAudioFlinger->mLock);
-
-#ifdef WITH_A2DP
-            if (mOutputTrack != NULL && !mAudioFlinger->isA2dpEnabled()) {
-                if (outputTrackActive) {
-                    mAudioFlinger->mLock.unlock();
-                    mOutputTrack->stop();
-                    mAudioFlinger->mLock.lock();
-                    outputTrackActive = false;
-                }
-            }
-            mAudioFlinger->checkA2dpEnabledChange_l();
-#endif
-
-            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
-
-            // put audio hardware into standby after short delay
-            if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
-                // wait until we have something to do...
-                LOGV("Audio hardware entering standby, output %d\n", mOutputType);
-                if (!mStandby) {
-                    mOutput->standby();
-                    mStandby = true;
-                }
-                
-#ifdef WITH_A2DP
-                if (outputTrackActive) {
-                    mAudioFlinger->mLock.unlock();
-                    mOutputTrack->stop();
-                    mAudioFlinger->mLock.lock();
-                    outputTrackActive = false;
-                }
-#endif
-                if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
-                    mAudioFlinger->handleForcedSpeakerRoute(FORCE_ROUTE_RESTORE);
-                }                
-                // we're about to wait, flush the binder command buffer
-                IPCThreadState::self()->flushCommands();
-                mAudioFlinger->mWaitWorkCV.wait(mAudioFlinger->mLock);
-                LOGV("Audio hardware exiting standby, output %d\n", mOutputType);
-                
-                if (mMasterMute == false) {
-                    char value[PROPERTY_VALUE_MAX];
-                    property_get("ro.audio.silent", value, "0");
-                    if (atoi(value)) {
-                        LOGD("Silence is golden");
-                        setMasterMute(true);
-                    }                    
-                }
-                
-                standbyTime = systemTime() + kStandbyTimeInNsecs;
-                continue;
-            }
-
-            // Forced route to speaker is handled by hardware mixer thread
-            if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
-                mAudioFlinger->handleForcedSpeakerRoute(CHECK_ROUTE_RESTORE_TIME);
-            }
-
-            // find out which tracks need to be processed
-            size_t count = activeTracks.size();
-            for (size_t i=0 ; i<count ; i++) {
-                sp<Track> t = activeTracks[i].promote();
-                if (t == 0) continue;
-
-                Track* const track = t.get();
-                audio_track_cblk_t* cblk = track->cblk();
-
-                // The first time a track is added we wait
-                // for all its buffers to be filled before processing it
-                mAudioMixer->setActiveTrack(track->name());
-                if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
-                        !track->isPaused())
-                {
-                    //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
-
-                    // compute volume for this track
-                    int16_t left, right;
-                    if (track->isMuted() || mMasterMute || track->isPausing()) {
-                        left = right = 0;
-                        if (track->isPausing()) {
-                            LOGV("paused(%d)", track->name());
-                            track->setPaused();
-                        }
-                    } else {
-                        float typeVolume = mStreamTypes[track->type()].volume;
-                        float v = mMasterVolume * typeVolume;
-                        float v_clamped = v * cblk->volume[0];
-                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
-                        left = int16_t(v_clamped);
-                        v_clamped = v * cblk->volume[1];
-                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
-                        right = int16_t(v_clamped);
-                    }
-
-                    // XXX: these things DON'T need to be done each time
-                    mAudioMixer->setBufferProvider(track);
-                    mAudioMixer->enable(AudioMixer::MIXING);
-
-                    int param;
-                    if ( track->mFillingUpStatus == Track::FS_FILLED) {
-                        // no ramp for the first volume setting
-                        track->mFillingUpStatus = Track::FS_ACTIVE;
-                        if (track->mState == TrackBase::RESUMING) {
-                            track->mState = TrackBase::ACTIVE;
-                            param = AudioMixer::RAMP_VOLUME;
-                        } else {
-                            param = AudioMixer::VOLUME;
-                        }
-                    } else {
-                        param = AudioMixer::RAMP_VOLUME;
-                    }
-                    mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
-                    mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
-                    mAudioMixer->setParameter(
-                        AudioMixer::TRACK,
-                        AudioMixer::FORMAT, track->format());
-                    mAudioMixer->setParameter(
-                        AudioMixer::TRACK,
-                        AudioMixer::CHANNEL_COUNT, track->channelCount());
-                    mAudioMixer->setParameter(
-                        AudioMixer::RESAMPLE,
-                        AudioMixer::SAMPLE_RATE,
-                        int(cblk->sampleRate));
-
-                    // reset retry count
-                    track->mRetryCount = kMaxTrackRetries;
-                    enabledTracks++;
-                } else {
-                    //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
-                    if (track->isStopped()) {
-                        track->reset();
-                    }
-                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
-                        // We have consumed all the buffers of this track.
-                        // Remove it from the list of active tracks.
-                        LOGV("remove(%d) from active list", track->name());
-                        tracksToRemove.add(track);
-                    } else {
-                        // No buffers for this track. Give it a few chances to
-                        // fill a buffer, then remove it from active list.
-                        if (--(track->mRetryCount) <= 0) {
-                            LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
-                            tracksToRemove.add(track);
-                        }
-                    }
-                    // LOGV("disable(%d)", track->name());
-                    mAudioMixer->disable(AudioMixer::MIXING);
-                }
-            }
-
-            // remove all the tracks that need to be...
-            count = tracksToRemove.size();
-            if (UNLIKELY(count)) {
-                for (size_t i=0 ; i<count ; i++) {
-                    const sp<Track>& track = tracksToRemove[i];
-                    removeActiveTrack_l(track);
-                    if (track->isTerminated()) {
-                        mTracks.remove(track);
-                        deleteTrackName_l(track->mName);
-                    }
-                }
-            }
-       }
-        
-        if (LIKELY(enabledTracks)) {
-            // mix buffers...
-            mAudioMixer->process(curBuf);
-
-#ifdef WITH_A2DP
-            if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
-                if (!outputTrackActive) {
-                    LOGV("starting output track in mixer for output %d", mOutputType);
-                    mOutputTrack->start();
-                    outputTrackActive = true;
-                }
-                mOutputTrack->write(curBuf, mFrameCount);
-            }
-#endif
-
-            // output audio to hardware
-            mLastWriteTime = systemTime();
-            mInWrite = true;
-            mOutput->write(curBuf, mixBufferSize);
-            mNumWrites++;
-            mInWrite = false;
-            mStandby = false;
-            nsecs_t temp = systemTime();
-            standbyTime = temp + kStandbyTimeInNsecs;
-            nsecs_t delta = temp - mLastWriteTime;
-            if (delta > maxPeriod) {
-                LOGW("write blocked for %llu msecs", ns2ms(delta));
-                mNumDelayedWrites++;
-            }
-            sleepTime = kBufferRecoveryInUsecs;
-        } else {         
-#ifdef WITH_A2DP
-            if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
-                if (outputTrackActive) {
-                    mOutputTrack->write(curBuf, 0);
-                    if (mOutputTrack->bufferQueueEmpty()) {
-                        mOutputTrack->stop();
-                        outputTrackActive = false;
-                    } else {
-                        standbyTime = systemTime() + kStandbyTimeInNsecs;
-                    }
-                }
-            }
-#endif
-            // There was nothing to mix this round, which means all
-            // active tracks were late. Sleep a little bit to give
-            // them another chance. If we're too late, the audio
-            // hardware will zero-fill for us.
-            //LOGV("no buffers - usleep(%lu)", sleepTime);
-            usleep(sleepTime);
-            if (sleepTime < kMaxBufferRecoveryInUsecs) {
-                sleepTime += kBufferRecoveryInUsecs;
-            }
-        }
-
-        // finally let go of all our tracks, without the lock held
-        // since we can't guarantee the destructors won't acquire that
-        // same lock.
-        tracksToRemove.clear();
-    } while (true);
-
-    return false;
-}
-
-status_t AudioFlinger::MixerThread::readyToRun()
+status_t AudioFlinger::PlaybackThread::readyToRun()
 {
     if (mSampleRate == 0) {
         LOGE("No working audio driver found.");
         return NO_INIT;
     }
-    LOGI("AudioFlinger's thread ready to run for output %d", mOutputType);
+    LOGI("AudioFlinger's thread %p ready to run", this);
     return NO_ERROR;
 }
 
-void AudioFlinger::MixerThread::onFirstRef()
+void AudioFlinger::PlaybackThread::onFirstRef()
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
 
-    snprintf(buffer, SIZE, "Mixer Thread for output %d", mOutputType);
+    snprintf(buffer, SIZE, "Playback Thread %p", this);
 
     run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
 }
 
-// MixerThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::MixerThread::Track>  AudioFlinger::MixerThread::createTrack_l(
+// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
         const sp<AudioFlinger::Client>& client,
         int streamType,
         uint32_t sampleRate,
@@ -1288,28 +903,39 @@
 {
     sp<Track> track;
     status_t lStatus;
-    
-    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
-    if (sampleRate > mSampleRate*2) {
-        LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
-        lStatus = BAD_VALUE;
-        goto Exit;
+
+    if (mType == DIRECT) {
+        if (sampleRate != mSampleRate || format != mFormat || channelCount != mChannelCount) {
+            LOGE("createTrack_l() Bad parameter:  sampleRate %d format %d, channelCount %d for output %p",
+                 sampleRate, format, channelCount, mOutput);
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+    } else {
+        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+        if (sampleRate > mSampleRate*2) {
+            LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
     }
 
-
-    if (mSampleRate == 0) {
+    if (mOutput == 0) {
         LOGE("Audio driver not initialized.");
         lStatus = NO_INIT;
         goto Exit;
     }
 
-    track = new Track(this, client, streamType, sampleRate, format,
-            channelCount, frameCount, sharedBuffer);
-    if (track->getCblk() == NULL) {
-        lStatus = NO_MEMORY;
-        goto Exit;
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+        track = new Track(this, client, streamType, sampleRate, format,
+                channelCount, frameCount, sharedBuffer);
+        if (track->getCblk() == NULL) {
+            lStatus = NO_MEMORY;
+            goto Exit;
+        }
+        mTracks.add(track);
     }
-    mTracks.add(track);
     lStatus = NO_ERROR;
 
 Exit:
@@ -1319,87 +945,7 @@
     return track;
 }
 
-// getTracks_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::getTracks_l(
-        SortedVector < sp<Track> >& tracks,
-        SortedVector < wp<Track> >& activeTracks)
-{
-    size_t size = mTracks.size();
-    LOGV ("MixerThread::getTracks_l() for output %d, mTracks.size %d, mActiveTracks.size %d", mOutputType,  mTracks.size(), mActiveTracks.size());
-    for (size_t i = 0; i < size; i++) {
-        sp<Track> t = mTracks[i];
-        if (AudioSystem::routedToA2dpOutput(t->mStreamType)) {
-            tracks.add(t);
-            int j = mActiveTracks.indexOf(t);
-            if (j >= 0) {
-                t = mActiveTracks[j].promote();
-                if (t != NULL) {
-                    activeTracks.add(t);                                    
-                }                            
-            }
-        }
-    }
-
-    size = activeTracks.size();
-    for (size_t i = 0; i < size; i++) {
-        removeActiveTrack_l(activeTracks[i]);
-    }
-    
-    size = tracks.size();
-    for (size_t i = 0; i < size; i++) {
-        sp<Track> t = tracks[i];
-        mTracks.remove(t);
-        deleteTrackName_l(t->name());
-    }
-}
-
-// putTracks_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::putTracks_l(
-        SortedVector < sp<Track> >& tracks,
-        SortedVector < wp<Track> >& activeTracks)
-{
-
-    LOGV ("MixerThread::putTracks_l() for output %d, tracks.size %d, activeTracks.size %d", mOutputType,  tracks.size(), activeTracks.size());
-
-    size_t size = tracks.size();
-    for (size_t i = 0; i < size ; i++) {
-        sp<Track> t = tracks[i];
-        int name = getTrackName_l();
-
-        if (name < 0) return;
-        
-        t->mName = name;
-        t->mMixerThread = this;
-        mTracks.add(t);
-
-        int j = activeTracks.indexOf(t);
-        if (j >= 0) {
-            addActiveTrack_l(t);
-        }            
-    }
-}
-
-uint32_t AudioFlinger::MixerThread::sampleRate() const
-{
-    return mSampleRate;
-}
-
-int AudioFlinger::MixerThread::channelCount() const
-{
-    return mChannelCount;
-}
-
-int AudioFlinger::MixerThread::format() const
-{
-    return mFormat;
-}
-
-size_t AudioFlinger::MixerThread::frameCount() const
-{
-    return mFrameCount;
-}
-
-uint32_t AudioFlinger::MixerThread::latency() const
+uint32_t AudioFlinger::PlaybackThread::latency() const
 {
     if (mOutput) {
         return mOutput->latency();
@@ -1409,53 +955,53 @@
     }
 }
 
-status_t AudioFlinger::MixerThread::setMasterVolume(float value)
+status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
 {
     mMasterVolume = value;
     return NO_ERROR;
 }
 
-status_t AudioFlinger::MixerThread::setMasterMute(bool muted)
+status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
 {
     mMasterMute = muted;
     return NO_ERROR;
 }
 
-float AudioFlinger::MixerThread::masterVolume() const
+float AudioFlinger::PlaybackThread::masterVolume() const
 {
     return mMasterVolume;
 }
 
-bool AudioFlinger::MixerThread::masterMute() const
+bool AudioFlinger::PlaybackThread::masterMute() const
 {
     return mMasterMute;
 }
 
-status_t AudioFlinger::MixerThread::setStreamVolume(int stream, float value)
+status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
 {
     mStreamTypes[stream].volume = value;
     return NO_ERROR;
 }
 
-status_t AudioFlinger::MixerThread::setStreamMute(int stream, bool muted)
+status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
 {
     mStreamTypes[stream].mute = muted;
     return NO_ERROR;
 }
 
-float AudioFlinger::MixerThread::streamVolume(int stream) const
+float AudioFlinger::PlaybackThread::streamVolume(int stream) const
 {
     return mStreamTypes[stream].volume;
 }
 
-bool AudioFlinger::MixerThread::streamMute(int stream) const
+bool AudioFlinger::PlaybackThread::streamMute(int stream) const
 {
     return mStreamTypes[stream].mute;
 }
 
-// isMusicActive_l() must be called with AudioFlinger::mLock held
-bool AudioFlinger::MixerThread::isMusicActive_l() const
+bool AudioFlinger::PlaybackThread::isMusicActive() const
 {
+    Mutex::Autolock _l(mLock);
     size_t count = mActiveTracks.size();
     for (size_t i = 0 ; i < count ; ++i) {
         sp<Track> t = mActiveTracks[i].promote();
@@ -1467,8 +1013,8 @@
     return false;
 }
 
-// addTrack_l() must be called with AudioFlinger::mLock held
-status_t AudioFlinger::MixerThread::addTrack_l(const sp<Track>& track)
+// addTrack_l() must be called with ThreadBase::mLock held
+status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
 {
     status_t status = ALREADY_EXISTS;
 
@@ -1489,18 +1035,18 @@
         // effectively get the latency it requested.
         track->mFillingUpStatus = Track::FS_FILLING;
         track->mResetDone = false;
-        addActiveTrack_l(track);
+        mActiveTracks.add(track);
         status = NO_ERROR;
     }
-    
+
     LOGV("mWaitWorkCV.broadcast");
-    mAudioFlinger->mWaitWorkCV.broadcast();
+    mWaitWorkCV.broadcast();
 
     return status;
 }
 
-// destroyTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::destroyTrack_l(const sp<Track>& track)
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
 {
     track->mState = TrackBase::TERMINATED;
     if (mActiveTracks.indexOf(track) < 0) {
@@ -1510,62 +1056,893 @@
     }
 }
 
-// addActiveTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::addActiveTrack_l(const wp<Track>& t)
+String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
 {
-    mActiveTracks.add(t);
+    return mOutput->getParameters(keys);
+}
 
-    // Force routing to speaker for certain stream types
-    // The forced routing to speaker is managed by hardware mixer
-    if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
-        sp<Track> track = t.promote();
-        if (track == NULL) return;
-   
-        if (streamForcedToSpeaker(track->type())) {
-            mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_ADDED);
-        }        
+void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
+    AudioSystem::OutputDescriptor desc;
+    void *param2 = 0;
+
+    LOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, param);
+
+    switch (event) {
+    case AudioSystem::OUTPUT_OPENED:
+    case AudioSystem::OUTPUT_CONFIG_CHANGED:
+        desc.channels = mChannelCount;
+        desc.samplingRate = mSampleRate;
+        desc.format = mFormat;
+        desc.frameCount = mFrameCount;
+        desc.latency = latency();
+        param2 = &desc;
+        break;
+
+    case AudioSystem::STREAM_CONFIG_CHANGED:
+        param2 = &param;
+    case AudioSystem::OUTPUT_CLOSED:
+    default:
+        break;
+    }
+    mAudioFlinger->audioConfigChanged(event, this, param2);
+}
+
+void AudioFlinger::PlaybackThread::readOutputParameters()
+{
+    mSampleRate = mOutput->sampleRate();
+    mChannelCount = AudioSystem::popCount(mOutput->channels());
+
+    mFormat = mOutput->format();
+    mFrameSize = mOutput->frameSize();
+    mFrameCount = mOutput->bufferSize() / mFrameSize;
+
+    mMinBytesToWrite = (mOutput->latency() * mSampleRate * mFrameSize) / 1000;
+    // FIXME - Current mixer implementation only supports stereo output: Always
+    // Allocate a stereo buffer even if HW output is mono.
+    if (mMixBuffer != NULL) delete mMixBuffer;
+    mMixBuffer = new int16_t[mFrameCount * 2];
+    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+    :   PlaybackThread(audioFlinger, output),
+        mAudioMixer(0)
+{
+    mType = PlaybackThread::MIXER;
+    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+
+    // FIXME - Current mixer implementation only supports stereo output
+    if (mChannelCount == 1) {
+        LOGE("Invalid audio hardware channel count");
     }
 }
 
-// removeActiveTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::removeActiveTrack_l(const wp<Track>& t)
+AudioFlinger::MixerThread::~MixerThread()
 {
-    mActiveTracks.remove(t);
+    delete mAudioMixer;
+}
 
-    // Force routing to speaker for certain stream types
-    // The forced routing to speaker is managed by hardware mixer
-    if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
-        sp<Track> track = t.promote();
-        if (track == NULL) return;
+bool AudioFlinger::MixerThread::threadLoop()
+{
+    unsigned long sleepTime = kBufferRecoveryInUsecs;
+    int16_t* curBuf = mMixBuffer;
+    Vector< sp<Track> > tracksToRemove;
+    size_t enabledTracks = 0;
+    nsecs_t standbyTime = systemTime();
+    size_t mixBufferSize = mFrameCount * mFrameSize;
+    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
 
-        if (streamForcedToSpeaker(track->type())) {
-            mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_REMOVED);
+    while (!exitPending())
+    {
+        processConfigEvents();
+
+        enabledTracks = 0;
+        { // scope for mLock
+
+            Mutex::Autolock _l(mLock);
+
+            if (checkForNewParameters_l()) {
+                mixBufferSize = mFrameCount * mFrameSize;
+                maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
+            }
+
+            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+            // put audio hardware into standby after short delay
+            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+                        mSuspended) {
+                if (!mStandby) {
+                    LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
+                    mOutput->standby();
+                    mStandby = true;
+                    mBytesWritten = 0;
+                }
+
+                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+                    // we're about to wait, flush the binder command buffer
+                    IPCThreadState::self()->flushCommands();
+
+                    if (exitPending()) break;
+
+                    // wait until we have something to do...
+                    LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+                    mWaitWorkCV.wait(mLock);
+                    LOGV("MixerThread %p TID %d waking up\n", this, gettid());
+
+                    if (mMasterMute == false) {
+                        char value[PROPERTY_VALUE_MAX];
+                        property_get("ro.audio.silent", value, "0");
+                        if (atoi(value)) {
+                            LOGD("Silence is golden");
+                            setMasterMute(true);
+                        }
+                    }
+
+                    standbyTime = systemTime() + kStandbyTimeInNsecs;
+                    continue;
+                }
+            }
+
+            enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove);
+       }
+
+        if (LIKELY(enabledTracks)) {
+            // mix buffers...
+            mAudioMixer->process(curBuf);
+
+            // output audio to hardware
+            if (mSuspended) {
+                usleep(kMaxBufferRecoveryInUsecs);
+            } else {
+                mLastWriteTime = systemTime();
+                mInWrite = true;
+                int bytesWritten = (int)mOutput->write(curBuf, mixBufferSize);
+                if (bytesWritten > 0) mBytesWritten += bytesWritten;
+                mNumWrites++;
+                mInWrite = false;
+                mStandby = false;
+                nsecs_t temp = systemTime();
+                standbyTime = temp + kStandbyTimeInNsecs;
+                nsecs_t delta = temp - mLastWriteTime;
+                if (delta > maxPeriod) {
+                    LOGW("write blocked for %llu msecs", ns2ms(delta));
+                    mNumDelayedWrites++;
+                }
+                sleepTime = kBufferRecoveryInUsecs;
+            }
+        } else {
+            // There was nothing to mix this round, which means all
+            // active tracks were late. Sleep a little bit to give
+            // them another chance. If we're too late, the audio
+            // hardware will zero-fill for us.
+            // LOGV("thread %p no buffers - usleep(%lu)", this, sleepTime);
+            usleep(sleepTime);
+            if (sleepTime < kMaxBufferRecoveryInUsecs) {
+                sleepTime += kBufferRecoveryInUsecs;
+            }
+        }
+
+        // finally let go of all our tracks, without the lock held
+        // since we can't guarantee the destructors won't acquire that
+        // same lock.
+        tracksToRemove.clear();
+    }
+
+    if (!mStandby) {
+        mOutput->standby();
+    }
+    sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
+    processConfigEvents();
+
+    LOGV("MixerThread %p exiting", this);
+    return false;
+}
+
+// prepareTracks_l() must be called with ThreadBase::mLock held
+size_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
+{
+
+    size_t enabledTracks = 0;
+    // find out which tracks need to be processed
+    size_t count = activeTracks.size();
+    for (size_t i=0 ; i<count ; i++) {
+        sp<Track> t = activeTracks[i].promote();
+        if (t == 0) continue;
+
+        Track* const track = t.get();
+        audio_track_cblk_t* cblk = track->cblk();
+
+        // The first time a track is added we wait
+        // for all its buffers to be filled before processing it
+        mAudioMixer->setActiveTrack(track->name());
+        if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+                !track->isPaused())
+        {
+            //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+            // compute volume for this track
+            int16_t left, right;
+            if (track->isMuted() || mMasterMute || track->isPausing() ||
+                mStreamTypes[track->type()].mute) {
+                left = right = 0;
+                if (track->isPausing()) {
+                    track->setPaused();
+                }
+            } else {
+                float typeVolume = mStreamTypes[track->type()].volume;
+                float v = mMasterVolume * typeVolume;
+                float v_clamped = v * cblk->volume[0];
+                if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                left = int16_t(v_clamped);
+                v_clamped = v * cblk->volume[1];
+                if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                right = int16_t(v_clamped);
+            }
+
+            // XXX: these things DON'T need to be done each time
+            mAudioMixer->setBufferProvider(track);
+            mAudioMixer->enable(AudioMixer::MIXING);
+
+            int param;
+            if ( track->mFillingUpStatus == Track::FS_FILLED) {
+                // no ramp for the first volume setting
+                track->mFillingUpStatus = Track::FS_ACTIVE;
+                if (track->mState == TrackBase::RESUMING) {
+                    track->mState = TrackBase::ACTIVE;
+                    param = AudioMixer::RAMP_VOLUME;
+                } else {
+                    param = AudioMixer::VOLUME;
+                }
+            } else {
+                param = AudioMixer::RAMP_VOLUME;
+            }
+            mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
+            mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
+            mAudioMixer->setParameter(
+                AudioMixer::TRACK,
+                AudioMixer::FORMAT, track->format());
+            mAudioMixer->setParameter(
+                AudioMixer::TRACK,
+                AudioMixer::CHANNEL_COUNT, track->channelCount());
+            mAudioMixer->setParameter(
+                AudioMixer::RESAMPLE,
+                AudioMixer::SAMPLE_RATE,
+                int(cblk->sampleRate));
+
+            // reset retry count
+            track->mRetryCount = kMaxTrackRetries;
+            enabledTracks++;
+        } else {
+            //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+            if (track->isStopped()) {
+                track->reset();
+            }
+            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+                // We have consumed all the buffers of this track.
+                // Remove it from the list of active tracks.
+                tracksToRemove->add(track);
+                mAudioMixer->disable(AudioMixer::MIXING);
+            } else {
+                // No buffers for this track. Give it a few chances to
+                // fill a buffer, then remove it from active list.
+                if (--(track->mRetryCount) <= 0) {
+                    LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+                    tracksToRemove->add(track);
+                }
+                // For tracks using static shared memry buffer, make sure that we have
+                // written enough data to audio hardware before disabling the track
+                // NOTE: this condition with arrive before track->mRetryCount <= 0 so we
+                // don't care about code removing track from active list above.
+                if ((track->mSharedBuffer == 0) || (mBytesWritten >= mMinBytesToWrite)) {
+                    mAudioMixer->disable(AudioMixer::MIXING);
+                } else {
+                    enabledTracks++;
+                }
+            }
+        }
+    }
+
+    // remove all the tracks that need to be...
+    count = tracksToRemove->size();
+    if (UNLIKELY(count)) {
+        for (size_t i=0 ; i<count ; i++) {
+            const sp<Track>& track = tracksToRemove->itemAt(i);
+            mActiveTracks.remove(track);
+            if (track->isTerminated()) {
+                mTracks.remove(track);
+                deleteTrackName_l(track->mName);
+            }
+        }
+    }
+
+    return enabledTracks;
+}
+
+void AudioFlinger::MixerThread::getTracks(
+        SortedVector < sp<Track> >& tracks,
+        SortedVector < wp<Track> >& activeTracks,
+        int streamType)
+{
+    LOGV ("MixerThread::getTracks() mixer %p, mTracks.size %d, mActiveTracks.size %d", this,  mTracks.size(), mActiveTracks.size());
+    Mutex::Autolock _l(mLock);
+    size_t size = mTracks.size();
+    for (size_t i = 0; i < size; i++) {
+        sp<Track> t = mTracks[i];
+        if (t->type() == streamType) {
+            tracks.add(t);
+            int j = mActiveTracks.indexOf(t);
+            if (j >= 0) {
+                t = mActiveTracks[j].promote();
+                if (t != NULL) {
+                    activeTracks.add(t);
+                }
+            }
+        }
+    }
+
+    size = activeTracks.size();
+    for (size_t i = 0; i < size; i++) {
+        mActiveTracks.remove(activeTracks[i]);
+    }
+
+    size = tracks.size();
+    for (size_t i = 0; i < size; i++) {
+        sp<Track> t = tracks[i];
+        mTracks.remove(t);
+        deleteTrackName_l(t->name());
+    }
+}
+
+void AudioFlinger::MixerThread::putTracks(
+        SortedVector < sp<Track> >& tracks,
+        SortedVector < wp<Track> >& activeTracks)
+{
+    LOGV ("MixerThread::putTracks() mixer %p, tracks.size %d, activeTracks.size %d", this,  tracks.size(), activeTracks.size());
+    Mutex::Autolock _l(mLock);
+    size_t size = tracks.size();
+    for (size_t i = 0; i < size ; i++) {
+        sp<Track> t = tracks[i];
+        int name = getTrackName_l();
+
+        if (name < 0) return;
+
+        t->mName = name;
+        t->mThread = this;
+        mTracks.add(t);
+
+        int j = activeTracks.indexOf(t);
+        if (j >= 0) {
+            mActiveTracks.add(t);
         }
     }
 }
 
-// getTrackName_l() must be called with AudioFlinger::mLock held
+// getTrackName_l() must be called with ThreadBase::mLock held
 int AudioFlinger::MixerThread::getTrackName_l()
 {
     return mAudioMixer->getTrackName();
 }
 
-// deleteTrackName_l() must be called with AudioFlinger::mLock held
+// deleteTrackName_l() must be called with ThreadBase::mLock held
 void AudioFlinger::MixerThread::deleteTrackName_l(int name)
 {
     mAudioMixer->deleteTrackName(name);
 }
 
-size_t AudioFlinger::MixerThread::getOutputFrameCount() 
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::MixerThread::checkForNewParameters_l()
 {
-    return mOutput->bufferSize() / mOutput->channelCount() / sizeof(int16_t);
+    bool reconfig = false;
+
+    if (mNewParameters != "") {
+        status_t status = NO_ERROR;
+        AudioParameter param = AudioParameter(mNewParameters);
+        int value;
+        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+            if (value != AudioSystem::PCM_16_BIT) {
+                status = BAD_VALUE;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+            if (value != AudioSystem::CHANNEL_OUT_STEREO) {
+                status = BAD_VALUE;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be garantied
+            // if frame count is changed after track creation
+            if (!mTracks.isEmpty()) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (status == NO_ERROR) {
+            status = mOutput->setParameters(mNewParameters);
+            if (!mStandby && status == INVALID_OPERATION) {
+               mOutput->standby();
+               mStandby = true;
+               mBytesWritten = 0;
+               status = mOutput->setParameters(mNewParameters);
+            }
+            if (status == NO_ERROR && reconfig) {
+                delete mAudioMixer;
+                readOutputParameters();
+                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+                for (size_t i = 0; i < mTracks.size() ; i++) {
+                    int name = getTrackName_l();
+                    if (name < 0) break;
+                    mTracks[i]->mName = name;
+                }
+                sendConfigEvent(AudioSystem::OUTPUT_CONFIG_CHANGED);
+            }
+        }
+        mParamStatus = status;
+        mNewParameters = "";
+        mParamCond.signal();
+    }
+    return reconfig;
+}
+
+status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    PlaybackThread::dumpInternals(fd, args);
+
+    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+    :   PlaybackThread(audioFlinger, output),
+    mLeftVolume (1.0), mRightVolume(1.0)
+{
+    mType = PlaybackThread::DIRECT;
+}
+
+AudioFlinger::DirectOutputThread::~DirectOutputThread()
+{
+}
+
+
+bool AudioFlinger::DirectOutputThread::threadLoop()
+{
+    unsigned long sleepTime = kBufferRecoveryInUsecs;
+    sp<Track> trackToRemove;
+    sp<Track> activeTrack;
+    nsecs_t standbyTime = systemTime();
+    int8_t *curBuf;
+    size_t mixBufferSize = mFrameCount*mFrameSize;
+
+    while (!exitPending())
+    {
+        processConfigEvents();
+
+        { // scope for the mLock
+
+            Mutex::Autolock _l(mLock);
+
+            if (checkForNewParameters_l()) {
+                mixBufferSize = mFrameCount*mFrameSize;
+            }
+
+            // put audio hardware into standby after short delay
+            if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+                        mSuspended) {
+                // wait until we have something to do...
+                if (!mStandby) {
+                    LOGV("Audio hardware entering standby, mixer %p\n", this);
+                    mOutput->standby();
+                    mStandby = true;
+                    mBytesWritten = 0;
+                }
+
+                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
+                    // we're about to wait, flush the binder command buffer
+                    IPCThreadState::self()->flushCommands();
+
+                    if (exitPending()) break;
+
+                    LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+                    mWaitWorkCV.wait(mLock);
+                    LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+
+                    if (mMasterMute == false) {
+                        char value[PROPERTY_VALUE_MAX];
+                        property_get("ro.audio.silent", value, "0");
+                        if (atoi(value)) {
+                            LOGD("Silence is golden");
+                            setMasterMute(true);
+                        }
+                    }
+
+                    standbyTime = systemTime() + kStandbyTimeInNsecs;
+                    continue;
+                }
+            }
+
+            // find out which tracks need to be processed
+            if (mActiveTracks.size() != 0) {
+                sp<Track> t = mActiveTracks[0].promote();
+                if (t == 0) continue;
+
+                Track* const track = t.get();
+                audio_track_cblk_t* cblk = track->cblk();
+
+                // The first time a track is added we wait
+                // for all its buffers to be filled before processing it
+                if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+                        !track->isPaused())
+                {
+                    //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+                    // compute volume for this track
+                    float left, right;
+                    if (track->isMuted() || mMasterMute || track->isPausing() ||
+                        mStreamTypes[track->type()].mute) {
+                        left = right = 0;
+                        if (track->isPausing()) {
+                            track->setPaused();
+                        }
+                    } else {
+                        float typeVolume = mStreamTypes[track->type()].volume;
+                        float v = mMasterVolume * typeVolume;
+                        float v_clamped = v * cblk->volume[0];
+                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                        left = v_clamped/MAX_GAIN;
+                        v_clamped = v * cblk->volume[1];
+                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                        right = v_clamped/MAX_GAIN;
+                    }
+
+                    if (left != mLeftVolume || right != mRightVolume) {
+                        mOutput->setVolume(left, right);
+                        left = mLeftVolume;
+                        right = mRightVolume;
+                    }
+
+                    if (track->mFillingUpStatus == Track::FS_FILLED) {
+                        track->mFillingUpStatus = Track::FS_ACTIVE;
+                        if (track->mState == TrackBase::RESUMING) {
+                            track->mState = TrackBase::ACTIVE;
+                        }
+                    }
+
+                    // reset retry count
+                    track->mRetryCount = kMaxTrackRetries;
+                    activeTrack = t;
+                } else {
+                    //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+                    if (track->isStopped()) {
+                        track->reset();
+                    }
+                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+                        // We have consumed all the buffers of this track.
+                        // Remove it from the list of active tracks.
+                        trackToRemove = track;
+                    } else {
+                        // No buffers for this track. Give it a few chances to
+                        // fill a buffer, then remove it from active list.
+                        if (--(track->mRetryCount) <= 0) {
+                            LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+                            trackToRemove = track;
+                        }
+
+                        // For tracks using static shared memry buffer, make sure that we have
+                        // written enough data to audio hardware before disabling the track
+                        // NOTE: this condition with arrive before track->mRetryCount <= 0 so we
+                        // don't care about code removing track from active list above.
+                        if ((track->mSharedBuffer != 0) && (mBytesWritten < mMinBytesToWrite)) {
+                            activeTrack = t;
+                        }
+                     }
+                }
+            }
+
+            // remove all the tracks that need to be...
+            if (UNLIKELY(trackToRemove != 0)) {
+                mActiveTracks.remove(trackToRemove);
+                if (trackToRemove->isTerminated()) {
+                    mTracks.remove(trackToRemove);
+                    deleteTrackName_l(trackToRemove->mName);
+                }
+            }
+       }
+
+        if (activeTrack != 0) {
+            AudioBufferProvider::Buffer buffer;
+            size_t frameCount = mFrameCount;
+            curBuf = (int8_t *)mMixBuffer;
+            // output audio to hardware
+            mLastWriteTime = systemTime();
+            mInWrite = true;
+            while(frameCount) {
+                buffer.frameCount = frameCount;
+                activeTrack->getNextBuffer(&buffer);
+                if (UNLIKELY(buffer.raw == 0)) {
+                    memset(curBuf, 0, frameCount * mFrameSize);
+                    break;
+                }
+                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
+                frameCount -= buffer.frameCount;
+                curBuf += buffer.frameCount * mFrameSize;
+                activeTrack->releaseBuffer(&buffer);
+            }
+            if (mSuspended) {
+                usleep(kMaxBufferRecoveryInUsecs);
+            } else {
+                int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
+                if (bytesWritten) mBytesWritten += bytesWritten;
+                mNumWrites++;
+                mInWrite = false;
+                mStandby = false;
+                nsecs_t temp = systemTime();
+                standbyTime = temp + kStandbyTimeInNsecs;
+                sleepTime = kBufferRecoveryInUsecs;
+            }
+        } else {
+            // There was nothing to mix this round, which means all
+            // active tracks were late. Sleep a little bit to give
+            // them another chance. If we're too late, the audio
+            // hardware will zero-fill for us.
+            //LOGV("no buffers - usleep(%lu)", sleepTime);
+            usleep(sleepTime);
+            if (sleepTime < kMaxBufferRecoveryInUsecs) {
+                sleepTime += kBufferRecoveryInUsecs;
+            }
+        }
+
+        // finally let go of removed track, without the lock held
+        // since we can't guarantee the destructors won't acquire that
+        // same lock.
+        trackToRemove.clear();
+        activeTrack.clear();
+    }
+
+    if (!mStandby) {
+        mOutput->standby();
+    }
+    sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
+    processConfigEvents();
+
+    LOGV("DirectOutputThread %p exiting", this);
+    return false;
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::DirectOutputThread::getTrackName_l()
+{
+    return 0;
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+{
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    if (mNewParameters != "") {
+        status_t status = NO_ERROR;
+        AudioParameter param = AudioParameter(mNewParameters);
+        int value;
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be garantied
+            // if frame count is changed after track creation
+            if (!mTracks.isEmpty()) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (status == NO_ERROR) {
+            status = mOutput->setParameters(mNewParameters);
+            if (!mStandby && status == INVALID_OPERATION) {
+               mOutput->standby();
+               mStandby = true;
+               mBytesWritten = 0;
+               status = mOutput->setParameters(mNewParameters);
+            }
+            if (status == NO_ERROR && reconfig) {
+                readOutputParameters();
+                sendConfigEvent(AudioSystem::OUTPUT_CONFIG_CHANGED);
+            }
+        }
+        mParamStatus = status;
+        mNewParameters = "";
+        mParamCond.signal();
+    }
+    return reconfig;
 }
 
 // ----------------------------------------------------------------------------
 
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread)
+    :   MixerThread(audioFlinger, mainThread->getOutput())
+{
+    mType = PlaybackThread::DUPLICATING;
+    addOutputTrack(mainThread);
+}
+
+AudioFlinger::DuplicatingThread::~DuplicatingThread()
+{
+    mOutputTracks.clear();
+}
+
+bool AudioFlinger::DuplicatingThread::threadLoop()
+{
+    unsigned long sleepTime = kBufferRecoveryInUsecs;
+    int16_t* curBuf = mMixBuffer;
+    Vector< sp<Track> > tracksToRemove;
+    size_t enabledTracks = 0;
+    nsecs_t standbyTime = systemTime();
+    size_t mixBufferSize = mFrameCount*mFrameSize;
+    SortedVector< sp<OutputTrack> > outputTracks;
+
+    while (!exitPending())
+    {
+        processConfigEvents();
+
+        enabledTracks = 0;
+        { // scope for the mLock
+
+            Mutex::Autolock _l(mLock);
+
+            if (checkForNewParameters_l()) {
+                mixBufferSize = mFrameCount*mFrameSize;
+            }
+
+            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+            for (size_t i = 0; i < mOutputTracks.size(); i++) {
+                outputTracks.add(mOutputTracks[i]);
+            }
+
+            // put audio hardware into standby after short delay
+            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+                         mSuspended) {
+                if (!mStandby) {
+                    for (size_t i = 0; i < outputTracks.size(); i++) {
+                        mLock.unlock();
+                        outputTracks[i]->stop();
+                        mLock.lock();
+                    }
+                    mStandby = true;
+                    mBytesWritten = 0;
+                }
+
+                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+                    // we're about to wait, flush the binder command buffer
+                    IPCThreadState::self()->flushCommands();
+                    outputTracks.clear();
+
+                    if (exitPending()) break;
+
+                    LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+                    mWaitWorkCV.wait(mLock);
+                    LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+                    if (mMasterMute == false) {
+                        char value[PROPERTY_VALUE_MAX];
+                        property_get("ro.audio.silent", value, "0");
+                        if (atoi(value)) {
+                            LOGD("Silence is golden");
+                            setMasterMute(true);
+                        }
+                    }
+
+                    standbyTime = systemTime() + kStandbyTimeInNsecs;
+                    sleepTime = kBufferRecoveryInUsecs;
+                    continue;
+                }
+            }
+
+            enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove);
+       }
+
+        bool mustSleep = true;
+        if (LIKELY(enabledTracks)) {
+            // mix buffers...
+            mAudioMixer->process(curBuf);
+            if (!mSuspended) {
+                for (size_t i = 0; i < outputTracks.size(); i++) {
+                    outputTracks[i]->write(curBuf, mFrameCount);
+                }
+                mStandby = false;
+                mustSleep = false;
+                mBytesWritten += mixBufferSize;
+            }
+        } else {
+            // flush remaining overflow buffers in output tracks
+            for (size_t i = 0; i < outputTracks.size(); i++) {
+                if (outputTracks[i]->isActive()) {
+                    outputTracks[i]->write(curBuf, 0);
+                    standbyTime = systemTime() + kStandbyTimeInNsecs;
+                    mustSleep = false;
+                }
+            }
+        }
+        if (mustSleep) {
+//            LOGV("threadLoop() sleeping %d", sleepTime);
+            usleep(sleepTime);
+            if (sleepTime < kMaxBufferRecoveryInUsecs) {
+                sleepTime += kBufferRecoveryInUsecs;
+            }
+        } else {
+            sleepTime = kBufferRecoveryInUsecs;
+        }
+
+        // finally let go of all our tracks, without the lock held
+        // since we can't guarantee the destructors won't acquire that
+        // same lock.
+        tracksToRemove.clear();
+        outputTracks.clear();
+    }
+
+    if (!mStandby) {
+        for (size_t i = 0; i < outputTracks.size(); i++) {
+            mLock.unlock();
+            outputTracks[i]->stop();
+            mLock.lock();
+        }
+    }
+
+    sendConfigEvent(AudioSystem::OUTPUT_CLOSED);
+    processConfigEvents();
+
+    return false;
+}
+
+void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+{
+    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
+    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
+                                            mSampleRate,
+                                            mFormat,
+                                            mChannelCount,
+                                            frameCount);
+    thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
+    mOutputTracks.add(outputTrack);
+    LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+}
+
+void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
+{
+    Mutex::Autolock _l(mLock);
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
+            mOutputTracks.removeAt(i);
+            return;
+        }
+    }
+    LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
+}
+
+
+// ----------------------------------------------------------------------------
+
 // TrackBase constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::TrackBase::TrackBase(
-            const sp<MixerThread>& mixerThread,
+AudioFlinger::ThreadBase::TrackBase::TrackBase(
+            const wp<ThreadBase>& thread,
             const sp<Client>& client,
             uint32_t sampleRate,
             int format,
@@ -1574,7 +1951,7 @@
             uint32_t flags,
             const sp<IMemory>& sharedBuffer)
     :   RefBase(),
-        mMixerThread(mixerThread),
+        mThread(thread),
         mClient(client),
         mFrameCount(0),
         mState(IDLE),
@@ -1582,13 +1959,6 @@
         mFormat(format),
         mFlags(flags & ~SYSTEM_FLAGS_MASK)
 {
-    mName = mixerThread->getTrackName_l();
-    LOGV("TrackBase contructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
-    if (mName < 0) {
-        LOGE("no more track names availlable");
-        return;
-    }
-
     LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
 
     // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
@@ -1642,16 +2012,19 @@
    }
 }
 
-AudioFlinger::MixerThread::TrackBase::~TrackBase()
+AudioFlinger::PlaybackThread::TrackBase::~TrackBase()
 {
     if (mCblk) {
-        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.        
+        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
+        if (mClient == NULL) {
+            delete mCblk;
+        }
     }
     mCblkMemory.clear();            // and free the shared memory
     mClient.clear();
 }
 
-void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+void AudioFlinger::PlaybackThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
 {
     buffer->raw = 0;
     mFrameCount = buffer->frameCount;
@@ -1659,7 +2032,7 @@
     buffer->frameCount = 0;
 }
 
-bool AudioFlinger::MixerThread::TrackBase::step() {
+bool AudioFlinger::PlaybackThread::TrackBase::step() {
     bool result;
     audio_track_cblk_t* cblk = this->cblk();
 
@@ -1671,7 +2044,7 @@
     return result;
 }
 
-void AudioFlinger::MixerThread::TrackBase::reset() {
+void AudioFlinger::PlaybackThread::TrackBase::reset() {
     audio_track_cblk_t* cblk = this->cblk();
 
     cblk->user = 0;
@@ -1682,27 +2055,27 @@
     LOGV("TrackBase::reset");
 }
 
-sp<IMemory> AudioFlinger::MixerThread::TrackBase::getCblk() const
+sp<IMemory> AudioFlinger::PlaybackThread::TrackBase::getCblk() const
 {
     return mCblkMemory;
 }
 
-int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
+int AudioFlinger::PlaybackThread::TrackBase::sampleRate() const {
     return (int)mCblk->sampleRate;
 }
 
-int AudioFlinger::MixerThread::TrackBase::channelCount() const {
+int AudioFlinger::PlaybackThread::TrackBase::channelCount() const {
     return (int)mCblk->channels;
 }
 
-void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
+void* AudioFlinger::PlaybackThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
     audio_track_cblk_t* cblk = this->cblk();
-    int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels;
-    int16_t *bufferEnd = bufferStart + frames * cblk->channels;
+    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
+    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
 
     // Check validity of returned pointer in case the track control block would have been corrupted.
-    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 
-        (cblk->channels == 2 && ((unsigned long)bufferStart & 3))) {
+    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
+        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
         LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
                 server %d, serverBase %d, user %d, userBase %d, channels %d",
                 bufferStart, bufferEnd, mBuffer, mBufferEnd,
@@ -1715,9 +2088,9 @@
 
 // ----------------------------------------------------------------------------
 
-// Track constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::Track::Track(
-            const sp<MixerThread>& mixerThread,
+// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
+AudioFlinger::PlaybackThread::Track::Track(
+            const wp<ThreadBase>& thread,
             const sp<Client>& client,
             int streamType,
             uint32_t sampleRate,
@@ -1725,40 +2098,58 @@
             int channelCount,
             int frameCount,
             const sp<IMemory>& sharedBuffer)
-    :   TrackBase(mixerThread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
+    :   TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer),
+    mMute(false), mSharedBuffer(sharedBuffer), mName(-1)
 {
+    sp<ThreadBase> baseThread = thread.promote();
+    if (baseThread != 0) {
+        PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
+        mName = playbackThread->getTrackName_l();
+    }
+    LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+    if (mName < 0) {
+        LOGE("no more track names available");
+    }
     mVolume[0] = 1.0f;
     mVolume[1] = 1.0f;
-    mMute = false;
-    mSharedBuffer = sharedBuffer;
     mStreamType = streamType;
+    // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
+    // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
+    mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
 }
 
-AudioFlinger::MixerThread::Track::~Track()
+AudioFlinger::PlaybackThread::Track::~Track()
 {
-    wp<Track> weak(this); // never create a strong ref from the dtor
-    Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
-    mState = TERMINATED;
-}
-
-void AudioFlinger::MixerThread::Track::destroy()
-{
-    // NOTE: destroyTrack_l() can remove a strong reference to this Track 
-    // by removing it from mTracks vector, so there is a risk that this Tracks's
-    // desctructor is called. As the destructor needs to lock AudioFlinger::mLock,
-    // we must acquire a strong reference on this Track before locking AudioFlinger::mLock
-    // here so that the destructor is called only when exiting this function.
-    // On the other hand, as long as Track::destroy() is only called by 
-    // TrackHandle destructor, the TrackHandle still holds a strong ref on 
-    // this Track with its member mTrack.
-    sp<Track> keep(this);
-    { // scope for AudioFlinger::mLock
-        Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
-        mMixerThread->destroyTrack_l(this);
+    LOGV("PlaybackThread::Track destructor");
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        mState = TERMINATED;
     }
 }
 
-void AudioFlinger::MixerThread::Track::dump(char* buffer, size_t size)
+void AudioFlinger::PlaybackThread::Track::destroy()
+{
+    // NOTE: destroyTrack_l() can remove a strong reference to this Track
+    // by removing it from mTracks vector, so there is a risk that this Tracks's
+    // desctructor is called. As the destructor needs to lock mLock,
+    // we must acquire a strong reference on this Track before locking mLock
+    // here so that the destructor is called only when exiting this function.
+    // On the other hand, as long as Track::destroy() is only called by
+    // TrackHandle destructor, the TrackHandle still holds a strong ref on
+    // this Track with its member mTrack.
+    sp<Track> keep(this);
+    { // scope for mLock
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            Mutex::Autolock _l(thread->mLock);
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            playbackThread->destroyTrack_l(this);
+        }
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
 {
     snprintf(buffer, size, "  %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
             mName - AudioMixer::TRACK0,
@@ -1777,7 +2168,7 @@
             mCblk->user);
 }
 
-status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
 {
      audio_track_cblk_t* cblk = this->cblk();
      uint32_t framesReady;
@@ -1814,76 +2205,90 @@
 getNextBuffer_exit:
      buffer->raw = 0;
      buffer->frameCount = 0;
+     LOGV("getNextBuffer() no more data");
      return NOT_ENOUGH_DATA;
 }
 
-bool AudioFlinger::MixerThread::Track::isReady() const {
+bool AudioFlinger::PlaybackThread::Track::isReady() const {
     if (mFillingUpStatus != FS_FILLING) return true;
 
     if (mCblk->framesReady() >= mCblk->frameCount ||
         mCblk->forceReady) {
         mFillingUpStatus = FS_FILLED;
         mCblk->forceReady = 0;
-        LOGV("Track::isReady() track %d for output %d", mName, mMixerThread->mOutputType);
         return true;
     }
     return false;
 }
 
-status_t AudioFlinger::MixerThread::Track::start()
+status_t AudioFlinger::PlaybackThread::Track::start()
 {
-    LOGV("start(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
-    Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
-    mMixerThread->addTrack_l(this);
+    LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        playbackThread->addTrack_l(this);
+    }
     return NO_ERROR;
 }
 
-void AudioFlinger::MixerThread::Track::stop()
+void AudioFlinger::PlaybackThread::Track::stop()
 {
-    LOGV("stop(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
-    Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
-    if (mState > STOPPED) {
-        mState = STOPPED;
-        // If the track is not active (PAUSED and buffers full), flush buffers
-        if (mMixerThread->mActiveTracks.indexOf(this) < 0) {
-            reset();
+    LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        if (mState > STOPPED) {
+            mState = STOPPED;
+            // If the track is not active (PAUSED and buffers full), flush buffers
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+                reset();
+            }
+            LOGV("(> STOPPED) => STOPPED (%d)", mName);
         }
-        LOGV("(> STOPPED) => STOPPED (%d)", mName);
     }
 }
 
-void AudioFlinger::MixerThread::Track::pause()
+void AudioFlinger::PlaybackThread::Track::pause()
 {
     LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
-    Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
-    if (mState == ACTIVE || mState == RESUMING) {
-        mState = PAUSING;
-        LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        if (mState == ACTIVE || mState == RESUMING) {
+            mState = PAUSING;
+            LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
+        }
     }
 }
 
-void AudioFlinger::MixerThread::Track::flush()
+void AudioFlinger::PlaybackThread::Track::flush()
 {
     LOGV("flush(%d)", mName);
-    Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
-    if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
-        return;
-    }
-    // No point remaining in PAUSED state after a flush => go to
-    // STOPPED state
-    mState = STOPPED;
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
+            return;
+        }
+        // No point remaining in PAUSED state after a flush => go to
+        // STOPPED state
+        mState = STOPPED;
 
-    mCblk->lock.lock();
-    // NOTE: reset() will reset cblk->user and cblk->server with
-    // the risk that at the same time, the AudioMixer is trying to read
-    // data. In this case, getNextBuffer() would return a NULL pointer
-    // as audio buffer => the AudioMixer code MUST always test that pointer
-    // returned by getNextBuffer() is not NULL!
-    reset();
-    mCblk->lock.unlock();
+        mCblk->lock.lock();
+        // NOTE: reset() will reset cblk->user and cblk->server with
+        // the risk that at the same time, the AudioMixer is trying to read
+        // data. In this case, getNextBuffer() would return a NULL pointer
+        // as audio buffer => the AudioMixer code MUST always test that pointer
+        // returned by getNextBuffer() is not NULL!
+        reset();
+        mCblk->lock.unlock();
+    }
 }
 
-void AudioFlinger::MixerThread::Track::reset()
+void AudioFlinger::PlaybackThread::Track::reset()
 {
     // Do not reset twice to avoid discarding data written just after a flush and before
     // the audioflinger thread detects the track is stopped.
@@ -1893,17 +2298,17 @@
         // written to buffer
         mCblk->flowControlFlag = 1;
         mCblk->forceReady = 0;
-        mFillingUpStatus = FS_FILLING;        
+        mFillingUpStatus = FS_FILLING;
         mResetDone = true;
     }
 }
 
-void AudioFlinger::MixerThread::Track::mute(bool muted)
+void AudioFlinger::PlaybackThread::Track::mute(bool muted)
 {
     mMute = muted;
 }
 
-void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
+void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
 {
     mVolume[0] = left;
     mVolume[1] = right;
@@ -1912,28 +2317,33 @@
 // ----------------------------------------------------------------------------
 
 // RecordTrack constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::RecordTrack::RecordTrack(
-            const sp<MixerThread>& mixerThread,
+AudioFlinger::RecordThread::RecordTrack::RecordTrack(
+            const wp<ThreadBase>& thread,
             const sp<Client>& client,
-            int inputSource,
             uint32_t sampleRate,
             int format,
             int channelCount,
             int frameCount,
             uint32_t flags)
-    :   TrackBase(mixerThread, client, sampleRate, format,
+    :   TrackBase(thread, client, sampleRate, format,
                   channelCount, frameCount, flags, 0),
-        mOverflow(false), mInputSource(inputSource)
+        mOverflow(false)
+{
+   LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+   if (format == AudioSystem::PCM_16_BIT) {
+       mCblk->frameSize = channelCount * sizeof(int16_t);
+   } else if (format == AudioSystem::PCM_8_BIT) {
+       mCblk->frameSize = channelCount * sizeof(int8_t);
+   } else {
+       mCblk->frameSize = sizeof(int8_t);
+   }
+}
+
+AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
 {
 }
 
-AudioFlinger::MixerThread::RecordTrack::~RecordTrack()
-{
-    Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
-    mMixerThread->deleteTrackName_l(mName);
-}
-
-status_t AudioFlinger::MixerThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
 {
     audio_track_cblk_t* cblk = this->cblk();
     uint32_t framesAvail;
@@ -1972,180 +2382,231 @@
     return NOT_ENOUGH_DATA;
 }
 
-status_t AudioFlinger::MixerThread::RecordTrack::start()
+status_t AudioFlinger::RecordThread::RecordTrack::start()
 {
-    return mMixerThread->mAudioFlinger->startRecord(this);
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        RecordThread *recordThread = (RecordThread *)thread.get();
+        return recordThread->start(this);
+    }
+    return NO_INIT;
 }
 
-void AudioFlinger::MixerThread::RecordTrack::stop()
+void AudioFlinger::RecordThread::RecordTrack::stop()
 {
-    mMixerThread->mAudioFlinger->stopRecord(this);
-    TrackBase::reset();
-    // Force overerrun condition to avoid false overrun callback until first data is
-    // read from buffer
-    mCblk->flowControlFlag = 1;
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        RecordThread *recordThread = (RecordThread *)thread.get();
+        recordThread->stop(this);
+        TrackBase::reset();
+        // Force overerrun condition to avoid false overrun callback until first data is
+        // read from buffer
+        mCblk->flowControlFlag = 1;
+    }
 }
 
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::MixerThread::OutputTrack::OutputTrack(
-            const sp<MixerThread>& mixerThread,
+AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
+            const wp<ThreadBase>& thread,
             uint32_t sampleRate,
             int format,
             int channelCount,
             int frameCount)
-    :   Track(mixerThread, NULL, AudioSystem::SYSTEM, sampleRate, format, channelCount, frameCount, NULL),
-    mOutputMixerThread(mixerThread)
+    :   Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL),
+    mActive(false)
 {
-                
+
+    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
     mCblk->out = 1;
     mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
     mCblk->volume[0] = mCblk->volume[1] = 0x1000;
     mOutBuffer.frameCount = 0;
-    mCblk->bufferTimeoutMs = 10;
-    
-    LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p", 
-            mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd);
-    
+    mWaitTimeMs = (playbackThread->frameCount() * 2 * 1000) / playbackThread->sampleRate();
+
+    LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p mWaitTimeMs %d",
+            mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd, mWaitTimeMs);
+
 }
 
-AudioFlinger::MixerThread::OutputTrack::~OutputTrack()
+AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
 {
     stop();
 }
 
-status_t AudioFlinger::MixerThread::OutputTrack::start()
+status_t AudioFlinger::PlaybackThread::OutputTrack::start()
 {
     status_t status = Track::start();
-    
+    if (status != NO_ERROR) {
+        return status;
+    }
+
+    mActive = true;
     mRetryCount = 127;
     return status;
 }
 
-void AudioFlinger::MixerThread::OutputTrack::stop()
+void AudioFlinger::PlaybackThread::OutputTrack::stop()
 {
     Track::stop();
     clearBufferQueue();
     mOutBuffer.frameCount = 0;
+    mActive = false;
 }
 
-void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frames)
+bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
 {
     Buffer *pInBuffer;
     Buffer inBuffer;
     uint32_t channels = mCblk->channels;
-        
+    bool outputBufferFull = false;
     inBuffer.frameCount = frames;
     inBuffer.i16 = data;
-    
-    if (mCblk->user == 0) {
-        mOutputMixerThread->mAudioFlinger->mLock.lock();
-        bool isMusicActive = mOutputMixerThread->isMusicActive_l();
-        mOutputMixerThread->mAudioFlinger->mLock.unlock();
-        if (isMusicActive) {
-            mCblk->forceReady = 1;
-            LOGV("OutputTrack::start() force ready");
-        } else if (mCblk->frameCount > frames){
-            if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
-                uint32_t startFrames = (mCblk->frameCount - frames);
-                LOGV("OutputTrack::start() write %d frames", startFrames);
-                pInBuffer = new Buffer;
-                pInBuffer->mBuffer = new int16_t[startFrames * channels];
-                pInBuffer->frameCount = startFrames;
-                pInBuffer->i16 = pInBuffer->mBuffer;
-                memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
-                mBufferQueue.add(pInBuffer);                
-            } else {
-                LOGW ("OutputTrack::write() no more buffers");
+
+    uint32_t waitTimeLeftMs = mWaitTimeMs;
+
+    if (!mActive) {
+        start();
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            MixerThread *mixerThread = (MixerThread *)thread.get();
+            if (mCblk->frameCount > frames){
+                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+                    uint32_t startFrames = (mCblk->frameCount - frames);
+                    pInBuffer = new Buffer;
+                    pInBuffer->mBuffer = new int16_t[startFrames * channels];
+                    pInBuffer->frameCount = startFrames;
+                    pInBuffer->i16 = pInBuffer->mBuffer;
+                    memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
+                    mBufferQueue.add(pInBuffer);
+                } else {
+                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
+                }
             }
-        }        
+        }
     }
 
-    while (1) { 
+    while (waitTimeLeftMs) {
         // First write pending buffers, then new data
         if (mBufferQueue.size()) {
             pInBuffer = mBufferQueue.itemAt(0);
         } else {
             pInBuffer = &inBuffer;
         }
- 
+
         if (pInBuffer->frameCount == 0) {
             break;
         }
-        
+
         if (mOutBuffer.frameCount == 0) {
             mOutBuffer.frameCount = pInBuffer->frameCount;
-            if (obtainBuffer(&mOutBuffer) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+            nsecs_t startTime = systemTime();
+            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+                LOGV ("OutputTrack::write() %p no more output buffers", this);
+                outputBufferFull = true;
                 break;
             }
+            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
+//            LOGV("OutputTrack::write() waitTimeMs %d waitTimeLeftMs %d", waitTimeMs, waitTimeLeftMs)
+            if (waitTimeLeftMs >= waitTimeMs) {
+                waitTimeLeftMs -= waitTimeMs;
+            } else {
+                waitTimeLeftMs = 0;
+            }
         }
-            
+
         uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
         memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));
         mCblk->stepUser(outFrames);
         pInBuffer->frameCount -= outFrames;
         pInBuffer->i16 += outFrames * channels;
         mOutBuffer.frameCount -= outFrames;
-        mOutBuffer.i16 += outFrames * channels;            
-        
+        mOutBuffer.i16 += outFrames * channels;
+
         if (pInBuffer->frameCount == 0) {
             if (mBufferQueue.size()) {
                 mBufferQueue.removeAt(0);
                 delete [] pInBuffer->mBuffer;
                 delete pInBuffer;
+                LOGV("OutputTrack::write() %p released overflow buffer %d", this, mBufferQueue.size());
             } else {
                 break;
             }
         }
     }
- 
+
     // If we could not write all frames, allocate a buffer and queue it for next time.
     if (inBuffer.frameCount) {
-        if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
+        if (mBufferQueue.size() < kMaxOverFlowBuffers) {
             pInBuffer = new Buffer;
             pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels];
             pInBuffer->frameCount = inBuffer.frameCount;
             pInBuffer->i16 = pInBuffer->mBuffer;
             memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t));
             mBufferQueue.add(pInBuffer);
+            LOGV("OutputTrack::write() %p adding overflow buffer %d", this, mBufferQueue.size());
         } else {
-            LOGW("OutputTrack::write() no more buffers");
+            LOGW("OutputTrack::write() %p no more overflow buffers", this);
         }
     }
-    
+
     // Calling write() with a 0 length buffer, means that no more data will be written:
-    // If no more buffers are pending, fill output track buffer to make sure it is started 
+    // If no more buffers are pending, fill output track buffer to make sure it is started
     // by output mixer.
-    if (frames == 0 && mBufferQueue.size() == 0 && mCblk->user < mCblk->frameCount) {
-        frames = mCblk->frameCount - mCblk->user;
-        pInBuffer = new Buffer;
-        pInBuffer->mBuffer = new int16_t[frames * channels];
-        pInBuffer->frameCount = frames;
-        pInBuffer->i16 = pInBuffer->mBuffer;
-        memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
-        mBufferQueue.add(pInBuffer);
+    if (frames == 0 && mBufferQueue.size() == 0) {
+        if (mCblk->user < mCblk->frameCount) {
+            frames = mCblk->frameCount - mCblk->user;
+            pInBuffer = new Buffer;
+            pInBuffer->mBuffer = new int16_t[frames * channels];
+            pInBuffer->frameCount = frames;
+            pInBuffer->i16 = pInBuffer->mBuffer;
+            memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
+            mBufferQueue.add(pInBuffer);
+        } else {
+            stop();
+        }
     }
 
+    return outputBufferFull;
 }
 
-status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
 {
     int active;
-    int timeout = 0;
     status_t result;
     audio_track_cblk_t* cblk = mCblk;
     uint32_t framesReq = buffer->frameCount;
 
-    LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
+//    LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
     buffer->frameCount  = 0;
-    
+
     uint32_t framesAvail = cblk->framesAvailable();
 
+
     if (framesAvail == 0) {
-        return AudioTrack::NO_MORE_BUFFERS;
+        Mutex::Autolock _l(cblk->lock);
+        goto start_loop_here;
+        while (framesAvail == 0) {
+            active = mActive;
+            if (UNLIKELY(!active)) {
+                LOGV("Not active and NO_MORE_BUFFERS");
+                return AudioTrack::NO_MORE_BUFFERS;
+            }
+            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+            if (result != NO_ERROR) {
+                return AudioTrack::NO_MORE_BUFFERS;
+            }
+            // read the server count again
+        start_loop_here:
+            framesAvail = cblk->framesAvailable_l();
+        }
     }
 
+//    if (framesAvail < framesReq) {
+//        return AudioTrack::NO_MORE_BUFFERS;
+//    }
+
     if (framesReq > framesAvail) {
         framesReq = framesAvail;
     }
@@ -2163,11 +2624,11 @@
 }
 
 
-void AudioFlinger::MixerThread::OutputTrack::clearBufferQueue()
+void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
 {
     size_t size = mBufferQueue.size();
     Buffer *pBuffer;
-    
+
     for (size_t i = 0; i < size; i++) {
         pBuffer = mBufferQueue.itemAt(i);
         delete [] pBuffer->mBuffer;
@@ -2199,7 +2660,7 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::MixerThread::Track>& track)
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
     : BnAudioTrack(),
       mTrack(track)
 {
@@ -2251,7 +2712,7 @@
 
 sp<IAudioRecord> AudioFlinger::openRecord(
         pid_t pid,
-        int inputSource,
+        void *input,
         uint32_t sampleRate,
         int format,
         int channelCount,
@@ -2259,14 +2720,13 @@
         uint32_t flags,
         status_t *status)
 {
-    sp<MixerThread::RecordTrack> recordTrack;
+    sp<RecordThread::RecordTrack> recordTrack;
     sp<RecordHandle> recordHandle;
     sp<Client> client;
     wp<Client> wclient;
-    AudioStreamIn* input = 0;
-    int inFrameCount;
-    size_t inputBufferSize;
     status_t lStatus;
+    RecordThread *thread;
+    size_t inFrameCount;
 
     // check calling permissions
     if (!recordingAllowed()) {
@@ -2274,30 +2734,15 @@
         goto Exit;
     }
 
-    if (uint32_t(inputSource) >= AudioRecord::NUM_INPUT_SOURCES) {
-        LOGE("invalid stream type");
-        lStatus = BAD_VALUE;
-        goto Exit;
-    }
-
-    if (mAudioRecordThread == 0) {
-        LOGE("Audio record thread not started");
-        lStatus = NO_INIT;
-        goto Exit;
-    }
-
-
-    // Check that audio input stream accepts requested audio parameters 
-    inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
-    if (inputBufferSize == 0) {
-        lStatus = BAD_VALUE;
-        LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d",  sampleRate, format, channelCount);
-        goto Exit;
-    }
-
     // add client to list
     { // scope for mLock
         Mutex::Autolock _l(mLock);
+        thread = checkRecordThread_l(input);
+        if (thread == NULL) {
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+
         wclient = mClients.valueFor(pid);
         if (wclient != NULL) {
             client = wclient.promote();
@@ -2306,12 +2751,8 @@
             mClients.add(pid, client);
         }
 
-        // frameCount must be a multiple of input buffer size
-        inFrameCount = inputBufferSize/channelCount/sizeof(short);
-        frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
-    
         // create new record track. The record track uses one track in mHardwareMixerThread by convention.
-        recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, inputSource, sampleRate,
+        recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate,
                                                    format, channelCount, frameCount, flags);
     }
     if (recordTrack->getCblk() == NULL) {
@@ -2331,22 +2772,9 @@
     return recordHandle;
 }
 
-status_t AudioFlinger::startRecord(MixerThread::RecordTrack* recordTrack) {
-    if (mAudioRecordThread != 0) {
-        return mAudioRecordThread->start(recordTrack);        
-    }
-    return NO_INIT;
-}
-
-void AudioFlinger::stopRecord(MixerThread::RecordTrack* recordTrack) {
-    if (mAudioRecordThread != 0) {
-        mAudioRecordThread->stop(recordTrack);
-    }
-}
-
 // ----------------------------------------------------------------------------
 
-AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::MixerThread::RecordTrack>& recordTrack)
+AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
     : BnAudioRecord(),
     mRecordTrack(recordTrack)
 {
@@ -2378,86 +2806,165 @@
 
 // ----------------------------------------------------------------------------
 
-AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware,
-            const sp<AudioFlinger>& audioFlinger) :
-    mAudioHardware(audioHardware),
-    mAudioFlinger(audioFlinger),
-    mActive(false)
+AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels) :
+    ThreadBase(audioFlinger),
+    mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
 {
+    mReqChannelCount = AudioSystem::popCount(channels);
+    mReqSampleRate = sampleRate;
+    readInputParameters();
+    sendConfigEvent(AudioSystem::INPUT_OPENED);
 }
 
-AudioFlinger::AudioRecordThread::~AudioRecordThread()
+
+AudioFlinger::RecordThread::~RecordThread()
 {
+    mAudioFlinger->mAudioHardware->closeInputStream(mInput);
+    delete[] mRsmpInBuffer;
+    if (mResampler != 0) {
+        delete mResampler;
+        delete[] mRsmpOutBuffer;
+    }
 }
 
-bool AudioFlinger::AudioRecordThread::threadLoop()
+void AudioFlinger::RecordThread::onFirstRef()
 {
-    LOGV("AudioRecordThread: start record loop");
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+
+    snprintf(buffer, SIZE, "Record Thread %p", this);
+
+    run(buffer, PRIORITY_URGENT_AUDIO);
+}
+bool AudioFlinger::RecordThread::threadLoop()
+{
     AudioBufferProvider::Buffer buffer;
-    int inBufferSize = 0;
-    int inFrameCount = 0;
-    AudioStreamIn* input = 0;
+    sp<RecordTrack> activeTrack;
 
-    mActive = 0;
-    
     // start recording
     while (!exitPending()) {
-        if (!mActive) {
-            mLock.lock();
-            if (!mActive && !exitPending()) {
-                LOGV("AudioRecordThread: loop stopping");
-                if (input) {
-                    delete input;
-                    input = 0;
-                }
-                mRecordTrack.clear();
-                mStopped.signal();
 
+        processConfigEvents();
+
+        { // scope for mLock
+            Mutex::Autolock _l(mLock);
+            checkForNewParameters_l();
+            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
+                if (!mStandby) {
+                    mInput->standby();
+                    mStandby = true;
+                }
+
+                if (exitPending()) break;
+
+                LOGV("RecordThread: loop stopping");
+                // go to sleep
                 mWaitWorkCV.wait(mLock);
-               
-                LOGV("AudioRecordThread: loop starting");
-                if (mRecordTrack != 0) {
-                    input = mAudioHardware->openInputStream(
-                                    mRecordTrack->inputSource(),
-                                    mRecordTrack->format(), 
-                                    mRecordTrack->channelCount(), 
-                                    mRecordTrack->sampleRate(), 
-                                    &mStartStatus,
-                                    (AudioSystem::audio_in_acoustics)(mRecordTrack->mFlags >> 16));
-                    if (input != 0) {
-                        inBufferSize = input->bufferSize();
-                        inFrameCount = inBufferSize/input->frameSize();                        
+                LOGV("RecordThread: loop starting");
+                continue;
+            }
+            if (mActiveTrack != 0) {
+                if (mActiveTrack->mState == TrackBase::PAUSING) {
+                    mActiveTrack.clear();
+                    mStartStopCond.broadcast();
+                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
+                    mRsmpInIndex = mFrameCount;
+                    if (mReqChannelCount != mActiveTrack->channelCount()) {
+                        mActiveTrack.clear();
+                    } else {
+                        mActiveTrack->mState == TrackBase::ACTIVE;
+                    }
+                    mStartStopCond.broadcast();
+                }
+                mStandby = false;
+            }
+        }
+
+        if (mActiveTrack != 0) {
+            buffer.frameCount = mFrameCount;
+            if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+                size_t framesOut = buffer.frameCount;
+                if (mResampler == 0) {
+                    // no resampling
+                    while (framesOut) {
+                        size_t framesIn = mFrameCount - mRsmpInIndex;
+                        if (framesIn) {
+                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
+                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
+                            if (framesIn > framesOut)
+                                framesIn = framesOut;
+                            mRsmpInIndex += framesIn;
+                            framesOut -= framesIn;
+                            if (mChannelCount == mReqChannelCount ||
+                                mFormat != AudioSystem::PCM_16_BIT) {
+                                memcpy(dst, src, framesIn * mFrameSize);
+                            } else {
+                                int16_t *src16 = (int16_t *)src;
+                                int16_t *dst16 = (int16_t *)dst;
+                                if (mChannelCount == 1) {
+                                    while (framesIn--) {
+                                        *dst16++ = *src16;
+                                        *dst16++ = *src16++;
+                                    }
+                                } else {
+                                    while (framesIn--) {
+                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
+                                        src16 += 2;
+                                    }
+                                }
+                            }
+                        }
+                        if (framesOut && mFrameCount == mRsmpInIndex) {
+                            ssize_t bytesRead;
+                            if (framesOut == mFrameCount &&
+                                (mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {
+                                bytesRead = mInput->read(buffer.raw, mInputBytes);
+                                framesOut = 0;
+                            } else {
+                                bytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+                                mRsmpInIndex = 0;
+                            }
+                            if (bytesRead < 0) {
+                                LOGE("Error reading audio input");
+                                sleep(1);
+                                mRsmpInIndex = mFrameCount;
+                                framesOut = 0;
+                                buffer.frameCount = 0;
+                            }
+                        }
                     }
                 } else {
-                    mStartStatus = NO_INIT;
-                }
-                if (mStartStatus !=NO_ERROR) {
-                    LOGW("record start failed, status %d", mStartStatus);
-                    mActive = false;
-                    mRecordTrack.clear();                    
-                }
-                mWaitWorkCV.signal();
-            }
-            mLock.unlock();
-        } else if (mRecordTrack != 0) {
+                    // resampling
 
-            buffer.frameCount = inFrameCount;
-            if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR &&
-                       (int)buffer.frameCount == inFrameCount)) {
-                LOGV("AudioRecordThread read: %d frames", buffer.frameCount);
-                ssize_t bytesRead = input->read(buffer.raw, inBufferSize);
-                if (bytesRead < 0) {
-                    LOGE("Error reading audio input");
-                    sleep(1);
-                }
-                mRecordTrack->releaseBuffer(&buffer);
-                mRecordTrack->overflow();
-            }
+                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
+                    // alter output frame count as if we were expecting stereo samples
+                    if (mChannelCount == 1 && mReqChannelCount == 1) {
+                        framesOut >>= 1;
+                    }
+                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
+                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
+                    // are 32 bit aligned which should be always true.
+                    if (mChannelCount == 2 && mReqChannelCount == 1) {
+                        AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+                        // the resampler always outputs stereo samples: do post stereo to mono conversion
+                        int16_t *src = (int16_t *)mRsmpOutBuffer;
+                        int16_t *dst = buffer.i16;
+                        while (framesOut--) {
+                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
+                            src += 2;
+                        }
+                    } else {
+                        AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+                    }
 
+                }
+                mActiveTrack->releaseBuffer(&buffer);
+                mActiveTrack->overflow();
+            }
             // client isn't retrieving buffers fast enough
             else {
-                if (!mRecordTrack->setOverflow())
-                    LOGW("AudioRecordThread: buffer overflow");
+                if (!mActiveTrack->setOverflow())
+                    LOGW("RecordThread: buffer overflow");
                 // Release the processor for a while before asking for a new buffer.
                 // This will give the application more chance to read from the buffer and
                 // clear the overflow.
@@ -2466,65 +2973,64 @@
         }
     }
 
-
-    if (input) {
-        delete input;
+    if (!mStandby) {
+        mInput->standby();
     }
-    mRecordTrack.clear();
-    
+    mActiveTrack.clear();
+
+    sendConfigEvent(AudioSystem::INPUT_CLOSED);
+    processConfigEvents();
+
+    LOGV("RecordThread %p exiting", this);
     return false;
 }
 
-status_t AudioFlinger::AudioRecordThread::start(MixerThread::RecordTrack* recordTrack)
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
 {
-    LOGV("AudioRecordThread::start");
+    LOGV("RecordThread::start");
     AutoMutex lock(&mLock);
-    mActive = true;
-    // If starting the active track, just reset mActive in case a stop
-    // was pending and exit
-    if (recordTrack == mRecordTrack.get()) return NO_ERROR;
 
-    if (mRecordTrack != 0) return -EBUSY;
+    if (mActiveTrack != 0) {
+        if (recordTrack != mActiveTrack.get()) return -EBUSY;
 
-    mRecordTrack = recordTrack;
+        if (mActiveTrack->mState == TrackBase::PAUSING) mActiveTrack->mState = TrackBase::RESUMING;
 
+        return NO_ERROR;
+    }
+
+    mActiveTrack = recordTrack;
+    mActiveTrack->mState = TrackBase::RESUMING;
     // signal thread to start
     LOGV("Signal record thread");
     mWaitWorkCV.signal();
-    mWaitWorkCV.wait(mLock);
-    LOGV("Record started, status %d", mStartStatus);
-    return mStartStatus;
+    mStartStopCond.wait(mLock);
+    if (mActiveTrack != 0) {
+        LOGV("Record started OK");
+        return NO_ERROR;
+    } else {
+        LOGV("Record failed to start");
+        return BAD_VALUE;
+    }
 }
 
-void AudioFlinger::AudioRecordThread::stop(MixerThread::RecordTrack* recordTrack) {
-    LOGV("AudioRecordThread::stop");
+void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
+    LOGV("RecordThread::stop");
     AutoMutex lock(&mLock);
-    if (mActive && (recordTrack == mRecordTrack.get())) {
-        mActive = false;
-        mStopped.wait(mLock);
+    if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
+        mActiveTrack->mState = TrackBase::PAUSING;
+        mStartStopCond.wait(mLock);
     }
 }
 
-void AudioFlinger::AudioRecordThread::exit()
-{
-    LOGV("AudioRecordThread::exit");
-    {
-        AutoMutex lock(&mLock);
-        requestExit();
-        mWaitWorkCV.signal();
-    }
-    requestExitAndWait();
-}
-
-status_t AudioFlinger::AudioRecordThread::dump(int fd, const Vector<String16>& args)
+status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
     pid_t pid = 0;
 
-    if (mRecordTrack != 0 && mRecordTrack->mClient != 0) {
-        snprintf(buffer, SIZE, "Record client pid: %d\n", mRecordTrack->mClient->pid());
+    if (mActiveTrack != 0 && mActiveTrack->mClient != 0) {
+        snprintf(buffer, SIZE, "Record client pid: %d\n", mActiveTrack->mClient->pid());
         result.append(buffer);
     } else {
         result.append("No record client\n");
@@ -2533,6 +3039,463 @@
     return NO_ERROR;
 }
 
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    size_t framesReq = buffer->frameCount;
+    size_t framesReady = mFrameCount - mRsmpInIndex;
+    int channelCount;
+
+    if (framesReady == 0) {
+        ssize_t bytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+        if (bytesRead < 0) {
+            LOGE("RecordThread::getNextBuffer() Error reading audio input");
+            sleep(1);
+            buffer->raw = 0;
+            buffer->frameCount = 0;
+            return NOT_ENOUGH_DATA;
+        }
+        mRsmpInIndex = 0;
+        framesReady = mFrameCount;
+    }
+
+    if (framesReq > framesReady) {
+        framesReq = framesReady;
+    }
+
+    if (mChannelCount == 1 && mReqChannelCount == 2) {
+        channelCount = 1;
+    } else {
+        channelCount = 2;
+    }
+    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
+    buffer->frameCount = framesReq;
+    return NO_ERROR;
+}
+
+void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    mRsmpInIndex += buffer->frameCount;
+    buffer->frameCount = 0;
+}
+
+bool AudioFlinger::RecordThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    if (mNewParameters != "") {
+        status_t status = NO_ERROR;
+        AudioParameter param = AudioParameter(mNewParameters);
+        int value;
+        int reqFormat = mFormat;
+        int reqSamplingRate = mReqSampleRate;
+        int reqChannelCount = mReqChannelCount;
+        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+            reqSamplingRate = value;
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+            reqFormat = value;
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+            reqChannelCount = AudioSystem::popCount(value);
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be garantied
+            // if frame count is changed after track creation
+            if (mActiveTrack != 0) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (status == NO_ERROR) {
+            status = mInput->setParameters(mNewParameters);
+            if (status == INVALID_OPERATION) {
+               mInput->standby();
+               status = mInput->setParameters(mNewParameters);
+            }
+            if (reconfig) {
+                if (status == BAD_VALUE &&
+                    reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT &&
+                    ((int)mInput->sampleRate() <= 2 * reqSamplingRate) &&
+                    (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
+                    status = NO_ERROR;
+                }
+                if (status == NO_ERROR) {
+                    readInputParameters();
+                    sendConfigEvent(AudioSystem::INPUT_CONFIG_CHANGED);
+                }
+            }
+        }
+        mNewParameters = "";
+        mParamStatus = status;
+        mParamCond.signal();
+    }
+    return reconfig;
+}
+
+String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+{
+    return mInput->getParameters(keys);
+}
+
+void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) {
+    AudioSystem::OutputDescriptor desc;
+    void *param2 = 0;
+
+    switch (event) {
+    case AudioSystem::INPUT_OPENED:
+    case AudioSystem::INPUT_CONFIG_CHANGED:
+        desc.channels = mChannelCount;
+        desc.samplingRate = mSampleRate;
+        desc.format = mFormat;
+        desc.frameCount = mFrameCount;
+        desc.latency = 0;
+        param2 = &desc;
+        break;
+
+    case AudioSystem::INPUT_CLOSED:
+    default:
+        break;
+    }
+    mAudioFlinger->audioConfigChanged(event, this, param2);
+}
+
+void AudioFlinger::RecordThread::readInputParameters()
+{
+    if (mRsmpInBuffer) delete mRsmpInBuffer;
+    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
+    if (mResampler) delete mResampler;
+    mResampler = 0;
+
+    mSampleRate = mInput->sampleRate();
+    mChannelCount = AudioSystem::popCount(mInput->channels());
+    mFormat = mInput->format();
+    mFrameSize = mInput->frameSize();
+    mInputBytes = mInput->bufferSize();
+    mFrameCount = mInputBytes / mFrameSize;
+    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+
+    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
+    {
+        int channelCount;
+         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+         // stereo to mono post process as the resampler always outputs stereo.
+        if (mChannelCount == 1 && mReqChannelCount == 2) {
+            channelCount = 1;
+        } else {
+            channelCount = 2;
+        }
+        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+        mResampler->setSampleRate(mSampleRate);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
+
+        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
+        if (mChannelCount == 1 && mReqChannelCount == 1) {
+            mFrameCount >>= 1;
+        }
+
+    }
+    mRsmpInIndex = mFrameCount;
+}
+
+// ----------------------------------------------------------------------------
+
+void *AudioFlinger::openOutput(uint32_t *pDevices,
+                                uint32_t *pSamplingRate,
+                                uint32_t *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t *pLatencyMs,
+                                uint32_t flags)
+{
+    status_t status;
+    PlaybackThread *thread = NULL;
+    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+    uint32_t format = pFormat ? *pFormat : 0;
+    uint32_t channels = pChannels ? *pChannels : 0;
+    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
+
+    LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
+            pDevices ? *pDevices : 0,
+            samplingRate,
+            format,
+            channels,
+            flags);
+
+    if (pDevices == NULL || *pDevices == 0) {
+        return NULL;
+    }
+    Mutex::Autolock _l(mLock);
+
+    AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
+                                                             (int *)&format,
+                                                             &channels,
+                                                             &samplingRate,
+                                                             &status);
+    LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
+            output,
+            samplingRate,
+            format,
+            channels,
+            status);
+
+    mHardwareStatus = AUDIO_HW_IDLE;
+    if (output != 0) {
+        if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+            (format != AudioSystem::PCM_16_BIT) ||
+            (channels != AudioSystem::CHANNEL_OUT_STEREO)) {
+            thread = new DirectOutputThread(this, output);
+            LOGV("openOutput() created direct output %p", thread);
+        } else {
+            thread = new MixerThread(this, output);
+            LOGV("openOutput() created mixer output %p", thread);
+        }
+        mPlaybackThreads.add(thread);
+
+        if (pSamplingRate) *pSamplingRate = samplingRate;
+        if (pFormat) *pFormat = format;
+        if (pChannels) *pChannels = channels;
+        if (pLatencyMs) *pLatencyMs = thread->latency();
+    }
+
+    return thread;
+}
+
+void *AudioFlinger::openDuplicateOutput(void *output1, void *output2)
+{
+    Mutex::Autolock _l(mLock);
+
+    if (checkMixerThread_l(output1) == NULL ||
+        checkMixerThread_l(output2) == NULL) {
+        LOGW("openDuplicateOutput() wrong output mixer type %p or %p", output1, output2);
+        return NULL;
+    }
+
+    DuplicatingThread *thread = new DuplicatingThread(this, (MixerThread *)output1);
+    thread->addOutputTrack( (MixerThread *)output2);
+    mPlaybackThreads.add(thread);
+    return thread;
+}
+
+status_t AudioFlinger::closeOutput(void *output)
+{
+    PlaybackThread *thread;
+    {
+        Mutex::Autolock _l(mLock);
+        thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            return BAD_VALUE;
+        }
+
+        LOGV("closeOutput() %p", thread);
+
+        if (thread->type() == PlaybackThread::MIXER) {
+            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+                if (mPlaybackThreads[i]->type() == PlaybackThread::DUPLICATING) {
+                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads[i].get();
+                    dupThread->removeOutputTrack((MixerThread *)thread);
+                }
+            }
+        }
+        mPlaybackThreads.remove(thread);
+    }
+    thread->exit();
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::suspendOutput(void *output)
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+
+    if (thread == NULL) {
+        return BAD_VALUE;
+    }
+
+    LOGV("suspendOutput() %p", output);
+    thread->suspend();
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::restoreOutput(void *output)
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+
+    if (thread == NULL) {
+        return BAD_VALUE;
+    }
+
+    LOGV("restoreOutput() %p", output);
+
+    thread->restore();
+
+    return NO_ERROR;
+}
+
+void *AudioFlinger::openInput(uint32_t *pDevices,
+                                uint32_t *pSamplingRate,
+                                uint32_t *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t acoustics)
+{
+    status_t status;
+    RecordThread *thread = NULL;
+    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+    uint32_t format = pFormat ? *pFormat : 0;
+    uint32_t channels = pChannels ? *pChannels : 0;
+    uint32_t reqSamplingRate = samplingRate;
+    uint32_t reqFormat = format;
+    uint32_t reqChannels = channels;
+
+    if (pDevices == NULL || *pDevices == 0) {
+        return NULL;
+    }
+    Mutex::Autolock _l(mLock);
+
+    AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices,
+                                                             (int *)&format,
+                                                             &channels,
+                                                             &samplingRate,
+                                                             &status,
+                                                             (AudioSystem::audio_in_acoustics)acoustics);
+    LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
+            input,
+            samplingRate,
+            format,
+            channels,
+            acoustics,
+            status);
+
+    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
+    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
+    // or stereo to mono conversions on 16 bit PCM inputs.
+    if (input == 0 && status == BAD_VALUE &&
+        reqFormat == format && format == AudioSystem::PCM_16_BIT &&
+        (samplingRate <= 2 * reqSamplingRate) &&
+        (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) {
+        LOGV("openInput() reopening with proposed sampling rate and channels");
+        input = mAudioHardware->openInputStream(*pDevices,
+                                                 (int *)&format,
+                                                 &channels,
+                                                 &samplingRate,
+                                                 &status,
+                                                 (AudioSystem::audio_in_acoustics)acoustics);
+    }
+
+    if (input != 0) {
+         // Start record thread
+        thread = new RecordThread(this, input, reqSamplingRate, reqChannels);
+        mRecordThreads.add(thread);
+
+        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
+        if (pFormat) *pFormat = format;
+        if (pChannels) *pChannels = reqChannels;
+
+        input->standby();
+    }
+
+    return thread;
+}
+
+status_t AudioFlinger::closeInput(void *input)
+{
+    RecordThread *thread;
+    {
+        Mutex::Autolock _l(mLock);
+        thread = checkRecordThread_l(input);
+        if (thread == NULL) {
+            return BAD_VALUE;
+        }
+
+        LOGV("closeInput() %p", thread);
+        mRecordThreads.remove(thread);
+    }
+    thread->exit();
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::setStreamOutput(uint32_t stream, void *output)
+{
+    Mutex::Autolock _l(mLock);
+    MixerThread *dstThread = checkMixerThread_l(output);
+    if (dstThread == NULL) {
+        LOGW("setStreamOutput() bad output thread %p", output);
+        return BAD_VALUE;
+    }
+
+    LOGV("setStreamOutput() stream %d to output %p", stream, dstThread);
+
+    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+        PlaybackThread *thread = mPlaybackThreads[i].get();
+        if (thread != dstThread &&
+            thread->type() != PlaybackThread::DIRECT) {
+            MixerThread *srcThread = (MixerThread *)thread;
+            SortedVector < sp<MixerThread::Track> > tracks;
+            SortedVector < wp<MixerThread::Track> > activeTracks;
+            srcThread->getTracks(tracks, activeTracks, stream);
+            if (tracks.size()) {
+                dstThread->putTracks(tracks, activeTracks);
+                dstThread->sendConfigEvent(AudioSystem::STREAM_CONFIG_CHANGED, stream);
+            }
+        }
+    }
+
+    return NO_ERROR;
+}
+
+// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(void *output) const
+{
+    PlaybackThread *thread = NULL;
+
+    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+        if (mPlaybackThreads[i] == output) {
+            thread = (PlaybackThread *)output;
+            break;
+        }
+    }
+
+    return thread;
+}
+
+// checkMixerThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(void *output) const
+{
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread != NULL) {
+        if (thread->type() == PlaybackThread::DIRECT) {
+            thread = NULL;
+        }
+    }
+    return (MixerThread *)thread;
+}
+
+// checkRecordThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(void *input) const
+{
+    RecordThread *thread = NULL;
+
+    for (size_t i = 0; i < mRecordThreads.size(); i++) {
+        if (mRecordThreads[i] == input) {
+            thread = (RecordThread *)input;
+            break;
+        }
+    }
+
+    return thread;
+}
+
+// ----------------------------------------------------------------------------
+
 status_t AudioFlinger::onTransact(
         uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
@@ -2540,6 +3503,7 @@
 }
 
 // ----------------------------------------------------------------------------
+
 void AudioFlinger::instantiate() {
     defaultServiceManager()->addService(
             String16("media.audio_flinger"), new AudioFlinger());
diff --git a/libs/audioflinger/AudioFlinger.h b/libs/audioflinger/AudioFlinger.h
index 3531a58..7d78749 100644
--- a/libs/audioflinger/AudioFlinger.h
+++ b/libs/audioflinger/AudioFlinger.h
@@ -31,7 +31,6 @@
 #include <utils/Errors.h>
 #include <utils/threads.h>
 #include <binder/MemoryDealer.h>
-#include <utils/KeyedVector.h>
 #include <utils/SortedVector.h>
 #include <utils/Vector.h>
 
@@ -44,6 +43,7 @@
 class audio_track_cblk_t;
 class AudioMixer;
 class AudioBuffer;
+class AudioResampler;
 
 
 // ----------------------------------------------------------------------------
@@ -56,7 +56,7 @@
 
 static const nsecs_t kStandbyTimeInNsecs = seconds(3);
 
-class AudioFlinger : public BnAudioFlinger, public IBinder::DeathRecipient 
+class AudioFlinger : public BnAudioFlinger, public IBinder::DeathRecipient
 {
 public:
     static void instantiate();
@@ -73,13 +73,14 @@
                                 int frameCount,
                                 uint32_t flags,
                                 const sp<IMemory>& sharedBuffer,
+                                void *output,
                                 status_t *status);
 
-    virtual     uint32_t    sampleRate(int output) const;
-    virtual     int         channelCount(int output) const;
-    virtual     int         format(int output) const;
-    virtual     size_t      frameCount(int output) const;
-    virtual     uint32_t    latency(int output) const;
+    virtual     uint32_t    sampleRate(void *output) const;
+    virtual     int         channelCount(void *output) const;
+    virtual     int         format(void *output) const;
+    virtual     size_t      frameCount(void *output) const;
+    virtual     uint32_t    latency(void *output) const;
 
     virtual     status_t    setMasterVolume(float value);
     virtual     status_t    setMasterMute(bool muted);
@@ -87,33 +88,51 @@
     virtual     float       masterVolume() const;
     virtual     bool        masterMute() const;
 
-    virtual     status_t    setStreamVolume(int stream, float value);
+    virtual     status_t    setStreamVolume(int stream, float value, void *output);
     virtual     status_t    setStreamMute(int stream, bool muted);
 
-    virtual     float       streamVolume(int stream) const;
+    virtual     float       streamVolume(int stream, void *output) const;
     virtual     bool        streamMute(int stream) const;
 
-    virtual     status_t    setRouting(int mode, uint32_t routes, uint32_t mask);
-    virtual     uint32_t    getRouting(int mode) const;
-
     virtual     status_t    setMode(int mode);
-    virtual     int         getMode() const;
 
     virtual     status_t    setMicMute(bool state);
     virtual     bool        getMicMute() const;
 
     virtual     bool        isMusicActive() const;
 
-    virtual     bool        isA2dpEnabled() const;
-
-    virtual     status_t    setParameter(const char* key, const char* value);
+    virtual     status_t    setParameters(void *ioHandle, const String8& keyValuePairs);
+    virtual     String8     getParameters(void *ioHandle, const String8& keys);
 
     virtual     void        registerClient(const sp<IAudioFlingerClient>& client);
-    
+
     virtual     size_t      getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
-    
-    virtual     void        wakeUp()    { mWaitWorkCV.broadcast(); }
-    
+
+    virtual void *openOutput(uint32_t *pDevices,
+                                    uint32_t *pSamplingRate,
+                                    uint32_t *pFormat,
+                                    uint32_t *pChannels,
+                                    uint32_t *pLatencyMs,
+                                    uint32_t flags);
+
+    virtual void *openDuplicateOutput(void *output1, void *output2);
+
+    virtual status_t closeOutput(void *output);
+
+    virtual status_t suspendOutput(void *output);
+
+    virtual status_t restoreOutput(void *output);
+
+    virtual void *openInput(uint32_t *pDevices,
+                            uint32_t *pSamplingRate,
+                            uint32_t *pFormat,
+                            uint32_t *pChannels,
+                            uint32_t acoustics);
+
+    virtual status_t closeInput(void *input);
+
+    virtual status_t setStreamOutput(uint32_t stream, void *output);
+
     // IBinder::DeathRecipient
     virtual     void        binderDied(const wp<IBinder>& who);
 
@@ -139,7 +158,7 @@
     // record interface
     virtual sp<IAudioRecord> openRecord(
                                 pid_t pid,
-                                int inputSource,
+                                void *input,
                                 uint32_t sampleRate,
                                 int format,
                                 int channelCount,
@@ -153,30 +172,12 @@
                                 Parcel* reply,
                                 uint32_t flags);
 
+    void audioConfigChanged(int event, void *param1, void *param2);
+
 private:
                             AudioFlinger();
     virtual                 ~AudioFlinger();
-    
-    void                    setOutput(int outputType);
-    void                    doSetOutput(int outputType);
 
-#ifdef WITH_A2DP
-    void                    setA2dpEnabled_l(bool enable);
-    void                    checkA2dpEnabledChange_l();
-#endif
-    static bool             streamForcedToSpeaker(int streamType);
-    
-    // Management of forced route to speaker for certain track types.
-    enum force_speaker_command {
-        ACTIVE_TRACK_ADDED = 0,
-        ACTIVE_TRACK_REMOVED,
-        CHECK_ROUTE_RESTORE_TIME,
-        FORCE_ROUTE_RESTORE
-    };
-    void                    handleForcedSpeakerRoute(int command);
-#ifdef WITH_A2DP
-    void                    handleRouteDisablesA2dp_l(int routes);
-#endif
 
     // Internal dump utilites.
     status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
@@ -201,14 +202,17 @@
 
     class TrackHandle;
     class RecordHandle;
-    class AudioRecordThread;
+    class RecordThread;
+    class PlaybackThread;
+    class MixerThread;
+    class DirectOutputThread;
+    class Track;
+    class RecordTrack;
 
-    
-    // --- MixerThread ---
-    class MixerThread : public Thread {
+    class ThreadBase : public Thread {
     public:
-        
-        // --- Track ---
+        ThreadBase (const sp<AudioFlinger>& audioFlinger);
+        virtual             ~ThreadBase();
 
         // base for record and playback
         class TrackBase : public AudioBufferProvider, public RefBase {
@@ -230,7 +234,7 @@
                 // The upper 16 bits are used for track-specific flags.
             };
 
-                                TrackBase(const sp<MixerThread>& mixerThread,
+                                TrackBase(const wp<ThreadBase>& thread,
                                         const sp<Client>& client,
                                         uint32_t sampleRate,
                                         int format,
@@ -245,9 +249,8 @@
                     sp<IMemory> getCblk() const;
 
         protected:
-            friend class MixerThread;
+            friend class ThreadBase;
             friend class RecordHandle;
-            friend class AudioRecordThread;
 
                                 TrackBase(const TrackBase&);
                                 TrackBase& operator = (const TrackBase&);
@@ -269,10 +272,6 @@
 
             void* getBuffer(uint32_t offset, uint32_t frames) const;
 
-            int name() const {
-                return mName;
-            }
-
             bool isStopped() const {
                 return mState == STOPPED;
             }
@@ -284,14 +283,13 @@
             bool step();
             void reset();
 
-            sp<MixerThread>     mMixerThread;
+            wp<ThreadBase>      mThread;
             sp<Client>          mClient;
             sp<IMemory>         mCblkMemory;
             audio_track_cblk_t* mCblk;
             void*               mBuffer;
             void*               mBufferEnd;
             uint32_t            mFrameCount;
-            int                 mName;
             // we don't really need a lock for these
             int                 mState;
             int                 mClientTid;
@@ -299,10 +297,67 @@
             uint32_t            mFlags;
         };
 
+        class ConfigEvent {
+        public:
+            ConfigEvent() : mEvent(0), mParam(0) {}
+
+            int mEvent;
+            int mParam;
+        };
+
+                    uint32_t    sampleRate() const;
+                    int         channelCount() const;
+                    int         format() const;
+                    size_t      frameCount() const;
+                    void        wakeUp()    { mWaitWorkCV.broadcast(); }
+                    void        exit();
+        virtual     bool        checkForNewParameters_l() = 0;
+        virtual     status_t    setParameters(const String8& keyValuePairs);
+        virtual     String8     getParameters(const String8& keys) = 0;
+        virtual     void        audioConfigChanged(int event, int param = 0) = 0;
+                    void        sendConfigEvent(int event, int param = 0);
+                    void        processConfigEvents();
+
+    protected:
+
+        friend class Track;
+        friend class TrackBase;
+        friend class PlaybackThread;
+        friend class MixerThread;
+        friend class DirectOutputThread;
+        friend class DuplicatingThread;
+        friend class RecordThread;
+        friend class RecordTrack;
+
+        mutable     Mutex                   mLock;
+                    Condition               mWaitWorkCV;
+                    sp<AudioFlinger>        mAudioFlinger;
+                    uint32_t                mSampleRate;
+                    size_t                  mFrameCount;
+                    int                     mChannelCount;
+                    int                     mFormat;
+                    uint32_t                mFrameSize;
+                    Condition               mParamCond;
+                    String8                 mNewParameters;
+                    status_t                mParamStatus;
+                    Vector<ConfigEvent *>   mConfigEvents;
+                    bool                    mStandby;
+    };
+
+    // --- PlaybackThread ---
+    class PlaybackThread : public ThreadBase {
+    public:
+
+        enum type {
+            MIXER,
+            DIRECT,
+            DUPLICATING
+        };
+
         // playback track
         class Track : public TrackBase {
         public:
-                                Track(  const sp<MixerThread>& mixerThread,
+                                Track(  const wp<ThreadBase>& thread,
                                         const sp<Client>& client,
                                         int streamType,
                                         uint32_t sampleRate,
@@ -321,6 +376,9 @@
                     void        destroy();
                     void        mute(bool);
                     void        setVolume(float left, float right);
+                    int name() const {
+                        return mName;
+                    }
 
                     int type() const {
                         return mStreamType;
@@ -328,7 +386,7 @@
 
 
         protected:
-            friend class MixerThread;
+            friend class ThreadBase;
             friend class AudioFlinger;
             friend class AudioFlinger::TrackHandle;
 
@@ -336,21 +394,14 @@
                                 Track& operator = (const Track&);
 
             virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
-
-            bool isMuted() const {
-                return (mMute || mMixerThread->mStreamTypes[mStreamType].mute);
-            }
-
+            bool isMuted() { return mMute; }
             bool isPausing() const {
                 return mState == PAUSING;
             }
-
             bool isPaused() const {
                 return mState == PAUSED;
             }
-
             bool isReady() const;
-
             void setPaused() { mState = PAUSED; }
             void reset();
 
@@ -364,54 +415,20 @@
             sp<IMemory>         mSharedBuffer;
             bool                mResetDone;
             int                 mStreamType;
+            int                 mName;
         };  // end of Track
 
-        // record track
-        class RecordTrack : public TrackBase {
-        public:
-                                RecordTrack(const sp<MixerThread>& mixerThread,
-                                        const sp<Client>& client,
-                                        int inputSource,
-                                        uint32_t sampleRate,
-                                        int format,
-                                        int channelCount,
-                                        int frameCount,
-                                        uint32_t flags);
-                                ~RecordTrack();
-
-            virtual status_t    start();
-            virtual void        stop();
-
-                    bool        overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
-                    bool        setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
-
-                    int         inputSource() const { return mInputSource; }
-
-        private:
-            friend class AudioFlinger;
-            friend class AudioFlinger::RecordHandle;
-            friend class AudioFlinger::AudioRecordThread;
-            friend class MixerThread;
-
-                                RecordTrack(const Track&);
-                                RecordTrack& operator = (const Track&);
-
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
-
-            bool                mOverflow;
-            int                 mInputSource;
-        };
 
         // playback track
         class OutputTrack : public Track {
         public:
-            
+
             class Buffer: public AudioBufferProvider::Buffer {
             public:
                 int16_t *mBuffer;
             };
-            
-                                OutputTrack(  const sp<MixerThread>& mixerThread,
+
+                                OutputTrack(  const wp<ThreadBase>& thread,
                                         uint32_t sampleRate,
                                         int format,
                                         int channelCount,
@@ -420,35 +437,35 @@
 
             virtual status_t    start();
             virtual void        stop();
-                    void        write(int16_t* data, uint32_t frames);
+                    bool        write(int16_t* data, uint32_t frames);
                     bool        bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; }
+                    bool        isActive() { return mActive; }
+            wp<ThreadBase>&     thread()  { return mThread; }
 
         private:
 
-            status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer);
+            status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
             void                clearBufferQueue();
-            
-            sp<MixerThread>             mOutputMixerThread;
+
+            // Maximum number of pending buffers allocated by OutputTrack::write()
+            static const uint8_t kMaxOverFlowBuffers = 3;
+
             Vector < Buffer* >          mBufferQueue;
             AudioBufferProvider::Buffer mOutBuffer;
-            uint32_t                    mFramesWritten;
-            
-         };  // end of OutputTrack
+            uint32_t                    mWaitTimeMs;
+            bool                        mActive;
 
-        MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int outputType);
-        virtual             ~MixerThread();
+        };  // end of OutputTrack
+
+        PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
+        virtual             ~PlaybackThread();
 
         virtual     status_t    dump(int fd, const Vector<String16>& args);
 
         // Thread virtuals
-        virtual     bool        threadLoop();
         virtual     status_t    readyToRun();
         virtual     void        onFirstRef();
 
-        virtual     uint32_t    sampleRate() const;
-        virtual     int         channelCount() const;
-        virtual     int         format() const;
-        virtual     size_t      frameCount() const;
         virtual     uint32_t    latency() const;
 
         virtual     status_t    setMasterVolume(float value);
@@ -463,9 +480,8 @@
         virtual     float       streamVolume(int stream) const;
         virtual     bool        streamMute(int stream) const;
 
-                    bool        isMusicActive_l() const;
-        
-                    
+                    bool        isMusicActive() const;
+
                     sp<Track>   createTrack_l(
                                     const sp<AudioFlinger::Client>& client,
                                     int streamType,
@@ -475,13 +491,15 @@
                                     int frameCount,
                                     const sp<IMemory>& sharedBuffer,
                                     status_t *status);
-                    
-                    void        getTracks_l(SortedVector < sp<Track> >& tracks,
-                                          SortedVector < wp<Track> >& activeTracks);
-                    void        putTracks_l(SortedVector < sp<Track> >& tracks,
-                                          SortedVector < wp<Track> >& activeTracks);
-                    void        setOuputTrack(OutputTrack *track) { mOutputTrack = track; }
-                    
+
+                    AudioStreamOut* getOutput() { return mOutput; }
+
+        virtual     int         type() const { return mType; }
+                    void        suspend() { mSuspended = true; }
+                    void        restore() { mSuspended = false; }
+        virtual     String8     getParameters(const String8& keys);
+        virtual     void        audioConfigChanged(int event, int param = 0);
+
         struct  stream_type_t {
             stream_type_t()
                 :   volume(1.0f),
@@ -494,54 +512,109 @@
 
     private:
 
-
         friend class AudioFlinger;
         friend class Track;
         friend class TrackBase;
-        friend class RecordTrack;
-        
-        MixerThread(const Client&);
-        MixerThread& operator = (const MixerThread&);
-  
+        friend class MixerThread;
+        friend class DirectOutputThread;
+        friend class DuplicatingThread;
+
+        PlaybackThread(const Client&);
+        PlaybackThread& operator = (const PlaybackThread&);
+
         status_t    addTrack_l(const sp<Track>& track);
         void        destroyTrack_l(const sp<Track>& track);
-        int         getTrackName_l();
-        void        deleteTrackName_l(int name);
-        void        addActiveTrack_l(const wp<Track>& t);
-        void        removeActiveTrack_l(const wp<Track>& t);
-        size_t      getOutputFrameCount();
+        virtual int         getTrackName_l() = 0;
+        virtual void        deleteTrackName_l(int name) = 0;
+        void        readOutputParameters();
 
-        status_t    dumpInternals(int fd, const Vector<String16>& args);
+        virtual status_t    dumpInternals(int fd, const Vector<String16>& args);
         status_t    dumpTracks(int fd, const Vector<String16>& args);
-        
-        sp<AudioFlinger>                mAudioFlinger;       
+
         SortedVector< wp<Track> >       mActiveTracks;
         SortedVector< sp<Track> >       mTracks;
-        stream_type_t                   mStreamTypes[AudioSystem::NUM_STREAM_TYPES];
-        AudioMixer*                     mAudioMixer;
+        // mStreamTypes[] uses 1 additionnal stream type internally for the OutputTrack used by DuplicatingThread
+        stream_type_t                   mStreamTypes[AudioSystem::NUM_STREAM_TYPES + 1];
         AudioStreamOut*                 mOutput;
-        int                             mOutputType;
-        uint32_t                        mSampleRate;
-        size_t                          mFrameCount;
-        int                             mChannelCount;
-        int                             mFormat;
-        int16_t*                        mMixBuffer;
         float                           mMasterVolume;
         bool                            mMasterMute;
         nsecs_t                         mLastWriteTime;
         int                             mNumWrites;
         int                             mNumDelayedWrites;
-        bool                            mStandby;
         bool                            mInWrite;
-        sp <OutputTrack>                mOutputTrack;
+        int16_t*                        mMixBuffer;
+        bool                            mSuspended;
+        int                             mType;
+        int                             mBytesWritten;
+        int                             mMinBytesToWrite;
     };
 
-    
+    class MixerThread : public PlaybackThread {
+    public:
+        MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
+        virtual             ~MixerThread();
+
+        // Thread virtuals
+        virtual     bool        threadLoop();
+
+                    void        getTracks(SortedVector < sp<Track> >& tracks,
+                                      SortedVector < wp<Track> >& activeTracks,
+                                      int streamType);
+                    void        putTracks(SortedVector < sp<Track> >& tracks,
+                                      SortedVector < wp<Track> >& activeTracks);
+        virtual     int         getTrackName_l();
+        virtual     void        deleteTrackName_l(int name);
+        virtual     bool        checkForNewParameters_l();
+        virtual     status_t    dumpInternals(int fd, const Vector<String16>& args);
+
+    protected:
+        size_t prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove);
+
+        AudioMixer*                     mAudioMixer;
+    };
+
+    class DirectOutputThread : public PlaybackThread {
+    public:
+
+        DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output);
+        ~DirectOutputThread();
+
+        // Thread virtuals
+        virtual     bool        threadLoop();
+
+        virtual     int         getTrackName_l();
+        virtual     void        deleteTrackName_l(int name);
+        virtual     bool        checkForNewParameters_l();
+
+    private:
+        float mLeftVolume;
+        float mRightVolume;
+    };
+
+    class DuplicatingThread : public MixerThread {
+    public:
+        DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread);
+        ~DuplicatingThread();
+
+        // Thread virtuals
+        virtual     bool        threadLoop();
+                    void        addOutputTrack(MixerThread* thread);
+                    void        removeOutputTrack(MixerThread* thread);
+
+    private:
+        SortedVector < sp<OutputTrack> >  mOutputTracks;
+    };
+
+              PlaybackThread *checkPlaybackThread_l(void *output) const;
+              MixerThread *checkMixerThread_l(void *output) const;
+              RecordThread *checkRecordThread_l(void *input) const;
+              float streamVolumeInternal(int stream) const { return mStreamTypes[stream].volume; }
+
     friend class AudioBuffer;
 
     class TrackHandle : public android::BnAudioTrack {
     public:
-                            TrackHandle(const sp<MixerThread::Track>& track);
+                            TrackHandle(const sp<PlaybackThread::Track>& track);
         virtual             ~TrackHandle();
         virtual status_t    start();
         virtual void        stop();
@@ -553,20 +626,90 @@
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
     private:
-        sp<MixerThread::Track> mTrack;
+        sp<PlaybackThread::Track> mTrack;
     };
 
     friend class Client;
-    friend class MixerThread::Track;
+    friend class PlaybackThread::Track;
 
 
                 void        removeClient(pid_t pid);
 
 
+    // record thread
+    class RecordThread : public ThreadBase, public AudioBufferProvider
+    {
+    public:
+
+        // record track
+        class RecordTrack : public TrackBase {
+        public:
+                                RecordTrack(const wp<ThreadBase>& thread,
+                                        const sp<Client>& client,
+                                        uint32_t sampleRate,
+                                        int format,
+                                        int channelCount,
+                                        int frameCount,
+                                        uint32_t flags);
+                                ~RecordTrack();
+
+            virtual status_t    start();
+            virtual void        stop();
+
+                    bool        overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
+                    bool        setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
+
+        private:
+            friend class AudioFlinger;
+
+                                RecordTrack(const RecordTrack&);
+                                RecordTrack& operator = (const RecordTrack&);
+
+            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+
+            bool                mOverflow;
+        };
+
+
+                RecordThread(const sp<AudioFlinger>& audioFlinger,
+                        AudioStreamIn *input,
+                        uint32_t sampleRate,
+                        uint32_t channels);
+                ~RecordThread();
+
+        virtual bool        threadLoop();
+        virtual status_t    readyToRun() { return NO_ERROR; }
+        virtual void        onFirstRef();
+
+                status_t    start(RecordTrack* recordTrack);
+                void        stop(RecordTrack* recordTrack);
+                status_t    dump(int fd, const Vector<String16>& args);
+                AudioStreamIn* getInput() { return mInput; }
+
+        virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer);
+        virtual void        releaseBuffer(AudioBufferProvider::Buffer* buffer);
+        virtual bool        checkForNewParameters_l();
+        virtual String8     getParameters(const String8& keys);
+        virtual void        audioConfigChanged(int event, int param = 0);
+                void        readInputParameters();
+
+    private:
+                RecordThread();
+                AudioStreamIn                       *mInput;
+                sp<RecordTrack>                     mActiveTrack;
+                Condition                           mStartStopCond;
+                AudioResampler                      *mResampler;
+                int32_t                             *mRsmpOutBuffer;
+                int16_t                             *mRsmpInBuffer;
+                size_t                              mRsmpInIndex;
+                size_t                              mInputBytes;
+                int                                 mReqChannelCount;
+                uint32_t                            mReqSampleRate;
+    };
 
     class RecordHandle : public android::BnAudioRecord {
     public:
-        RecordHandle(const sp<MixerThread::RecordTrack>& recordTrack);
+        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
         virtual             ~RecordHandle();
         virtual status_t    start();
         virtual void        stop();
@@ -574,66 +717,30 @@
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
     private:
-        sp<MixerThread::RecordTrack> mRecordTrack;
+        sp<RecordThread::RecordTrack> mRecordTrack;
     };
 
-    // record thread
-    class AudioRecordThread : public Thread
-    {
-    public:
-        AudioRecordThread(AudioHardwareInterface* audioHardware, const sp<AudioFlinger>& audioFlinger);
-        virtual             ~AudioRecordThread();
-        virtual bool        threadLoop();
-        virtual status_t    readyToRun() { return NO_ERROR; }
-        virtual void        onFirstRef() {}
+    friend class RecordThread;
+    friend class PlaybackThread;
 
-                status_t    start(MixerThread::RecordTrack* recordTrack);
-                void        stop(MixerThread::RecordTrack* recordTrack);
-                void        exit();
-                status_t    dump(int fd, const Vector<String16>& args);
 
-    private:
-                AudioRecordThread();
-                AudioHardwareInterface              *mAudioHardware;
-                sp<AudioFlinger>                    mAudioFlinger;
-                sp<MixerThread::RecordTrack>        mRecordTrack;
-                Mutex                               mLock;
-                Condition                           mWaitWorkCV;
-                Condition                           mStopped;
-                volatile bool                       mActive;
-                status_t                            mStartStatus;
-    };
-
-    friend class AudioRecordThread;
-    friend class MixerThread;
-
-                status_t    startRecord(MixerThread::RecordTrack* recordTrack);
-                void        stopRecord(MixerThread::RecordTrack* recordTrack);
-
-    mutable     Mutex                               mHardwareLock;
     mutable     Mutex                               mLock;
-    mutable     Condition                           mWaitWorkCV;
 
                 DefaultKeyedVector< pid_t, wp<Client> >     mClients;
 
-                sp<MixerThread>                     mA2dpMixerThread;
-                sp<MixerThread>                     mHardwareMixerThread;
+                mutable     Mutex                   mHardwareLock;
                 AudioHardwareInterface*             mAudioHardware;
-                AudioHardwareInterface*             mA2dpAudioInterface;
-                sp<AudioRecordThread>               mAudioRecordThread;
-                bool                                mA2dpEnabled;
-                bool                                mNotifyA2dpChange;
     mutable     int                                 mHardwareStatus;
-                SortedVector< wp<IBinder> >         mNotificationClients;
-                int                                 mForcedSpeakerCount;
-                int                                 mA2dpDisableCount;
 
-                // true if A2DP should resume when mA2dpDisableCount returns to zero
-                bool                                mA2dpSuppressed;
-                uint32_t                            mSavedRoute;
-                uint32_t                            mForcedRoute;
-                nsecs_t                             mRouteRestoreTime;
-                bool                                mMusicMuteSaved;
+
+                SortedVector< sp<PlaybackThread> >  mPlaybackThreads;
+                PlaybackThread::stream_type_t       mStreamTypes[AudioSystem::NUM_STREAM_TYPES];
+                float                               mMasterVolume;
+                bool                                mMasterMute;
+
+                SortedVector< sp<RecordThread> >    mRecordThreads;
+
+                SortedVector< sp<IBinder> >         mNotificationClients;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/libs/audioflinger/AudioHardwareGeneric.cpp b/libs/audioflinger/AudioHardwareGeneric.cpp
index 1e159b8..57874f3 100644
--- a/libs/audioflinger/AudioHardwareGeneric.cpp
+++ b/libs/audioflinger/AudioHardwareGeneric.cpp
@@ -49,8 +49,8 @@
 AudioHardwareGeneric::~AudioHardwareGeneric()
 {
     if (mFd >= 0) ::close(mFd);
-    delete mOutput;
-    delete mInput;
+    closeOutputStream((AudioStreamOut *)mOutput);
+    closeInputStream((AudioStreamIn *)mInput);
 }
 
 status_t AudioHardwareGeneric::initCheck()
@@ -63,7 +63,7 @@
 }
 
 AudioStreamOut* AudioHardwareGeneric::openOutputStream(
-        int format, int channelCount, uint32_t sampleRate, status_t *status)
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
 {
     AutoMutex lock(mLock);
 
@@ -77,7 +77,7 @@
 
     // create new output stream
     AudioStreamOutGeneric* out = new AudioStreamOutGeneric();
-    status_t lStatus = out->set(this, mFd, format, channelCount, sampleRate);
+    status_t lStatus = out->set(this, mFd, devices, format, channels, sampleRate);
     if (status) {
         *status = lStatus;
     }
@@ -89,17 +89,19 @@
     return mOutput;
 }
 
-void AudioHardwareGeneric::closeOutputStream(AudioStreamOutGeneric* out) {
-    if (out == mOutput) mOutput = 0;
+void AudioHardwareGeneric::closeOutputStream(AudioStreamOut* out) {
+    if (mOutput && out == mOutput) {
+        delete mOutput;
+        mOutput = 0;
+    }
 }
 
 AudioStreamIn* AudioHardwareGeneric::openInputStream(
-        int inputSource, int format, int channelCount, uint32_t sampleRate,
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
         status_t *status, AudioSystem::audio_in_acoustics acoustics)
 {
     // check for valid input source
-    if ((inputSource < AudioRecord::DEFAULT_INPUT) ||
-        (inputSource >= AudioRecord::NUM_INPUT_SOURCES)) {
+    if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
         return 0;
     }
 
@@ -115,7 +117,7 @@
 
     // create new output stream
     AudioStreamInGeneric* in = new AudioStreamInGeneric();
-    status_t lStatus = in->set(this, mFd, format, channelCount, sampleRate, acoustics);
+    status_t lStatus = in->set(this, mFd, devices, format, channels, sampleRate, acoustics);
     if (status) {
         *status = lStatus;
     }
@@ -127,8 +129,11 @@
     return mInput;
 }
 
-void AudioHardwareGeneric::closeInputStream(AudioStreamInGeneric* in) {
-    if (in == mInput) mInput = 0;
+void AudioHardwareGeneric::closeInputStream(AudioStreamIn* in) {
+    if (mInput && in == mInput) {
+        delete mInput;
+        mInput = 0;
+    }
 }
 
 status_t AudioHardwareGeneric::setVoiceVolume(float v)
@@ -185,30 +190,42 @@
 status_t AudioStreamOutGeneric::set(
         AudioHardwareGeneric *hw,
         int fd,
-        int format,
-        int channels,
-        uint32_t rate)
+        uint32_t devices,
+        int *pFormat,
+        uint32_t *pChannels,
+        uint32_t *pRate)
 {
+    int lFormat = pFormat ? *pFormat : 0;
+    uint32_t lChannels = pChannels ? *pChannels : 0;
+    uint32_t lRate = pRate ? *pRate : 0;
+
     // fix up defaults
-    if (format == 0) format = AudioSystem::PCM_16_BIT;
-    if (channels == 0) channels = channelCount();
-    if (rate == 0) rate = sampleRate();
+    if (lFormat == 0) lFormat = format();
+    if (lChannels == 0) lChannels = channels();
+    if (lRate == 0) lRate = sampleRate();
 
     // check values
-    if ((format != AudioSystem::PCM_16_BIT) ||
-            (channels != channelCount()) ||
-            (rate != sampleRate()))
+    if ((lFormat != format()) ||
+            (lChannels != channels()) ||
+            (lRate != sampleRate())) {
+        if (pFormat) *pFormat = format();
+        if (pChannels) *pChannels = channels();
+        if (pRate) *pRate = sampleRate();
         return BAD_VALUE;
+    }
+
+    if (pFormat) *pFormat = lFormat;
+    if (pChannels) *pChannels = lChannels;
+    if (pRate) *pRate = lRate;
 
     mAudioHardware = hw;
     mFd = fd;
+    mDevice = devices;
     return NO_ERROR;
 }
 
 AudioStreamOutGeneric::~AudioStreamOutGeneric()
 {
-    if (mAudioHardware)
-        mAudioHardware->closeOutputStream(this);
 }
 
 ssize_t AudioStreamOutGeneric::write(const void* buffer, size_t bytes)
@@ -234,10 +251,12 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
     result.append(buffer);
-    snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
     result.append(buffer);
     snprintf(buffer, SIZE, "\tformat: %d\n", format());
     result.append(buffer);
+    snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice);
+    result.append(buffer);
     snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware);
     result.append(buffer);
     snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
@@ -246,29 +265,68 @@
     return NO_ERROR;
 }
 
+status_t AudioStreamOutGeneric::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 key = String8(AudioParameter::keyRouting);
+    status_t status = NO_ERROR;
+    int device;
+    LOGV("setParameters() %s", keyValuePairs.string());
+
+    if (param.getInt(key, device) == NO_ERROR) {
+        mDevice = device;
+        param.remove(key);
+    }
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+    return status;
+}
+
+String8 AudioStreamOutGeneric::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8(AudioParameter::keyRouting);
+
+    if (param.get(key, value) == NO_ERROR) {
+        param.addInt(key, (int)mDevice);
+    }
+
+    LOGV("getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
 // ----------------------------------------------------------------------------
 
 // record functions
 status_t AudioStreamInGeneric::set(
         AudioHardwareGeneric *hw,
         int fd,
-        int format,
-        int channels,
-        uint32_t rate,
+        uint32_t devices,
+        int *pFormat,
+        uint32_t *pChannels,
+        uint32_t *pRate,
         AudioSystem::audio_in_acoustics acoustics)
 {
     // FIXME: remove logging
-    LOGD("AudioStreamInGeneric::set(%p, %d, %d, %d, %u)", hw, fd, format, channels, rate);
+    if (pFormat == 0 || pChannels == 0 || pRate == 0) return BAD_VALUE;
+    LOGD("AudioStreamInGeneric::set(%p, %d, %d, %d, %u)", hw, fd, *pFormat, *pChannels, *pRate);
     // check values
-    if ((format != AudioSystem::PCM_16_BIT) ||
-            (channels != channelCount()) ||
-            (rate != sampleRate())) {
+    if ((*pFormat != format()) ||
+        (*pChannels != channels()) ||
+        (*pRate != sampleRate())) {
         LOGE("Error opening input channel");
+        *pFormat = format();
+        *pChannels = channels();
+        *pRate = sampleRate();
         return BAD_VALUE;
     }
 
     mAudioHardware = hw;
     mFd = fd;
+    mDevice = devices;
     return NO_ERROR;
 }
 
@@ -276,14 +334,12 @@
 {
     // FIXME: remove logging
     LOGD("AudioStreamInGeneric destructor");
-    if (mAudioHardware)
-        mAudioHardware->closeInputStream(this);
 }
 
 ssize_t AudioStreamInGeneric::read(void* buffer, ssize_t bytes)
 {
     // FIXME: remove logging
-    LOGD("AudioStreamInGeneric::read(%p, %d) from fd %d", buffer, bytes, mFd);
+    LOGD("AudioStreamInGeneric::read(%p, %d) from fd %d", buffer, (int)bytes, mFd);
     AutoMutex lock(mLock);
     if (mFd < 0) {
         LOGE("Attempt to read from unopened device");
@@ -303,10 +359,12 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
     result.append(buffer);
-    snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
     result.append(buffer);
     snprintf(buffer, SIZE, "\tformat: %d\n", format());
     result.append(buffer);
+    snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice);
+    result.append(buffer);
     snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware);
     result.append(buffer);
     snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
@@ -315,6 +373,39 @@
     return NO_ERROR;
 }
 
+status_t AudioStreamInGeneric::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 key = String8(AudioParameter::keyRouting);
+    status_t status = NO_ERROR;
+    int device;
+    LOGV("setParameters() %s", keyValuePairs.string());
+
+    if (param.getInt(key, device) == NO_ERROR) {
+        mDevice = device;
+        param.remove(key);
+    }
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+    return status;
+}
+
+String8 AudioStreamInGeneric::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8(AudioParameter::keyRouting);
+
+    if (param.get(key, value) == NO_ERROR) {
+        param.addInt(key, (int)mDevice);
+    }
+
+    LOGV("getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
 // ----------------------------------------------------------------------------
 
 }; // namespace android
diff --git a/libs/audioflinger/AudioHardwareGeneric.h b/libs/audioflinger/AudioHardwareGeneric.h
index c89df87..42da413 100644
--- a/libs/audioflinger/AudioHardwareGeneric.h
+++ b/libs/audioflinger/AudioHardwareGeneric.h
@@ -39,24 +39,28 @@
     virtual status_t    set(
             AudioHardwareGeneric *hw,
             int mFd,
-            int format,
-            int channelCount,
-            uint32_t sampleRate);
+            uint32_t devices,
+            int *pFormat,
+            uint32_t *pChannels,
+            uint32_t *pRate);
 
     virtual uint32_t    sampleRate() const { return 44100; }
     virtual size_t      bufferSize() const { return 4096; }
-    virtual int         channelCount() const { return 2; }
+    virtual uint32_t    channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
     virtual int         format() const { return AudioSystem::PCM_16_BIT; }
     virtual uint32_t    latency() const { return 20; }
-    virtual status_t    setVolume(float volume) { return INVALID_OPERATION; }
+    virtual status_t    setVolume(float left, float right) { return INVALID_OPERATION; }
     virtual ssize_t     write(const void* buffer, size_t bytes);
     virtual status_t    standby();
     virtual status_t    dump(int fd, const Vector<String16>& args);
+    virtual status_t    setParameters(const String8& keyValuePairs);
+    virtual String8     getParameters(const String8& keys);
 
 private:
     AudioHardwareGeneric *mAudioHardware;
     Mutex   mLock;
     int     mFd;
+    uint32_t mDevice;
 };
 
 class AudioStreamInGeneric : public AudioStreamIn {
@@ -67,24 +71,28 @@
     virtual status_t    set(
             AudioHardwareGeneric *hw,
             int mFd,
-            int format,
-            int channelCount,
-            uint32_t sampleRate,
+            uint32_t devices,
+            int *pFormat,
+            uint32_t *pChannels,
+            uint32_t *pRate,
             AudioSystem::audio_in_acoustics acoustics);
 
-    uint32_t    sampleRate() const { return 8000; }
+    virtual uint32_t    sampleRate() const { return 8000; }
     virtual size_t      bufferSize() const { return 320; }
-    virtual int         channelCount() const { return 1; }
+    virtual uint32_t    channels() const { return AudioSystem::CHANNEL_IN_MONO; }
     virtual int         format() const { return AudioSystem::PCM_16_BIT; }
     virtual status_t    setGain(float gain) { return INVALID_OPERATION; }
     virtual ssize_t     read(void* buffer, ssize_t bytes);
     virtual status_t    dump(int fd, const Vector<String16>& args);
     virtual status_t    standby() { return NO_ERROR; }
+    virtual status_t    setParameters(const String8& keyValuePairs);
+    virtual String8     getParameters(const String8& keys);
 
 private:
     AudioHardwareGeneric *mAudioHardware;
     Mutex   mLock;
     int     mFd;
+    uint32_t mDevice;
 };
 
 
@@ -101,28 +109,27 @@
     virtual status_t    setMicMute(bool state);
     virtual status_t    getMicMute(bool* state);
 
-    virtual status_t    setParameter(const char* key, const char* value)
-            { return NO_ERROR; }
-
     // create I/O streams
     virtual AudioStreamOut* openOutputStream(
-            int format=0,
-            int channelCount=0,
-            uint32_t sampleRate=0,
+            uint32_t devices,
+            int *format=0,
+            uint32_t *channels=0,
+            uint32_t *sampleRate=0,
             status_t *status=0);
+    virtual    void        closeOutputStream(AudioStreamOut* out);
 
     virtual AudioStreamIn* openInputStream(
-            int inputSource,
-            int format,
-            int channelCount,
-            uint32_t sampleRate,
+            uint32_t devices,
+            int *format,
+            uint32_t *channels,
+            uint32_t *sampleRate,
             status_t *status,
             AudioSystem::audio_in_acoustics acoustics);
+    virtual    void        closeInputStream(AudioStreamIn* in);
 
             void            closeOutputStream(AudioStreamOutGeneric* out);
             void            closeInputStream(AudioStreamInGeneric* in);
 protected:
-    virtual status_t        doRouting() { return NO_ERROR; }
     virtual status_t        dump(int fd, const Vector<String16>& args);
 
 private:
diff --git a/libs/audioflinger/AudioHardwareInterface.cpp b/libs/audioflinger/AudioHardwareInterface.cpp
index cc1bd8f..37be329 100644
--- a/libs/audioflinger/AudioHardwareInterface.cpp
+++ b/libs/audioflinger/AudioHardwareInterface.cpp
@@ -18,6 +18,7 @@
 #include <cutils/properties.h>
 #include <string.h>
 #include <unistd.h>
+//#define LOG_NDEBUG 0
 
 #define LOG_TAG "AudioHardwareInterface"
 #include <utils/Log.h>
@@ -25,15 +26,17 @@
 
 #include "AudioHardwareStub.h"
 #include "AudioHardwareGeneric.h"
+#ifdef WITH_A2DP
+#include "A2dpAudioInterface.h"
+#endif
 
-//#define DUMP_FLINGER_OUT        // if defined allows recording samples in a file
-#ifdef DUMP_FLINGER_OUT
+#ifdef ENABLE_AUDIO_DUMP
 #include "AudioDumpInterface.h"
 #endif
 
 
 // change to 1 to log routing calls
-#define LOG_ROUTING_CALLS 0
+#define LOG_ROUTING_CALLS 1
 
 namespace android {
 
@@ -48,14 +51,6 @@
     "IN_CALL"
 };
 
-static const char* routeStrings[] =
-{
-    "EARPIECE ",
-    "SPEAKER ",
-    "BLUETOOTH ",
-    "HEADSET ",
-    "BLUETOOTH_A2DP "
-};
 static const char* routeNone = "NONE";
 
 static const char* displayMode(int mode)
@@ -64,22 +59,6 @@
         return routingModeStrings[0];
     return routingModeStrings[mode+3];
 }
-
-static const char* displayRoutes(uint32_t routes)
-{
-    static char routeStr[80];
-    if (routes == 0)
-        return routeNone;
-    routeStr[0] = 0;
-    int bitMask = 1;
-    for (int i = 0; i < 4; ++i, bitMask <<= 1) {
-        if (routes & bitMask) {
-            strcat(routeStr, routeStrings[i]);
-        }
-    }
-    routeStr[strlen(routeStr)-1] = 0;
-    return routeStr;
-}
 #endif
 
 // ----------------------------------------------------------------------------
@@ -112,13 +91,17 @@
         hw = new AudioHardwareStub();
     }
     
-#ifdef DUMP_FLINGER_OUT
+#ifdef WITH_A2DP
+    hw = new A2dpAudioInterface(hw);
+#endif
+
+#ifdef ENABLE_AUDIO_DUMP
     // This code adds a record of buffers in a file to write calls made by AudioFlinger.
     // It replaces the current AudioHardwareInterface object by an intermediate one which
     // will record buffers in a file (after sending them to hardware) for testing purpose.
-    // This feature is enabled by defining symbol DUMP_FLINGER_OUT.
-    // The output file is FLINGER_DUMP_NAME. Pause are not recorded in the file.
-    
+    // This feature is enabled by defining symbol ENABLE_AUDIO_DUMP.
+    // The output file is set with setParameters("test_cmd_file_name=<name>"). Pause are not recorded in the file.
+    LOGV("opening PCM dump interface");
     hw = new AudioDumpInterface(hw);    // replace interface
 #endif
     return hw;
@@ -132,48 +115,9 @@
 
 AudioHardwareBase::AudioHardwareBase()
 {
-    // force a routing update on initialization
-    memset(&mRoutes, 0, sizeof(mRoutes));
     mMode = 0;
 }
 
-// generics for audio routing - the real work is done in doRouting
-status_t AudioHardwareBase::setRouting(int mode, uint32_t routes)
-{
-#if LOG_ROUTING_CALLS
-    LOGD("setRouting: mode=%s, routes=[%s]", displayMode(mode), displayRoutes(routes));
-#endif
-    if (mode == AudioSystem::MODE_CURRENT)
-        mode = mMode;
-    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
-        return BAD_VALUE;
-    uint32_t old = mRoutes[mode];
-    mRoutes[mode] = routes;
-    if ((mode != mMode) || (old == routes))
-        return NO_ERROR;
-#if LOG_ROUTING_CALLS
-    const char* oldRouteStr = strdup(displayRoutes(old));
-    LOGD("doRouting: mode=%s, old route=[%s], new route=[%s]",
-           displayMode(mode), oldRouteStr, displayRoutes(routes));
-    delete oldRouteStr;
-#endif
-    return doRouting();
-}
-
-status_t AudioHardwareBase::getRouting(int mode, uint32_t* routes)
-{
-    if (mode == AudioSystem::MODE_CURRENT)
-        mode = mMode;
-    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
-        return BAD_VALUE;
-    *routes = mRoutes[mode];
-#if LOG_ROUTING_CALLS
-    LOGD("getRouting: mode=%s, routes=[%s]",
-           displayMode(mode), displayRoutes(*routes));
-#endif
-    return NO_ERROR;
-}
-
 status_t AudioHardwareBase::setMode(int mode)
 {
 #if LOG_ROUTING_CALLS
@@ -182,28 +126,23 @@
     if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
         return BAD_VALUE;
     if (mMode == mode)
-        return NO_ERROR;
-#if LOG_ROUTING_CALLS
-    LOGD("doRouting: old mode=%s, new mode=%s route=[%s]",
-            displayMode(mMode), displayMode(mode), displayRoutes(mRoutes[mode]));
-#endif
+        return ALREADY_EXISTS;
     mMode = mode;
-    return doRouting();
-}
-
-status_t AudioHardwareBase::getMode(int* mode)
-{
-    // Implement: set audio routing
-    *mode = mMode;
     return NO_ERROR;
 }
 
-status_t AudioHardwareBase::setParameter(const char* key, const char* value)
+// default implementation
+status_t AudioHardwareBase::setParameters(const String8& keyValuePairs)
 {
-    // default implementation is to ignore
     return NO_ERROR;
 }
 
+// default implementation
+String8 AudioHardwareBase::getParameters(const String8& keys)
+{
+    String8 result = String8("");
+    return result;
+}
 
 // default implementation
 size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
@@ -233,10 +172,6 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "\tmMode: %d\n", mMode);
     result.append(buffer);
-    for (int i = 0, n = AudioSystem::NUM_MODES; i < n; ++i) {
-        snprintf(buffer, SIZE, "\tmRoutes[%d]: %d\n", i, mRoutes[i]);
-        result.append(buffer);
-    }
     ::write(fd, result.string(), result.size());
     dump(fd, args);  // Dump the state of the concrete child.
     return NO_ERROR;
diff --git a/libs/audioflinger/AudioHardwareStub.cpp b/libs/audioflinger/AudioHardwareStub.cpp
index 0ab4c60..1a03059 100644
--- a/libs/audioflinger/AudioHardwareStub.cpp
+++ b/libs/audioflinger/AudioHardwareStub.cpp
@@ -43,10 +43,10 @@
 }
 
 AudioStreamOut* AudioHardwareStub::openOutputStream(
-        int format, int channelCount, uint32_t sampleRate, status_t *status)
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
 {
     AudioStreamOutStub* out = new AudioStreamOutStub();
-    status_t lStatus = out->set(format, channelCount, sampleRate);
+    status_t lStatus = out->set(format, channels, sampleRate);
     if (status) {
         *status = lStatus;
     }
@@ -56,18 +56,22 @@
     return 0;
 }
 
+void AudioHardwareStub::closeOutputStream(AudioStreamOut* out)
+{
+    delete out;
+}
+
 AudioStreamIn* AudioHardwareStub::openInputStream(
-        int inputSource, int format, int channelCount, uint32_t sampleRate,
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
         status_t *status, AudioSystem::audio_in_acoustics acoustics)
 {
     // check for valid input source
-    if ((inputSource < AudioRecord::DEFAULT_INPUT) ||
-        (inputSource >= AudioRecord::NUM_INPUT_SOURCES)) {
+    if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
         return 0;
     }
 
     AudioStreamInStub* in = new AudioStreamInStub();
-    status_t lStatus = in->set(format, channelCount, sampleRate, acoustics);
+    status_t lStatus = in->set(format, channels, sampleRate, acoustics);
     if (status) {
         *status = lStatus;
     }
@@ -77,6 +81,11 @@
     return 0;
 }
 
+void AudioHardwareStub::closeInputStream(AudioStreamIn* in)
+{
+    delete in;
+}
+
 status_t AudioHardwareStub::setVoiceVolume(float volume)
 {
     return NO_ERROR;
@@ -107,24 +116,19 @@
 
 // ----------------------------------------------------------------------------
 
-status_t AudioStreamOutStub::set(int format, int channels, uint32_t rate)
+status_t AudioStreamOutStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate)
 {
-    // fix up defaults
-    if (format == 0) format = AudioSystem::PCM_16_BIT;
-    if (channels == 0) channels = channelCount();
-    if (rate == 0) rate = sampleRate();
+    if (pFormat) *pFormat = format();
+    if (pChannels) *pChannels = channels();
+    if (pRate) *pRate = sampleRate();
 
-    if ((format == AudioSystem::PCM_16_BIT) &&
-            (channels == channelCount()) &&
-            (rate == sampleRate()))
-        return NO_ERROR;
-    return BAD_VALUE;
+    return NO_ERROR;
 }
 
 ssize_t AudioStreamOutStub::write(const void* buffer, size_t bytes)
 {
     // fake timing for audio output
-    usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
+    usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
     return bytes;
 }
 
@@ -141,7 +145,7 @@
     snprintf(buffer, SIZE, "AudioStreamOutStub::dump\n");
     snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
     snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
-    snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
     snprintf(buffer, SIZE, "\tformat: %d\n", format());
     result.append(buffer);
     ::write(fd, result.string(), result.size());
@@ -150,20 +154,16 @@
 
 // ----------------------------------------------------------------------------
 
-status_t AudioStreamInStub::set(int format, int channels, uint32_t rate,
+status_t AudioStreamInStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate,
 				AudioSystem::audio_in_acoustics acoustics)
 {
-    if ((format == AudioSystem::PCM_16_BIT) &&
-            (channels == channelCount()) &&
-            (rate == sampleRate()))
-        return NO_ERROR;
-    return BAD_VALUE;
+    return NO_ERROR;
 }
 
 ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
 {
     // fake timing for audio input
-    usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
+    usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
     memset(buffer, 0, bytes);
     return bytes;
 }
@@ -179,7 +179,7 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
     result.append(buffer);
-    snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
     result.append(buffer);
     snprintf(buffer, SIZE, "\tformat: %d\n", format());
     result.append(buffer);
diff --git a/libs/audioflinger/AudioHardwareStub.h b/libs/audioflinger/AudioHardwareStub.h
index bf63cc5..8f43259 100644
--- a/libs/audioflinger/AudioHardwareStub.h
+++ b/libs/audioflinger/AudioHardwareStub.h
@@ -29,29 +29,33 @@
 
 class AudioStreamOutStub : public AudioStreamOut {
 public:
-    virtual status_t    set(int format, int channelCount, uint32_t sampleRate);
+    virtual status_t    set(int *pFormat, uint32_t *pChannels, uint32_t *pRate);
     virtual uint32_t    sampleRate() const { return 44100; }
     virtual size_t      bufferSize() const { return 4096; }
-    virtual int         channelCount() const { return 2; }
+    virtual uint32_t    channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
     virtual int         format() const { return AudioSystem::PCM_16_BIT; }
     virtual uint32_t    latency() const { return 0; }
-    virtual status_t    setVolume(float volume) { return NO_ERROR; }
+    virtual status_t    setVolume(float left, float right) { return NO_ERROR; }
     virtual ssize_t     write(const void* buffer, size_t bytes);
     virtual status_t    standby();
     virtual status_t    dump(int fd, const Vector<String16>& args);
+    virtual status_t    setParameters(const String8& keyValuePairs) { return NO_ERROR;}
+    virtual String8     getParameters(const String8& keys) {String8 result = String8(""); return result;}
 };
 
 class AudioStreamInStub : public AudioStreamIn {
 public:
-    virtual status_t    set(int format, int channelCount, uint32_t sampleRate, AudioSystem::audio_in_acoustics acoustics);
+    virtual status_t    set(int *pFormat, uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics);
     virtual uint32_t    sampleRate() const { return 8000; }
     virtual size_t      bufferSize() const { return 320; }
-    virtual int         channelCount() const { return 1; }
+    virtual uint32_t    channels() const { return AudioSystem::CHANNEL_IN_MONO; }
     virtual int         format() const { return AudioSystem::PCM_16_BIT; }
     virtual status_t    setGain(float gain) { return NO_ERROR; }
     virtual ssize_t     read(void* buffer, ssize_t bytes);
     virtual status_t    dump(int fd, const Vector<String16>& args);
     virtual status_t    standby() { return NO_ERROR; }
+    virtual status_t    setParameters(const String8& keyValuePairs) { return NO_ERROR;}
+    virtual String8     getParameters(const String8& keys) {String8 result = String8(""); return result;}
 };
 
 class AudioHardwareStub : public  AudioHardwareBase
@@ -67,26 +71,25 @@
     virtual status_t    setMicMute(bool state) { mMicMute = state;  return  NO_ERROR; }
     virtual status_t    getMicMute(bool* state) { *state = mMicMute ; return NO_ERROR; }
 
-    virtual status_t    setParameter(const char* key, const char* value)
-            { return NO_ERROR; }
-
     // create I/O streams
     virtual AudioStreamOut* openOutputStream(
-                                int format=0,
-                                int channelCount=0,
-                                uint32_t sampleRate=0,
+                                uint32_t devices,
+                                int *format=0,
+                                uint32_t *channels=0,
+                                uint32_t *sampleRate=0,
                                 status_t *status=0);
+    virtual    void        closeOutputStream(AudioStreamOut* out);
 
     virtual AudioStreamIn* openInputStream(
-                                int inputSource,
-                                int format,
-                                int channelCount,
-                                uint32_t sampleRate,
+                                uint32_t devices,
+                                int *format,
+                                uint32_t *channels,
+                                uint32_t *sampleRate,
                                 status_t *status,
-				AudioSystem::audio_in_acoustics acoustics);
+                                AudioSystem::audio_in_acoustics acoustics);
+    virtual    void        closeInputStream(AudioStreamIn* in);
 
 protected:
-    virtual status_t    doRouting() { return NO_ERROR; }
     virtual status_t    dump(int fd, const Vector<String16>& args);
 
             bool        mMicMute;
diff --git a/libs/audioflinger/AudioMixer.cpp b/libs/audioflinger/AudioMixer.cpp
index b02efcc..19a442a 100644
--- a/libs/audioflinger/AudioMixer.cpp
+++ b/libs/audioflinger/AudioMixer.cpp
@@ -610,7 +610,6 @@
     t->in = in;
 }
 
-inline
 void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
 {
     for (size_t i=0 ; i<c ; i++) {
diff --git a/libs/audioflinger/AudioMixer.h b/libs/audioflinger/AudioMixer.h
index 72ca28a..15766cd 100644
--- a/libs/audioflinger/AudioMixer.h
+++ b/libs/audioflinger/AudioMixer.h
@@ -85,6 +85,8 @@
 
     uint32_t    trackNames() const { return mTrackNames; }
 
+    static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
+
 private:
 
     enum {
@@ -176,7 +178,6 @@
     static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp);
     static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
     static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp);
-    static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
 
     static void process__validate(state_t* state, void* output);
     static void process__nop(state_t* state, void* output);
diff --git a/libs/audioflinger/AudioPolicyManagerGeneric.cpp b/libs/audioflinger/AudioPolicyManagerGeneric.cpp
new file mode 100644
index 0000000..cf9ab88
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyManagerGeneric.cpp
@@ -0,0 +1,764 @@
+/*
+ * 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 "AudioPolicyManagerGeneric"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+#include "AudioPolicyManagerGeneric.h"
+#include <media/mediarecorder.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+
+status_t AudioPolicyManagerGeneric::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)) {
+        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;
+            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;
+            break;
+
+        default:
+            LOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+        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;
+        }
+        return NO_ERROR;
+    }
+
+    LOGW("setDeviceConnectionState() invalid device: %x", device);
+    return BAD_VALUE;
+}
+
+AudioSystem::device_connection_state AudioPolicyManagerGeneric::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) {
+            state = AudioSystem::DEVICE_STATE_AVAILABLE;
+        }
+    } else if (AudioSystem::isInputDevice(device)) {
+        if (device & mAvailableInputDevices) {
+            state = AudioSystem::DEVICE_STATE_AVAILABLE;
+        }
+    }
+
+    return state;
+}
+
+void AudioPolicyManagerGeneric::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;
+    }
+    // store previous phone state for management of sonification strategy below
+    int oldState = mPhoneState;
+    mPhoneState = state;
+
+    // if leaving or entering in call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (state == AudioSystem::MODE_IN_CALL ||
+        oldState == AudioSystem::MODE_IN_CALL) {
+        bool starting = (state == AudioSystem::MODE_IN_CALL) ? true : false;
+        LOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, starting);
+        }
+    }
+}
+
+void AudioPolicyManagerGeneric::setRingerMode(uint32_t mode, uint32_t mask)
+{
+    LOGV("setRingerMode() mode %x, mask %x", mode, mask);
+
+    mRingerMode = mode;
+}
+
+void AudioPolicyManagerGeneric::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+{
+    LOGV("setForceUse) usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+    mForceUse[usage] = config;
+}
+
+AudioSystem::forced_config AudioPolicyManagerGeneric::getForceUse(AudioSystem::force_use usage)
+{
+    return mForceUse[usage];
+}
+
+void AudioPolicyManagerGeneric::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 AudioPolicyManagerGeneric::getOutput(AudioSystem::stream_type stream,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    AudioSystem::output_flags flags)
+{
+    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, channelcount %d, mDirectOutput %d",
+                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannelcount, 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 = (mTestChannelcount == 1) ? AudioSystem::CHANNEL_OUT_MONO : AudioSystem::CHANNEL_OUT_STEREO;
+            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);
+            mOutputs.add(mTestOutputs[mCurOutput], outputDesc);
+        }
+        return mTestOutputs[mCurOutput];
+    }
+#endif //AUDIO_POLICY_TEST
+    if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+        (format != 0 && !AudioSystem::isLinearPCM(format)) ||
+        (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO && channels != AudioSystem::CHANNEL_OUT_STEREO)) {
+        return NULL;
+    }
+
+    return mHardwareOutput;
+}
+
+status_t AudioPolicyManagerGeneric::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    LOGV("startOutput() output %p, stream %d", output, stream);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("startOutput() unknow output %p", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+    // handle special case for sonification while in call
+    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        handleIncallSonification(stream, true);
+    }
+
+    // incremenent usage count for this stream on the requested output:
+    outputDesc->changeRefCount(stream, 1);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerGeneric::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    LOGV("stopOutput() output %p, stream %d", output, stream);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("stopOutput() unknow output %p", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+    // handle special case for sonification while in call
+    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        handleIncallSonification(stream, false);
+    }
+
+    if (outputDesc->isUsedByStream(stream)) {
+        // decrement usage count of this stream on the output
+        outputDesc->changeRefCount(stream, -1);
+        return NO_ERROR;
+    } else {
+        LOGW("stopOutput() refcount is already 0 for output %p", output);
+        return INVALID_OPERATION;
+    }
+}
+
+void AudioPolicyManagerGeneric::releaseOutput(audio_io_handle_t output)
+{
+    LOGV("releaseOutput() %p", output);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("releaseOutput() releasing unknown output %p", 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;
+        }
+    }
+#endif //AUDIO_POLICY_TEST
+}
+
+audio_io_handle_t AudioPolicyManagerGeneric::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;
+
+    LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
+
+    AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
+    inputDesc->mDevice = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+    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 ((samplingRate != inputDesc->mSamplingRate) ||
+        (format != inputDesc->mFormat) ||
+        (channels != inputDesc->mChannels)) {
+        LOGV("getOutput() failed opening input: samplingRate %d, format %d, channels %d",
+                samplingRate, format, channels);
+        mpClientInterface->closeInput(input);
+        delete inputDesc;
+        return NULL;
+    }
+    mInputs.add(input, inputDesc);
+    return input;
+}
+
+status_t AudioPolicyManagerGeneric::startInput(audio_io_handle_t input)
+{
+    LOGV("startInput() input %p", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("startInput() unknow input %p", 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
+        for (size_t i = 0; i < mInputs.size(); i++) {
+            if (mInputs.valueAt(i)->mRefCount > 0) {
+                LOGW("startInput() input %p, other input %p already started", input, mInputs.keyAt(i));
+                return INVALID_OPERATION;
+            }
+        }
+    }
+
+    inputDesc->mRefCount = 1;
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerGeneric::stopInput(audio_io_handle_t input)
+{
+    LOGV("stopInput() input %p", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("stopInput() unknow input %p", input);
+        return BAD_VALUE;
+    }
+    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+    if (inputDesc->mRefCount == 0) {
+        LOGW("stopInput() input %p already stopped", input);
+        return INVALID_OPERATION;
+    } else {
+        inputDesc->mRefCount = 0;
+        return NO_ERROR;
+    }
+}
+
+void AudioPolicyManagerGeneric::releaseInput(audio_io_handle_t input)
+{
+    LOGV("releaseInput() %p", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("releaseInput() releasing unknown input %p", input);
+        return;
+    }
+    mpClientInterface->closeInput(input);
+    delete mInputs.valueAt(index);
+    mInputs.removeItem(input);
+}
+
+
+
+void AudioPolicyManagerGeneric::initStreamVolume(AudioSystem::stream_type stream,
+                                            int indexMin,
+                                            int indexMax)
+{
+    LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+    mStreams[stream].mIndexMin = indexMin;
+    mStreams[stream].mIndexMax = indexMax;
+}
+
+status_t AudioPolicyManagerGeneric::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
+{
+
+    if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+        return BAD_VALUE;
+    }
+
+    LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
+    mStreams[stream].mIndexCur = index;
+
+    // do not change actual stream volume if the stream is muted
+    if (mStreams[stream].mMuteCount != 0) {
+        return NO_ERROR;
+    }
+
+    // Do not changed 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("setStreamVolumeIndex() cannot set stream %d volume with force use = %d for comm",
+             stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
+        return INVALID_OPERATION;
+    }
+
+    // compute and apply stream volume on all outputs according to connected device
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+        uint32_t device = outputDesc->device();
+
+        float volume = computeVolume((int)stream, index, device);
+
+        LOGV("setStreamVolume() for output %p stream %d, volume %f", mOutputs.keyAt(i), stream, volume);
+        mpClientInterface->setStreamVolume(stream, volume, mOutputs.keyAt(i));
+    }
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerGeneric::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;
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManagerGeneric
+// ----------------------------------------------------------------------------
+
+// ---  class factory
+
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+{
+    return new AudioPolicyManagerGeneric(clientInterface);
+}
+
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+{
+    delete interface;
+}
+
+AudioPolicyManagerGeneric::AudioPolicyManagerGeneric(AudioPolicyClientInterface *clientInterface)
+    :
+#ifdef AUDIO_POLICY_TEST
+    Thread(false),
+#endif //AUDIO_POLICY_TEST
+    mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0)
+{
+    mpClientInterface = clientInterface;
+
+    for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
+        mForceUse[i] = AudioSystem::FORCE_NONE;
+    }
+
+    // devices available by default are speaker, ear piece and microphone
+    mAvailableOutputDevices = AudioSystem::DEVICE_OUT_SPEAKER;
+    mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+
+    // 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 {
+        mOutputs.add(mHardwareOutput, outputDesc);
+    }
+
+#ifdef AUDIO_POLICY_TEST
+    mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
+    mTestSamplingRate = 44100;
+    mTestFormat = AudioSystem::PCM_16_BIT;
+    mTestChannelcount = 2;
+    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
+}
+
+AudioPolicyManagerGeneric::~AudioPolicyManagerGeneric()
+{
+#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();
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManagerGeneric::threadLoop()
+{
+    LOGV("entering threadLoop()");
+    while (!exitPending())
+    {
+        Mutex::Autolock _l(mLock);
+        mWaitWorkCV.waitRelative(mLock, milliseconds(50));
+        String8 command;
+        command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
+        if (command != "") {
+            LOGV("Test command %s received", command.string());
+            AudioParameter param = AudioParameter(command);
+            int valueInt;
+            String8 value;
+            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"));
+                if (value == "PCM 16 bits") {
+                    mTestFormat = AudioSystem::PCM_16_BIT;
+                } else if (value == "PCM 8 bits") {
+                    mTestFormat = AudioSystem::PCM_8_BIT;
+                } else if (value == "Compressed MP3") {
+                    mTestFormat = AudioSystem::MP3;
+                }
+            }
+            if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_channels"));
+                if (value == "Channels Stereo") {
+                    mTestChannelcount = 2;
+                } else if (value == "Channels Mono") {
+                    mTestChannelcount = 1;
+                }
+            }
+            if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
+                param.remove(String8("test_cmd_policy_sampleRate"));
+                if (valueInt >= 0 && valueInt <= 96000) {
+                    mTestSamplingRate = valueInt;
+                }
+            }
+            mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+        }
+    }
+    return false;
+}
+
+void AudioPolicyManagerGeneric::exit()
+{
+    {
+        AutoMutex _l(mLock);
+        requestExit();
+        mWaitWorkCV.signal();
+    }
+    requestExitAndWait();
+}
+
+int AudioPolicyManagerGeneric::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
+
+// ---
+
+AudioPolicyManagerGeneric::routing_strategy AudioPolicyManagerGeneric::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:
+    case AudioSystem::ENFORCED_AUDIBLE:
+        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;
+    }
+}
+
+
+float AudioPolicyManagerGeneric::computeVolume(int stream, int index, uint32_t device)
+{
+    float volume = 1.0;
+
+    StreamDescriptor &streamDesc = mStreams[stream];
+
+    // Force max volume if stream cannot be muted
+    if (!streamDesc.mCanBeMuted) index = streamDesc.mIndexMax;
+
+    int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
+    volume = AudioSystem::linearToLog(volInt);
+
+    return volume;
+}
+
+void AudioPolicyManagerGeneric::setStreamMute(int stream, bool on, audio_io_handle_t output)
+{
+    LOGV("setStreamMute() stream %d, mute %d, output %p", stream, on, output);
+
+    StreamDescriptor &streamDesc = mStreams[stream];
+
+    if (on) {
+        if (streamDesc.mMuteCount++ == 0) {
+            if (streamDesc.mCanBeMuted) {
+                mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, 0, output);
+            }
+        }
+    } else {
+        if (streamDesc.mMuteCount == 0) {
+            LOGW("setStreamMute() unmuting non muted stream!");
+            return;
+        }
+        if (--streamDesc.mMuteCount == 0) {
+            uint32_t device = mOutputs.valueFor(output)->mDevice;
+            float volume = computeVolume(stream, streamDesc.mIndexCur, device);
+            mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output);
+        }
+    }
+}
+
+void AudioPolicyManagerGeneric::handleIncallSonification(int stream, bool starting)
+{
+    // 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 (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
+        LOGV("handleIncallSonification() stream %d starting %d device %x", stream, starting, outputDesc->mDevice);
+        if (outputDesc->isUsedByStream((AudioSystem::stream_type)stream)) {
+            if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
+                LOGV("handleIncallSonification() low visibility");
+                setStreamMute(stream, starting, mHardwareOutput);
+            } else {
+                if (starting) {
+                    mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
+                } else {
+                    mpClientInterface->stopTone();
+                }
+            }
+        }
+    }
+}
+
+
+// --- AudioOutputDescriptor class implementation
+
+AudioPolicyManagerGeneric::AudioOutputDescriptor::AudioOutputDescriptor()
+    : mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
+    mFlags((AudioSystem::output_flags)0), mDevice(0)
+{
+    // clear usage count for all stream types
+    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        mRefCount[i] = 0;
+    }
+}
+
+uint32_t AudioPolicyManagerGeneric::AudioOutputDescriptor::device()
+{
+    return mDevice;
+}
+
+void AudioPolicyManagerGeneric::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int 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 AudioPolicyManagerGeneric::AudioOutputDescriptor::refCount()
+{
+    uint32_t refcount = 0;
+    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+        refcount += mRefCount[i];
+    }
+    return refcount;
+}
+
+// --- AudioInputDescriptor class implementation
+
+AudioPolicyManagerGeneric::AudioInputDescriptor::AudioInputDescriptor()
+    : mSamplingRate(0), mFormat(0), mChannels(0),
+     mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
+{
+}
+
+}; // namespace android
diff --git a/libs/audioflinger/AudioPolicyManagerGeneric.h b/libs/audioflinger/AudioPolicyManagerGeneric.h
new file mode 100644
index 0000000..ddcb306
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyManagerGeneric.h
@@ -0,0 +1,189 @@
+/*
+ * 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/Errors.h>
+#include <utils/KeyedVector.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+#include <utils/threads.h>
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+#define MAX_DEVICE_ADDRESS_LEN 20
+#define NUM_TEST_OUTPUTS 5
+
+class AudioPolicyManagerGeneric: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+    , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+                AudioPolicyManagerGeneric(AudioPolicyClientInterface *clientInterface);
+        virtual ~AudioPolicyManagerGeneric();
+
+        // 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 audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+                                            uint32_t samplingRate,
+                                            uint32_t format,
+                                            uint32_t channels,
+                                            AudioSystem::output_flags flags);
+        virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+        virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+        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);
+
+private:
+
+        enum routing_strategy {
+            STRATEGY_MEDIA,
+            STRATEGY_PHONE,
+            STRATEGY_SONIFICATION,
+            STRATEGY_DTMF,
+            NUM_STRATEGIES
+        };
+
+        // 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();
+
+
+            uint32_t device();
+            void changeRefCount(AudioSystem::stream_type, int delta);
+            bool isUsedByStream(AudioSystem::stream_type stream) { return mRefCount[stream] > 0 ? true : false; }
+            uint32_t refCount();
+
+            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
+        };
+
+        // 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();
+
+            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
+        };
+
+        // stream descriptor used for volume control
+        class StreamDescriptor
+        {
+        public:
+            StreamDescriptor()
+            :   mIndexMin(0), mIndexMax(1), mIndexCur(1), mMuteCount(0), mCanBeMuted(true) {}
+
+            int mIndexMin;      // min volume index
+            int mIndexMax;      // max volume index
+            int mIndexCur;      // current volume index
+            int mMuteCount;     // mute request counter
+            bool mCanBeMuted;   // true is the stream can be muted
+        };
+
+        // return the strategy corresponding to a given stream type
+        static routing_strategy getStrategy(AudioSystem::stream_type stream);
+        // return the output handle of an output routed to the specified device, 0 if no output
+        // is routed to the device
+        float computeVolume(int stream, int index, uint32_t device);
+        // Mute or unmute the stream on the specified output
+        void setStreamMute(int stream, bool on, audio_io_handle_t output);
+        // 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);
+
+
+#ifdef AUDIO_POLICY_TEST
+        virtual     bool        threadLoop();
+                    void        exit();
+        int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+
+        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
+        audio_io_handle_t mHardwareOutput;              // hardware output handler
+
+        KeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;   // list ot output descritors
+        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
+
+#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        mTestChannelcount;
+        uint32_t        mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+};
+
+};
diff --git a/libs/audioflinger/AudioPolicyService.cpp b/libs/audioflinger/AudioPolicyService.cpp
new file mode 100644
index 0000000..4810a44
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyService.cpp
@@ -0,0 +1,677 @@
+/*
+ * 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 "AudioPolicyService"
+//#define LOG_NDEBUG 0
+#include <binder/IServiceManager.h>
+#include <utils/Log.h>
+#include <cutils/properties.h>
+#include <binder/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+#include "AudioPolicyService.h"
+#include <cutils/properties.h>
+#include <dlfcn.h>
+
+// ----------------------------------------------------------------------------
+// the sim build doesn't have gettid
+
+#ifndef HAVE_GETTID
+# define gettid getpid
+#endif
+
+namespace android {
+
+const char *AudioPolicyService::sAudioPolicyLibrary = "/system/lib/libaudiopolicy.so";
+const char *AudioPolicyService::sAudioPolicyGenericLibrary = "/system/lib/libaudiopolicygeneric.so";
+
+static bool checkPermission() {
+#ifndef HAVE_ANDROID_OS
+    return true;
+#endif
+    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
+    if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+    return ok;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioPolicyService::AudioPolicyService()
+    : BnAudioPolicyService() , mpPolicyManager(NULL), mpPolicyManagerLibHandle(NULL)
+{
+    const char *audioPolicyLibrary;
+    char value[PROPERTY_VALUE_MAX];
+
+#if (defined GENERIC_AUDIO) || (defined AUDIO_POLICY_TEST)
+    audioPolicyLibrary = sAudioPolicyGenericLibrary;
+    LOGV("build for GENERIC_AUDIO - using generic audio policy");
+#else
+    // if running in emulation - use the emulator driver
+    if (property_get("ro.kernel.qemu", value, 0)) {
+        LOGV("Running in emulation - using generic audio policy");
+        audioPolicyLibrary = sAudioPolicyGenericLibrary;
+    }
+    else {
+        LOGV("Using hardware specific audio policy");
+        audioPolicyLibrary = sAudioPolicyLibrary;
+    }
+#endif
+
+
+    mpPolicyManagerLibHandle = dlopen(audioPolicyLibrary, RTLD_NOW | RTLD_LOCAL);
+    if (mpPolicyManagerLibHandle == NULL) {
+       LOGW("Could not load libaudio policy library");
+       return;
+    }
+
+    AudioPolicyInterface *(*createManager)(AudioPolicyClientInterface *) =
+            reinterpret_cast<AudioPolicyInterface* (*)(AudioPolicyClientInterface *)>(dlsym(mpPolicyManagerLibHandle, "createAudioPolicyManager"));
+
+    if (createManager == NULL ) {
+        LOGW("Could not get createAudioPolicyManager method");
+        return;
+    }
+
+    // start tone playback thread
+    mTonePlaybacThread = new AudioCommandThread();
+    // start audio commands thread
+    mAudioCommandThread = new AudioCommandThread();
+
+    mpPolicyManager = (*createManager)(this);
+
+    // load properties
+    property_get("ro.camera.sound.forced", value, "0");
+    mpPolicyManager->setSystemProperty("ro.camera.sound.forced", value);
+}
+
+AudioPolicyService::~AudioPolicyService()
+{
+    mTonePlaybacThread->exit();
+    mTonePlaybacThread.clear();
+    mAudioCommandThread->exit();
+    mAudioCommandThread.clear();
+
+    if (mpPolicyManager) {
+        void(*destroyManager)(AudioPolicyInterface *) =
+                reinterpret_cast<void(*)(AudioPolicyInterface *)>(dlsym(mpPolicyManagerLibHandle, "destroyAudioPolicyManager"));
+
+        if (destroyManager == NULL ) {
+            LOGW("Could not get destroyAudioPolicyManager method");
+            return;
+        }
+        (*destroyManager)(mpPolicyManager);
+    }
+}
+
+
+status_t AudioPolicyService::setDeviceConnectionState(AudioSystem::audio_devices device,
+                                                  AudioSystem::device_connection_state state,
+                                                  const char *device_address)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!checkPermission()) {
+        return PERMISSION_DENIED;
+    }
+    if (!AudioSystem::isOutputDevice(device) && !AudioSystem::isInputDevice(device)) {
+        return BAD_VALUE;
+    }
+    if (state != AudioSystem::DEVICE_STATE_AVAILABLE && state != AudioSystem::DEVICE_STATE_UNAVAILABLE) {
+        return BAD_VALUE;
+    }
+
+    LOGV("setDeviceConnectionState() tid %d", gettid());
+    Mutex::Autolock _l(mLock);
+    return mpPolicyManager->setDeviceConnectionState(device, state, device_address);
+}
+
+AudioSystem::device_connection_state AudioPolicyService::getDeviceConnectionState(AudioSystem::audio_devices device,
+                                                  const char *device_address)
+{
+    if (mpPolicyManager == NULL) {
+        return AudioSystem::DEVICE_STATE_UNAVAILABLE;
+    }
+    if (!checkPermission()) {
+        return AudioSystem::DEVICE_STATE_UNAVAILABLE;
+    }
+    return mpPolicyManager->getDeviceConnectionState(device, device_address);
+}
+
+status_t AudioPolicyService::setPhoneState(int state)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!checkPermission()) {
+        return PERMISSION_DENIED;
+    }
+    if (state < 0 || state >= AudioSystem::NUM_MODES) {
+        return BAD_VALUE;
+    }
+
+    LOGV("setPhoneState() tid %d", gettid());
+
+    // TODO: check if it is more appropriate to do it in platform specific policy manager
+    AudioSystem::setMode(state);
+
+    Mutex::Autolock _l(mLock);
+    mpPolicyManager->setPhoneState(state);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyService::setRingerMode(uint32_t mode, uint32_t mask)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!checkPermission()) {
+        return PERMISSION_DENIED;
+    }
+
+    mpPolicyManager->setRingerMode(mode, mask);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyService::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!checkPermission()) {
+        return PERMISSION_DENIED;
+    }
+    if (usage < 0 || usage >= AudioSystem::NUM_FORCE_USE) {
+        return BAD_VALUE;
+    }
+    if (config < 0 || config >= AudioSystem::NUM_FORCE_CONFIG) {
+        return BAD_VALUE;
+    }
+    LOGV("setForceUse() tid %d", gettid());
+    Mutex::Autolock _l(mLock);
+    mpPolicyManager->setForceUse(usage, config);
+    return NO_ERROR;
+}
+
+AudioSystem::forced_config AudioPolicyService::getForceUse(AudioSystem::force_use usage)
+{
+    if (mpPolicyManager == NULL) {
+        return AudioSystem::FORCE_NONE;
+    }
+    if (!checkPermission()) {
+        return AudioSystem::FORCE_NONE;
+    }
+    if (usage < 0 || usage >= AudioSystem::NUM_FORCE_USE) {
+        return AudioSystem::FORCE_NONE;
+    }
+    return mpPolicyManager->getForceUse(usage);
+}
+
+audio_io_handle_t AudioPolicyService::getOutput(AudioSystem::stream_type stream,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    AudioSystem::output_flags flags)
+{
+    if (mpPolicyManager == NULL) {
+        return NULL;
+    }
+    LOGV("getOutput() tid %d", gettid());
+    Mutex::Autolock _l(mLock);
+    return mpPolicyManager->getOutput(stream, samplingRate, format, channels, flags);
+}
+
+status_t AudioPolicyService::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    LOGV("startOutput() tid %d", gettid());
+    Mutex::Autolock _l(mLock);
+    return mpPolicyManager->startOutput(output, stream);
+}
+
+status_t AudioPolicyService::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    LOGV("stopOutput() tid %d", gettid());
+    Mutex::Autolock _l(mLock);
+    return mpPolicyManager->stopOutput(output, stream);
+}
+
+void AudioPolicyService::releaseOutput(audio_io_handle_t output)
+{
+    if (mpPolicyManager == NULL) {
+        return;
+    }
+    LOGV("releaseOutput() tid %d", gettid());
+    Mutex::Autolock _l(mLock);
+    mpPolicyManager->releaseOutput(output);
+}
+
+audio_io_handle_t AudioPolicyService::getInput(int inputSource,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    AudioSystem::audio_in_acoustics acoustics)
+{
+    if (mpPolicyManager == NULL) {
+        return NULL;
+    }
+    Mutex::Autolock _l(mLock);
+    return mpPolicyManager->getInput(inputSource, samplingRate, format, channels, acoustics);
+}
+
+status_t AudioPolicyService::startInput(audio_io_handle_t input)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    Mutex::Autolock _l(mLock);
+    return mpPolicyManager->startInput(input);
+}
+
+status_t AudioPolicyService::stopInput(audio_io_handle_t input)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    Mutex::Autolock _l(mLock);
+    return mpPolicyManager->stopInput(input);
+}
+
+void AudioPolicyService::releaseInput(audio_io_handle_t input)
+{
+    if (mpPolicyManager == NULL) {
+        return;
+    }
+    Mutex::Autolock _l(mLock);
+    mpPolicyManager->releaseInput(input);
+}
+
+status_t AudioPolicyService::initStreamVolume(AudioSystem::stream_type stream,
+                                            int indexMin,
+                                            int indexMax)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!checkPermission()) {
+        return PERMISSION_DENIED;
+    }
+    if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
+        return BAD_VALUE;
+    }
+    mpPolicyManager->initStreamVolume(stream, indexMin, indexMax);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyService::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!checkPermission()) {
+        return PERMISSION_DENIED;
+    }
+    if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
+        return BAD_VALUE;
+    }
+
+    return mpPolicyManager->setStreamVolumeIndex(stream, index);
+}
+
+status_t AudioPolicyService::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
+{
+    if (mpPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!checkPermission()) {
+        return PERMISSION_DENIED;
+    }
+    if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
+        return BAD_VALUE;
+    }
+    return mpPolicyManager->getStreamVolumeIndex(stream, index);
+}
+
+void AudioPolicyService::binderDied(const wp<IBinder>& who) {
+    LOGW("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
+}
+
+status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
+{
+    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+        dumpPermissionDenial(fd, args);
+    } else {
+
+    }
+    return NO_ERROR;
+}
+
+status_t AudioPolicyService::dumpPermissionDenial(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    snprintf(buffer, SIZE, "Permission Denial: "
+            "can't dump AudioPolicyService from pid=%d, uid=%d\n",
+            IPCThreadState::self()->getCallingPid(),
+            IPCThreadState::self()->getCallingUid());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioPolicyService::onTransact(
+        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioPolicyService::onTransact(code, data, reply, flags);
+}
+
+
+// ----------------------------------------------------------------------------
+void AudioPolicyService::instantiate() {
+    defaultServiceManager()->addService(
+            String16("media.audio_policy"), new AudioPolicyService());
+}
+
+
+// ----------------------------------------------------------------------------
+// AudioPolicyClientInterface implementation
+// ----------------------------------------------------------------------------
+
+
+audio_io_handle_t AudioPolicyService::openOutput(uint32_t *pDevices,
+                                uint32_t *pSamplingRate,
+                                uint32_t *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t *pLatencyMs,
+                                AudioSystem::output_flags flags)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        LOGW("openOutput() could not get AudioFlinger");
+        return NULL;
+    }
+
+    return af->openOutput(pDevices, pSamplingRate, (uint32_t *)pFormat, pChannels, pLatencyMs, flags);
+}
+
+audio_io_handle_t AudioPolicyService::openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        LOGW("openDuplicateOutput() could not get AudioFlinger");
+        return NULL;
+    }
+    return af->openDuplicateOutput(output1, output2);
+}
+
+status_t AudioPolicyService::closeOutput(audio_io_handle_t output)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
+
+    return af->closeOutput(output);
+}
+
+
+status_t AudioPolicyService::suspendOutput(audio_io_handle_t output)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        LOGW("suspendOutput() could not get AudioFlinger");
+        return PERMISSION_DENIED;
+    }
+
+    return af->suspendOutput(output);
+}
+
+status_t AudioPolicyService::restoreOutput(audio_io_handle_t output)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        LOGW("restoreOutput() could not get AudioFlinger");
+        return PERMISSION_DENIED;
+    }
+
+    return af->restoreOutput(output);
+}
+
+audio_io_handle_t AudioPolicyService::openInput(uint32_t *pDevices,
+                                uint32_t *pSamplingRate,
+                                uint32_t *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t acoustics)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) {
+        LOGW("openInput() could not get AudioFlinger");
+        return NULL;
+    }
+
+    return af->openInput(pDevices, pSamplingRate, (uint32_t *)pFormat, pChannels, acoustics);
+}
+
+status_t AudioPolicyService::closeInput(audio_io_handle_t input)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
+
+    return af->closeInput(input);
+}
+
+status_t AudioPolicyService::setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output)
+{
+    return mAudioCommandThread->volumeCommand((int)stream, volume, (void *)output);
+}
+
+status_t AudioPolicyService::setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output)
+{
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
+
+    return af->setStreamOutput(stream, output);
+}
+
+
+void AudioPolicyService::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
+{
+    mAudioCommandThread->parametersCommand((void *)ioHandle, keyValuePairs);
+}
+
+String8 AudioPolicyService::getParameters(audio_io_handle_t ioHandle, const String8& keys)
+{
+    String8 result = AudioSystem::getParameters(ioHandle, keys);
+    return result;
+}
+
+status_t AudioPolicyService::startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream)
+{
+    mTonePlaybacThread->startToneCommand(tone, stream);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyService::stopTone()
+{
+    mTonePlaybacThread->stopToneCommand();
+    return NO_ERROR;
+}
+
+
+// -----------  AudioPolicyService::AudioCommandThread implementation ----------
+
+AudioPolicyService::AudioCommandThread::AudioCommandThread()
+    :   Thread(false)
+{
+    mpToneGenerator = NULL;
+}
+
+
+AudioPolicyService::AudioCommandThread::~AudioCommandThread()
+{
+    mAudioCommands.clear();
+    if (mpToneGenerator != NULL) delete mpToneGenerator;
+}
+
+void AudioPolicyService::AudioCommandThread::onFirstRef()
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+
+    snprintf(buffer, SIZE, "AudioCommandThread");
+
+    run(buffer, ANDROID_PRIORITY_AUDIO);
+}
+
+bool AudioPolicyService::AudioCommandThread::threadLoop()
+{
+    mLock.lock();
+    while (!exitPending())
+    {
+        while(!mAudioCommands.isEmpty()) {
+            AudioCommand *command = mAudioCommands[0];
+            mAudioCommands.removeAt(0);
+            switch (command->mCommand) {
+            case START_TONE: {
+                mLock.unlock();
+                ToneData *data = (ToneData *)command->mParam;
+                LOGV("AudioCommandThread() processing start tone %d on stream %d",
+                        data->mType, data->mStream);
+                if (mpToneGenerator != NULL)
+                    delete mpToneGenerator;
+                mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
+                mpToneGenerator->startTone(data->mType);
+                delete data;
+                mLock.lock();
+                }break;
+            case STOP_TONE: {
+                mLock.unlock();
+                LOGV("AudioCommandThread() processing stop tone");
+                if (mpToneGenerator != NULL) {
+                    mpToneGenerator->stopTone();
+                    delete mpToneGenerator;
+                    mpToneGenerator = NULL;
+                }
+                mLock.lock();
+                }break;
+            case SET_VOLUME: {
+                VolumeData *data = (VolumeData *)command->mParam;
+                LOGV("AudioCommandThread() processing set volume stream %d, volume %f, output %p", data->mStream, data->mVolume, data->mIO);
+                mCommandStatus = AudioSystem::setStreamVolume(data->mStream, data->mVolume, data->mIO);
+                mCommandCond.signal();
+                mWaitWorkCV.wait(mLock);
+                delete data;
+                }break;
+            case SET_PARAMETERS: {
+                 ParametersData *data = (ParametersData *)command->mParam;
+                 LOGV("AudioCommandThread() processing set parameters string %s, io %p", data->mKeyValuePairs.string(), data->mIO);
+                 mCommandStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
+                 mCommandCond.signal();
+                 mWaitWorkCV.wait(mLock);
+                 delete data;
+                 }break;
+            default:
+                LOGW("AudioCommandThread() unknown command %d", command->mCommand);
+            }
+            delete command;
+        }
+        LOGV("AudioCommandThread() going to sleep");
+        mWaitWorkCV.wait(mLock);
+        LOGV("AudioCommandThread() waking up");
+    }
+    mLock.unlock();
+    return false;
+}
+
+void AudioPolicyService::AudioCommandThread::startToneCommand(int type, int stream)
+{
+    Mutex::Autolock _l(mLock);
+    AudioCommand *command = new AudioCommand();
+    command->mCommand = START_TONE;
+    ToneData *data = new ToneData();
+    data->mType = type;
+    data->mStream = stream;
+    command->mParam = (void *)data;
+    mAudioCommands.add(command);
+    LOGV("AudioCommandThread() adding tone start type %d, stream %d", type, stream);
+    mWaitWorkCV.signal();
+}
+
+void AudioPolicyService::AudioCommandThread::stopToneCommand()
+{
+    Mutex::Autolock _l(mLock);
+    AudioCommand *command = new AudioCommand();
+    command->mCommand = STOP_TONE;
+    command->mParam = NULL;
+    mAudioCommands.add(command);
+    LOGV("AudioCommandThread() adding tone stop");
+    mWaitWorkCV.signal();
+}
+
+status_t AudioPolicyService::AudioCommandThread::volumeCommand(int stream, float volume, void *output)
+{
+    Mutex::Autolock _l(mLock);
+    AudioCommand *command = new AudioCommand();
+    command->mCommand = SET_VOLUME;
+    VolumeData *data = new VolumeData();
+    data->mStream = stream;
+    data->mVolume = volume;
+    data->mIO = output;
+    command->mParam = data;
+    mAudioCommands.add(command);
+    LOGV("AudioCommandThread() adding set volume stream %d, volume %f, output %p", stream, volume, output);
+    mWaitWorkCV.signal();
+    mCommandCond.wait(mLock);
+    status_t status =  mCommandStatus;
+    mWaitWorkCV.signal();
+    return status;
+}
+
+status_t AudioPolicyService::AudioCommandThread::parametersCommand(void *ioHandle, const String8& keyValuePairs)
+{
+    Mutex::Autolock _l(mLock);
+    AudioCommand *command = new AudioCommand();
+    command->mCommand = SET_PARAMETERS;
+    ParametersData *data = new ParametersData();
+    data->mIO = ioHandle;
+    data->mKeyValuePairs = keyValuePairs;
+    command->mParam = data;
+    mAudioCommands.add(command);
+    LOGV("AudioCommandThread() adding set parameter string %s, io %p", keyValuePairs.string(), ioHandle);
+    mWaitWorkCV.signal();
+    mCommandCond.wait(mLock);
+    status_t status =  mCommandStatus;
+    mWaitWorkCV.signal();
+    return status;
+}
+
+void AudioPolicyService::AudioCommandThread::exit()
+{
+    LOGV("AudioCommandThread::exit");
+    {
+        AutoMutex _l(mLock);
+        requestExit();
+        mWaitWorkCV.signal();
+    }
+    requestExitAndWait();
+}
+
+}; // namespace android
diff --git a/libs/audioflinger/AudioPolicyService.h b/libs/audioflinger/AudioPolicyService.h
new file mode 100644
index 0000000..47173dd
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyService.h
@@ -0,0 +1,201 @@
+/*
+ * 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_AUDIOPOLICYSERVICE_H
+#define ANDROID_AUDIOPOLICYSERVICE_H
+
+#include <media/IAudioPolicyService.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+#include <media/ToneGenerator.h>
+
+namespace android {
+
+class String8;
+
+// ----------------------------------------------------------------------------
+
+class AudioPolicyService: public BnAudioPolicyService, public AudioPolicyClientInterface, public IBinder::DeathRecipient
+{
+
+public:
+    static  void        instantiate();
+
+    virtual status_t    dump(int fd, const Vector<String16>& args);
+
+    //
+    // BnAudioPolicyService (see AudioPolicyInterface for method descriptions)
+    //
+
+    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 status_t setPhoneState(int state);
+    virtual status_t setRingerMode(uint32_t mode, uint32_t mask);
+    virtual status_t setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+    virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
+    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);
+    virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+    virtual void releaseOutput(audio_io_handle_t output);
+    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);
+    virtual status_t startInput(audio_io_handle_t input);
+    virtual status_t stopInput(audio_io_handle_t input);
+    virtual void releaseInput(audio_io_handle_t input);
+    virtual status_t 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);
+
+    virtual     status_t    onTransact(
+                                uint32_t code,
+                                const Parcel& data,
+                                Parcel* reply,
+                                uint32_t flags);
+
+    // IBinder::DeathRecipient
+    virtual     void        binderDied(const wp<IBinder>& who);
+
+    //
+    // AudioPolicyClientInterface
+    //
+    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 setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output);
+    virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output);
+    virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
+    virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
+    virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream);
+    virtual status_t stopTone();
+
+private:
+                        AudioPolicyService();
+    virtual             ~AudioPolicyService();
+
+    static const char *sAudioPolicyLibrary;
+    static const char *sAudioPolicyGenericLibrary;
+    // Thread used for tone playback and to send audio config commands to audio flinger
+    // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because startTone()
+    // and stopTone() are normally called with mLock locked and requesting a tone start or stop will cause
+    // calls to AudioPolicyService and an attempt to lock mLock.
+    // For audio config commands, it is necessary because audio flinger requires that the calling process (user)
+    // has permission to modify audio settings.
+    class AudioCommandThread : public Thread {
+    public:
+
+        // commands for tone AudioCommand
+        enum {
+            START_TONE,
+            STOP_TONE,
+            SET_VOLUME,
+            SET_PARAMETERS
+        };
+
+        AudioCommandThread ();
+        virtual             ~AudioCommandThread();
+
+        // Thread virtuals
+        virtual     void        onFirstRef();
+        virtual     bool        threadLoop();
+
+                    void        exit();
+                    void        startToneCommand(int type = 0, int stream = 0);
+                    void        stopToneCommand();
+                    status_t    volumeCommand(int stream, float volume, void *output);
+                    status_t    parametersCommand(void *ioHandle, const String8& keyValuePairs);
+
+    private:
+        // descriptor for requested tone playback event
+        class AudioCommand {
+        public:
+            int mCommand;   // START_TONE, STOP_TONE ...
+            void *mParam;
+        };
+
+        class ToneData {
+        public:
+            int mType;      // tone type (START_TONE only)
+            int mStream;    // stream type (START_TONE only)
+        };
+
+        class VolumeData {
+        public:
+            int mStream;
+            float mVolume;
+            void *mIO;
+        };
+        class ParametersData {
+        public:
+            void *mIO;
+            String8 mKeyValuePairs;
+        };
+
+
+        Mutex   mLock;
+        Condition mWaitWorkCV;
+        Vector<AudioCommand *> mAudioCommands;    // list of pending tone events
+        Condition              mCommandCond;
+        status_t               mCommandStatus;
+        ToneGenerator *mpToneGenerator;     // the tone generator
+    };
+
+    // Internal dump utilities.
+    status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
+
+
+    Mutex   mLock;      // prevents concurrent access to AudioPolicy manager functions changing device
+                        // connection stated our routing
+    AudioPolicyInterface* mpPolicyManager;          // the platform specific policy manager
+    sp <AudioCommandThread> mAudioCommandThread;    // audio commands thread
+    sp <AudioCommandThread> mTonePlaybacThread;     // tone playback thread
+    void *mpPolicyManagerLibHandle;
+};
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYSERVICE_H
+
+
+
+
+
+
+
+
diff --git a/media/java/android/media/AudioFormat.java b/media/java/android/media/AudioFormat.java
index 0732b61..500f6a4 100644
--- a/media/java/android/media/AudioFormat.java
+++ b/media/java/android/media/AudioFormat.java
@@ -37,15 +37,58 @@
     public static final int ENCODING_PCM_8BIT = 3;  // accessed by native code
 
     /** Invalid audio channel configuration */
-    public static final int CHANNEL_CONFIGURATION_INVALID   = 0;
+    /** @deprecated use CHANNEL_INVALID instead  */
+    @Deprecated    public static final int CHANNEL_CONFIGURATION_INVALID   = 0;
     /** Default audio channel configuration */
-    public static final int CHANNEL_CONFIGURATION_DEFAULT   = 1;
+    /** @deprecated use CHANNEL_OUT_DEFAULT or CHANNEL_IN_DEFAULT instead  */
+    @Deprecated    public static final int CHANNEL_CONFIGURATION_DEFAULT   = 1;
     /** Mono audio configuration */
-    public static final int CHANNEL_CONFIGURATION_MONO      = 2;
+    /** @deprecated use CHANNEL_OUT_MONO or CHANNEL_IN_MONO instead  */
+    @Deprecated    public static final int CHANNEL_CONFIGURATION_MONO      = 2;
     /** Stereo (2 channel) audio configuration */
-    public static final int CHANNEL_CONFIGURATION_STEREO    = 3;
+    /** @deprecated use CHANNEL_OUT_STEREO or CHANNEL_IN_STEREO instead  */
+    @Deprecated    public static final int CHANNEL_CONFIGURATION_STEREO    = 3;
+
+    /** Invalid audio channel mask */
+    public static final int CHANNEL_INVALID   = -1;
+    /** Default audio channel mask */
+    // Channel mask definitions must be kept in sync with native values in include/media/AudioSystem.h
+    public static final int CHANNEL_OUT_DEFAULT = 0;
+    public static final int CHANNEL_OUT_FRONT_LEFT = 0x1;
+    public static final int CHANNEL_OUT_FRONT_RIGHT = 0x2;
+    public static final int CHANNEL_OUT_FRONT_CENTER = 0x4;
+    public static final int CHANNEL_OUT_LOW_FREQUENCY = 0x8;
+    public static final int CHANNEL_OUT_BACK_LEFT = 0x10;
+    public static final int CHANNEL_OUT_BACK_RIGHT = 0x20;
+    public static final int CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40;
+    public static final int CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80;
+    public static final int CHANNEL_OUT_BACK_CENTER = 0x100;
+    public static final int CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT;
+    public static final int CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT);
+    public static final int CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
+            CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT);
+    public static final int CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
+            CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER);
+    public static final int 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);
+    public static final int 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);
+
+    public static final int CHANNEL_IN_DEFAULT = 0;
+    public static final int CHANNEL_IN_LEFT = 0x10000;
+    public static final int CHANNEL_IN_RIGHT = 0x20000;
+    public static final int CHANNEL_IN_FRONT = 0x40000;
+    public static final int CHANNEL_IN_BACK = 0x80000;
+    public static final int CHANNEL_IN_LEFT_PROCESSED = 0x100000;
+    public static final int CHANNEL_IN_RIGHT_PROCESSED = 0x200000;
+    public static final int CHANNEL_IN_FRONT_PROCESSED = 0x400000;
+    public static final int CHANNEL_IN_BACK_PROCESSED = 0x800000;
+    public static final int CHANNEL_IN_PRESSURE = 0x1000000;
+    public static final int CHANNEL_IN_X_AXIS = 0x2000000;
+    public static final int CHANNEL_IN_Y_AXIS = 0x4000000;
+    public static final int CHANNEL_IN_Z_AXIS = 0x8000000;
+    public static final int CHANNEL_IN_MONO = CHANNEL_IN_FRONT;
+    public static final int CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT);
 
 }
-
-
-
diff --git a/media/java/android/media/AudioManager.java b/media/java/android/media/AudioManager.java
index a65a417..040d4bc 100644
--- a/media/java/android/media/AudioManager.java
+++ b/media/java/android/media/AudioManager.java
@@ -140,11 +140,17 @@
     public static final int STREAM_NOTIFICATION = AudioSystem.STREAM_NOTIFICATION;
     /** @hide The audio stream for phone calls when connected to bluetooth */
     public static final int STREAM_BLUETOOTH_SCO = AudioSystem.STREAM_BLUETOOTH_SCO;
+    /** @hide The audio stream for enforced system sounds in certain countries (e.g camera in Japan) */
+    public static final int STREAM_SYSTEM_ENFORCED = AudioSystem.STREAM_SYSTEM_ENFORCED;
+    /** The audio stream for DTMF Tones */
+    public static final int STREAM_DTMF = AudioSystem.STREAM_DTMF;
+    /** @hide The audio stream for text to speech (TTS) */
+    public static final int STREAM_TTS = AudioSystem.STREAM_TTS;
     /** Number of audio streams */
     /**
      * @deprecated Use AudioSystem.getNumStreamTypes() instead
      */
-    public static final int NUM_STREAMS = AudioSystem.NUM_STREAMS;
+    @Deprecated public static final int NUM_STREAMS = AudioSystem.NUM_STREAMS;
 
 
     /** @hide Maximum volume index values for audio streams */
@@ -156,6 +162,9 @@
         8,  // STREAM_ALARM
         8,  // STREAM_NOTIFICATION
         16, // STREAM_BLUETOOTH_SCO
+        8,  // STREAM_SYSTEM_ENFORCED
+        16, // STREAM_DTMF
+        16  // STREAM_TTS
     };
 
     /**  @hide Default volume index values for audio streams */
@@ -166,7 +175,10 @@
         11, // STREAM_MUSIC
         6,  // STREAM_ALARM
         5,  // STREAM_NOTIFICATION
-        7   // STREAM_BLUETOOTH_SCO
+        7,  // STREAM_BLUETOOTH_SCO
+        5,  // STREAM_SYSTEM_ENFORCED
+        11, // STREAM_DTMF
+        11  // STREAM_TTS
     };
 
     /**
@@ -637,9 +649,11 @@
      *           <var>false</var> to turn it off
      */
     public void setSpeakerphoneOn(boolean on){
-        // Temporary fix for issue #1713090 until audio routing is refactored in eclair release.
-        // MODE_INVALID indicates to AudioService that setRouting() was initiated by AudioManager
-        setRoutingP(MODE_INVALID, on ? ROUTE_SPEAKER: 0, ROUTE_SPEAKER);
+        if (on) {
+            AudioSystem.setForceUse(AudioSystem.FOR_COMMUNICATION, AudioSystem.FORCE_SPEAKER);
+        } else {
+            AudioSystem.setForceUse(AudioSystem.FOR_COMMUNICATION, AudioSystem.FORCE_NONE);
+        }
     }
 
     /**
@@ -648,41 +662,47 @@
      * @return true if speakerphone is on, false if it's off
      */
     public boolean isSpeakerphoneOn() {
-        return (getRoutingP(MODE_IN_CALL) & ROUTE_SPEAKER) == 0 ? false : true;
+        if (AudioSystem.getForceUse(AudioSystem.FOR_COMMUNICATION) == AudioSystem.FORCE_SPEAKER) {
+            return true;
+        } else {
+            return false;
+        }
      }
 
     /**
-     * Sets audio routing to the Bluetooth headset on or off.
+     * Request use of Bluetooth SCO headset for communications.
      *
-     * @param on set <var>true</var> to route SCO (voice) audio to/from Bluetooth
-     *           headset; <var>false</var> to route audio to/from phone earpiece
+     * @param on set <var>true</var> to use bluetooth SCO for communications;
+     *               <var>false</var> to not use bluetooth SCO for communications
      */
     public void setBluetoothScoOn(boolean on){
-        // Temporary fix for issue #1713090 until audio routing is refactored in eclair release.
-        // MODE_INVALID indicates to AudioService that setRouting() was initiated by AudioManager
-        setRoutingP(MODE_INVALID, on ? ROUTE_BLUETOOTH_SCO: 0, ROUTE_BLUETOOTH_SCO);
+        if (on) {
+            AudioSystem.setForceUse(AudioSystem.FOR_COMMUNICATION, AudioSystem.FORCE_BT_SCO);
+        } else {
+            AudioSystem.setForceUse(AudioSystem.FOR_COMMUNICATION, AudioSystem.FORCE_NONE);
+        }
     }
 
     /**
-     * Checks whether audio routing to the Bluetooth headset is on or off.
+     * Checks whether communications use Bluetooth SCO.
      *
-     * @return true if SCO audio is being routed to/from Bluetooth headset;
+     * @return true if SCO is used for communications;
      *         false if otherwise
      */
     public boolean isBluetoothScoOn() {
-        return (getRoutingP(MODE_IN_CALL) & ROUTE_BLUETOOTH_SCO) == 0 ? false : true;
+        if (AudioSystem.getForceUse(AudioSystem.FOR_COMMUNICATION) == AudioSystem.FORCE_BT_SCO) {
+            return true;
+        } else {
+            return false;
+        }
     }
 
     /**
-     * Sets A2DP audio routing to the Bluetooth headset on or off.
-     *
      * @param on set <var>true</var> to route A2DP audio to/from Bluetooth
      *           headset; <var>false</var> disable A2DP audio
+     * @deprecated Do not use.
      */
-    public void setBluetoothA2dpOn(boolean on){
-        // Temporary fix for issue #1713090 until audio routing is refactored in eclair release.
-        // MODE_INVALID indicates to AudioService that setRouting() was initiated by AudioManager
-        setRoutingP(MODE_INVALID, on ? ROUTE_BLUETOOTH_A2DP: 0, ROUTE_BLUETOOTH_A2DP);
+    @Deprecated public void setBluetoothA2dpOn(boolean on){
     }
 
     /**
@@ -692,7 +712,12 @@
      *         false if otherwise
      */
     public boolean isBluetoothA2dpOn() {
-        return (getRoutingP(MODE_NORMAL) & ROUTE_BLUETOOTH_A2DP) == 0 ? false : true;
+        if (AudioSystem.getDeviceConnectionState(AudioSystem.DEVICE_OUT_BLUETOOTH_A2DP,"") 
+            == AudioSystem.DEVICE_STATE_UNAVAILABLE) {
+            return false;
+        } else {
+            return true;
+        }
     }
 
     /**
@@ -700,12 +725,9 @@
      *
      * @param on set <var>true</var> to route audio to/from wired
      *           headset; <var>false</var> disable wired headset audio
-     * @hide
+     * @deprecated Do not use.
      */
-    public void setWiredHeadsetOn(boolean on){
-        // Temporary fix for issue #1713090 until audio routing is refactored in eclair release.
-        // MODE_INVALID indicates to AudioService that setRouting() was initiated by AudioManager
-        setRoutingP(MODE_INVALID, on ? ROUTE_HEADSET: 0, ROUTE_HEADSET);
+    @Deprecated public void setWiredHeadsetOn(boolean on){
     }
 
     /**
@@ -713,10 +735,14 @@
      *
      * @return true if audio is being routed to/from wired headset;
      *         false if otherwise
-     * @hide
      */
     public boolean isWiredHeadsetOn() {
-        return (getRoutingP(MODE_NORMAL) & ROUTE_HEADSET) == 0 ? false : true;
+        if (AudioSystem.getDeviceConnectionState(AudioSystem.DEVICE_OUT_WIRED_HEADSET,"") 
+                == AudioSystem.DEVICE_STATE_UNAVAILABLE) {
+            return false;
+        } else {
+            return true;
+        }
     }
 
     /**
@@ -726,12 +752,7 @@
      *           <var>false</var> to turn mute off
      */
     public void setMicrophoneMute(boolean on){
-        IAudioService service = getService();
-        try {
-            service.setMicrophoneMute(on);
-        } catch (RemoteException e) {
-            Log.e(TAG, "Dead object in setMicrophoneMute", e);
-        }
+        AudioSystem.muteMicrophone(on);
     }
 
     /**
@@ -740,13 +761,7 @@
      * @return true if microphone is muted, false if it's not
      */
     public boolean isMicrophoneMute() {
-        IAudioService service = getService();
-        try {
-            return service.isMicrophoneMute();
-        } catch (RemoteException e) {
-            Log.e(TAG, "Dead object in isMicrophoneMute", e);
-            return false;
-        }
+        return AudioSystem.isMicrophoneMuted();
     }
 
     /**
@@ -809,32 +824,39 @@
     /* Routing bits for setRouting/getRouting API */
     /**
      * Routing audio output to earpiece
+     * @deprecated
      */
-    public static final int ROUTE_EARPIECE          = AudioSystem.ROUTE_EARPIECE;
+    @Deprecated public static final int ROUTE_EARPIECE          = AudioSystem.ROUTE_EARPIECE;
     /**
      * Routing audio output to spaker
+     * @deprecated
      */
-    public static final int ROUTE_SPEAKER           = AudioSystem.ROUTE_SPEAKER;
+    @Deprecated public static final int ROUTE_SPEAKER           = AudioSystem.ROUTE_SPEAKER;
     /**
      * @deprecated use {@link #ROUTE_BLUETOOTH_SCO}
+     * @deprecated
      */
     @Deprecated public static final int ROUTE_BLUETOOTH = AudioSystem.ROUTE_BLUETOOTH_SCO;
     /**
      * Routing audio output to bluetooth SCO
+     * @deprecated
      */
-    public static final int ROUTE_BLUETOOTH_SCO     = AudioSystem.ROUTE_BLUETOOTH_SCO;
+    @Deprecated public static final int ROUTE_BLUETOOTH_SCO     = AudioSystem.ROUTE_BLUETOOTH_SCO;
     /**
      * Routing audio output to headset
+     * @deprecated
      */
-    public static final int ROUTE_HEADSET           = AudioSystem.ROUTE_HEADSET;
+    @Deprecated public static final int ROUTE_HEADSET           = AudioSystem.ROUTE_HEADSET;
     /**
      * Routing audio output to bluetooth A2DP
+     * @deprecated
      */
-    public static final int ROUTE_BLUETOOTH_A2DP    = AudioSystem.ROUTE_BLUETOOTH_A2DP;
+    @Deprecated public static final int ROUTE_BLUETOOTH_A2DP    = AudioSystem.ROUTE_BLUETOOTH_A2DP;
     /**
      * Used for mask parameter of {@link #setRouting(int,int,int)}.
+     * @deprecated
      */
-    public static final int ROUTE_ALL               = AudioSystem.ROUTE_ALL;
+    @Deprecated public static final int ROUTE_ALL               = AudioSystem.ROUTE_ALL;
 
     /**
      * Sets the audio routing for a specified mode
@@ -846,16 +868,10 @@
      * ROUTE_xxx types. Unset bits indicate the route should be left unchanged
      *
      * @deprecated   Do not set audio routing directly, use setSpeakerphoneOn(),
-     * setBluetoothScoOn(), setBluetoothA2dpOn() and setWiredHeadsetOn() methods instead.
+     * setBluetoothScoOn() methods instead.
      */
-
+    @Deprecated
     public void setRouting(int mode, int routes, int mask) {
-        IAudioService service = getService();
-        try {
-            service.setRouting(mode, routes, mask);
-        } catch (RemoteException e) {
-            Log.e(TAG, "Dead object in setRouting", e);
-        }
     }
 
     /**
@@ -869,13 +885,7 @@
      */
     @Deprecated
     public int getRouting(int mode) {
-        IAudioService service = getService();
-        try {
-            return service.getRouting(mode);
-        } catch (RemoteException e) {
-            Log.e(TAG, "Dead object in getRouting", e);
-            return -1;
-        }
+        return -1;
     }
 
     /**
@@ -884,13 +894,7 @@
      * @return true if any music tracks are active.
      */
     public boolean isMusicActive() {
-        IAudioService service = getService();
-        try {
-            return service.isMusicActive();
-        } catch (RemoteException e) {
-            Log.e(TAG, "Dead object in isMusicActive", e);
-            return false;
-        }
+        return AudioSystem.isMusicActive();
     }
 
     /*
@@ -906,14 +910,32 @@
      */
     /**
      * @hide
+     * @deprecated Use {@link #setPrameters(String)} instead
      */
-    public void setParameter(String key, String value) {
-        IAudioService service = getService();
-        try {
-            service.setParameter(key, value);
-        } catch (RemoteException e) {
-            Log.e(TAG, "Dead object in setParameter", e);
-        }
+    @Deprecated public void setParameter(String key, String value) {
+        setParameters(key+"="+value);
+    }
+
+    /**
+     * Sets a variable number of parameter values to audio hardware.
+     *
+     * @param keyValuePairs list of parameters key value pairs in the form:
+     *    key1=value1;key2=value2;...
+     *
+     */
+    public void setParameters(String keyValuePairs) {
+        AudioSystem.setParameters(keyValuePairs);
+    }
+
+    /**
+     * Sets a varaible number of parameter values to audio hardware.
+     *
+     * @param keys list of parameters
+     * @return list of parameters key value pairs in the form:
+     *    key1=value1;key2=value2;...
+     */
+    public String getParameters(String keys) {
+        return AudioSystem.getParameters(keys);
     }
 
     /* Sound effect identifiers */
@@ -1082,31 +1104,4 @@
       * {@hide}
       */
      private IBinder mICallBack = new Binder();
-
-     /**
-      * {@hide}
-      */
-     private void setRoutingP(int mode, int routes, int mask) {
-         IAudioService service = getService();
-         try {
-             service.setRouting(mode, routes, mask);
-         } catch (RemoteException e) {
-             Log.e(TAG, "Dead object in setRouting", e);
-         }
-     }
-
-
-     /**
-      * {@hide}
-      */
-     private int getRoutingP(int mode) {
-         IAudioService service = getService();
-         try {
-             return service.getRouting(mode);
-         } catch (RemoteException e) {
-             Log.e(TAG, "Dead object in getRouting", e);
-             return -1;
-         }
-     }
-
 }
diff --git a/media/java/android/media/AudioRecord.java b/media/java/android/media/AudioRecord.java
index 4d1535f9..d96331f 100644
--- a/media/java/android/media/AudioRecord.java
+++ b/media/java/android/media/AudioRecord.java
@@ -86,7 +86,7 @@
     public static final int ERROR_INVALID_OPERATION = -3;
     
     private static final int AUDIORECORD_ERROR_SETUP_ZEROFRAMECOUNT      = -16;
-    private static final int AUDIORECORD_ERROR_SETUP_INVALIDCHANNELCOUNT = -17;
+    private static final int AUDIORECORD_ERROR_SETUP_INVALIDCHANNELMASK  = -17;
     private static final int AUDIORECORD_ERROR_SETUP_INVALIDFORMAT       = -18;
     private static final int AUDIORECORD_ERROR_SETUP_INVALIDSOURCE       = -19;
     private static final int AUDIORECORD_ERROR_SETUP_NATIVEINITFAILED    = -20;
@@ -135,7 +135,7 @@
     /**
      * The current audio channel configuration
      */
-    private int mChannelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+    private int mChannelConfiguration = AudioFormat.CHANNEL_IN_MONO;
     /**
      * The encoding of the audio samples.
      * @see AudioFormat#ENCODING_PCM_8BIT
@@ -193,8 +193,8 @@
      * @param sampleRateInHz the sample rate expressed in Hertz. Examples of rates are (but
      *   not limited to) 44100, 22050 and 11025.
      * @param channelConfig describes the configuration of the audio channels. 
-     *   See {@link AudioFormat#CHANNEL_CONFIGURATION_MONO} and
-     *   {@link AudioFormat#CHANNEL_CONFIGURATION_STEREO}
+     *   See {@link AudioFormat#CHANNEL_IN_MONO} and
+     *   {@link AudioFormat#CHANNEL_IN_STEREO}
      * @param audioFormat the format in which the audio data is represented. 
      *   See {@link AudioFormat#ENCODING_PCM_16BIT} and 
      *   {@link AudioFormat#ENCODING_PCM_8BIT}
@@ -265,18 +265,18 @@
         //--------------
         // channel config
         switch (channelConfig) {
-        case AudioFormat.CHANNEL_CONFIGURATION_DEFAULT:
-        case AudioFormat.CHANNEL_CONFIGURATION_MONO:
+        case AudioFormat.CHANNEL_IN_DEFAULT:
+        case AudioFormat.CHANNEL_IN_MONO:
             mChannelCount = 1;
-            mChannelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+            mChannelConfiguration = AudioFormat.CHANNEL_IN_MONO;
             break;
-        case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
+        case AudioFormat.CHANNEL_IN_STEREO:
             mChannelCount = 2;
-            mChannelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+            mChannelConfiguration = AudioFormat.CHANNEL_IN_STEREO;
             break;
         default:
             mChannelCount = 0;
-        mChannelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_INVALID;
+        mChannelConfiguration = AudioFormat.CHANNEL_INVALID;
         throw (new IllegalArgumentException("Unsupported channel configuration."));
         }
 
@@ -368,8 +368,8 @@
 
     /**
      * Returns the configured channel configuration. 
-     * See {@link AudioFormat#CHANNEL_CONFIGURATION_MONO}
-     * and {@link AudioFormat#CHANNEL_CONFIGURATION_STEREO}.
+     * See {@link AudioFormat#CHANNEL_IN_MONO}
+     * and {@link AudioFormat#CHANNEL_IN_STEREO}.
      */
     public int getChannelConfiguration() {
         return mChannelConfiguration;
@@ -425,8 +425,8 @@
      * will be polled for new data.
      * @param sampleRateInHz the sample rate expressed in Hertz.
      * @param channelConfig describes the configuration of the audio channels. 
-     *   See {@link AudioFormat#CHANNEL_CONFIGURATION_MONO} and
-     *   {@link AudioFormat#CHANNEL_CONFIGURATION_STEREO}
+     *   See {@link AudioFormat#CHANNEL_IN_MONO} and
+     *   {@link AudioFormat#CHANNEL_IN_STEREO}
      * @param audioFormat the format in which the audio data is represented. 
      *   See {@link AudioFormat#ENCODING_PCM_16BIT}.
      * @return {@link #ERROR_BAD_VALUE} if the recording parameters are not supported by the 
@@ -438,14 +438,14 @@
     static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
         int channelCount = 0;
         switch(channelConfig) {
-        case AudioFormat.CHANNEL_CONFIGURATION_DEFAULT:
-        case AudioFormat.CHANNEL_CONFIGURATION_MONO:
+        case AudioFormat.CHANNEL_IN_DEFAULT:
+        case AudioFormat.CHANNEL_IN_MONO:
             channelCount = 1;
             break;
-        case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
+        case AudioFormat.CHANNEL_IN_STEREO:
             channelCount = 2;
             break;
-        case AudioFormat.CHANNEL_CONFIGURATION_INVALID:
+        case AudioFormat.CHANNEL_INVALID:
         default:
             loge("getMinBufferSize(): Invalid channel configuration.");
             return AudioRecord.ERROR_BAD_VALUE;
diff --git a/media/java/android/media/AudioService.java b/media/java/android/media/AudioService.java
index 58c04f3..177a7cf 100644
--- a/media/java/android/media/AudioService.java
+++ b/media/java/android/media/AudioService.java
@@ -20,6 +20,12 @@
 import android.content.ContentResolver;
 import android.content.Context;
 import android.content.Intent;
+import android.content.IntentFilter;
+import android.bluetooth.BluetoothIntent;
+import android.content.BroadcastReceiver;
+import android.bluetooth.BluetoothHeadset;
+import android.bluetooth.BluetoothA2dp;
+
 import android.content.pm.PackageManager;
 import android.database.ContentObserver;
 import android.media.MediaPlayer.OnCompletionListener;
@@ -94,16 +100,10 @@
     /** @see VolumeStreamState */
     private VolumeStreamState[] mStreamStates;
     private SettingsObserver mSettingsObserver;
-    
-    private boolean mMicMute;
+
     private int mMode;
-    private int[] mRoutes = new int[AudioSystem.NUM_MODES];
     private Object mSettingsLock = new Object();
     private boolean mMediaServerOk;
-    private boolean mSpeakerIsOn;
-    private boolean mBluetoothScoIsConnected;
-    private boolean mHeadsetIsConnected;
-    private boolean mBluetoothA2dpIsConnected;
 
     private SoundPool mSoundPool;
     private Object mSoundEffectsLock = new Object();
@@ -135,6 +135,23 @@
         {4, -1}   // FX_FOCUS_RETURN
     };
 
+    /* STREAM_VOLUME_ALIAS[] indicates for each stream if it uses the volume settings
+     * of another stream: This avoids multiplying the volume settings for hidden
+     * stream types that follow other stream behavior for volume settings
+     * NOTE: do not create loops in aliases! */
+    private int[] STREAM_VOLUME_ALIAS = new int[] {
+        AudioSystem.STREAM_VOICE_CALL,  // STREAM_VOICE_CALL
+        AudioSystem.STREAM_SYSTEM,  // STREAM_SYSTEM
+        AudioSystem.STREAM_RING,  // STREAM_RING
+        AudioSystem.STREAM_MUSIC, // STREAM_MUSIC
+        AudioSystem.STREAM_ALARM,  // STREAM_ALARM
+        AudioSystem.STREAM_NOTIFICATION,  // STREAM_NOTIFICATION
+        AudioSystem.STREAM_VOICE_CALL, // STREAM_BLUETOOTH_SCO
+        AudioSystem.STREAM_SYSTEM,  // STREAM_SYSTEM_ENFORCED
+        AudioSystem.STREAM_VOICE_CALL, // STREAM_DTMF
+        AudioSystem.STREAM_MUSIC  // STREAM_TTS
+    };
+
     private AudioSystem.ErrorCallback mAudioSystemCallback = new AudioSystem.ErrorCallback() {
         public void onError(int error) {
             switch (error) {
@@ -178,6 +195,21 @@
      */
     private int mVibrateSetting;
 
+    /** @see System#NOTIFICATIONS_USE_RING_VOLUME */
+    private int mNotificationsUseRingVolume;
+
+    // Broadcast receiver for device connections intent broadcasts
+    private final BroadcastReceiver mReceiver = new AudioServiceBroadcastReceiver();
+
+    //TODO: use common definitions with HeadsetObserver
+    private static final int BIT_HEADSET = (1 << 0);
+    private static final int BIT_HEADSET_NO_MIC = (1 << 1);
+    private static final int BIT_TTY = (1 << 2);
+    private static final int BIT_FM_HEADSET = (1 << 3);
+    private static final int BIT_FM_SPEAKER = (1 << 4);
+
+    private int mHeadsetState;
+
     ///////////////////////////////////////////////////////////////////////////
     // Construction
     ///////////////////////////////////////////////////////////////////////////
@@ -188,18 +220,20 @@
         mContentResolver = context.getContentResolver();
         mVolumePanel = new VolumePanel(context, this);
         mSettingsObserver = new SettingsObserver();
-        
+
         createAudioSystemThread();
-        createStreamStates();
         readPersistedSettings();
-        readAudioSettings();
+        createStreamStates();
         mMediaServerOk = true;
         AudioSystem.setErrorCallback(mAudioSystemCallback);
         loadSoundEffects();
-        mSpeakerIsOn = false;
-        mBluetoothScoIsConnected = false;
-        mHeadsetIsConnected = false;
-        mBluetoothA2dpIsConnected = false;
+
+        // Register for device connection intent broadcasts.
+        IntentFilter intentFilter =
+                new IntentFilter(Intent.ACTION_HEADSET_PLUG);
+        intentFilter.addAction(BluetoothA2dp.SINK_STATE_CHANGED_ACTION);
+        intentFilter.addAction(BluetoothIntent.HEADSET_AUDIO_STATE_CHANGED_ACTION);
+        context.registerReceiver(mReceiver, intentFilter);
     }
 
     private void createAudioSystemThread() {
@@ -223,65 +257,25 @@
     }
 
     private void createStreamStates() {
-        final int[] volumeLevelsPhone =
-            createVolumeLevels(0, AudioManager.MAX_STREAM_VOLUME[AudioManager.STREAM_VOICE_CALL]);
-        final int[] volumeLevelsCoarse =
-            createVolumeLevels(0, AudioManager.MAX_STREAM_VOLUME[AudioManager.STREAM_SYSTEM]);
-        final int[] volumeLevelsFine =
-            createVolumeLevels(0, AudioManager.MAX_STREAM_VOLUME[AudioManager.STREAM_MUSIC]);
-        final int[] volumeLevelsBtPhone =
-            createVolumeLevels(0,
-                    AudioManager.MAX_STREAM_VOLUME[AudioManager.STREAM_BLUETOOTH_SCO]);
-
         int numStreamTypes = AudioSystem.getNumStreamTypes();
         VolumeStreamState[] streams = mStreamStates = new VolumeStreamState[numStreamTypes];
 
         for (int i = 0; i < numStreamTypes; i++) {
-            final int[] levels;
+            streams[i] = new VolumeStreamState(System.VOLUME_SETTINGS[STREAM_VOLUME_ALIAS[i]], i);
+        }
 
-            switch (i) {
-
-                case AudioSystem.STREAM_MUSIC:
-                    levels = volumeLevelsFine;
-                    break;
-
-                case AudioSystem.STREAM_VOICE_CALL:
-                    levels = volumeLevelsPhone;
-                    break;
-
-                case AudioSystem.STREAM_BLUETOOTH_SCO:
-                    levels = volumeLevelsBtPhone;
-                    break;
-
-                default:
-                    levels = volumeLevelsCoarse;
-                    break;
-            }
-
-            if (i == AudioSystem.STREAM_BLUETOOTH_SCO) {
-                streams[i] = new VolumeStreamState(AudioManager.DEFAULT_STREAM_VOLUME[i], i,levels);
-            } else {
-                streams[i] = new VolumeStreamState(System.VOLUME_SETTINGS[i], i, levels);
+        // Correct stream index values for streams with aliases
+        for (int i = 0; i < numStreamTypes; i++) {
+            if (STREAM_VOLUME_ALIAS[i] != i) {
+                int index = rescaleIndex(streams[i].mIndex, STREAM_VOLUME_ALIAS[i], i);
+                streams[i].mIndex = streams[i].getValidIndex(index);
+                setStreamVolumeIndex(i, index);
+                index = rescaleIndex(streams[i].mLastAudibleIndex, STREAM_VOLUME_ALIAS[i], i);
+                streams[i].mLastAudibleIndex = streams[i].getValidIndex(index);
             }
         }
     }
 
-    private static int[] createVolumeLevels(int offset, int numlevels) {
-        double curve = 1.0f; // 1.4f
-        int [] volumes = new int[numlevels + offset];
-        for (int i = 0; i < offset; i++) {
-            volumes[i] = 0;
-        }
-
-        double val = 0;
-        double max = Math.pow(numlevels - 1, curve);
-        for (int i = 0; i < numlevels; i++) {
-            val = Math.pow(i, curve) / max;
-            volumes[offset + i] = (int) (val * 100.0f);
-        }
-        return volumes;
-    }
-
     private void readPersistedSettings() {
         final ContentResolver cr = mContentResolver;
 
@@ -291,12 +285,19 @@
 
         mRingerModeAffectedStreams = Settings.System.getInt(cr,
                 Settings.System.MODE_RINGER_STREAMS_AFFECTED,
-                ((1 << AudioManager.STREAM_RING)|(1 << AudioManager.STREAM_NOTIFICATION)|(1 << AudioManager.STREAM_SYSTEM)));
+                ((1 << AudioSystem.STREAM_RING)|(1 << AudioSystem.STREAM_NOTIFICATION)|
+                 (1 << AudioSystem.STREAM_SYSTEM)|(1 << AudioSystem.STREAM_SYSTEM_ENFORCED)));
 
         mMuteAffectedStreams = System.getInt(cr,
                 System.MUTE_STREAMS_AFFECTED,
                 ((1 << AudioSystem.STREAM_MUSIC)|(1 << AudioSystem.STREAM_RING)|(1 << AudioSystem.STREAM_SYSTEM)));
 
+        mNotificationsUseRingVolume = System.getInt(cr,
+                Settings.System.NOTIFICATIONS_USE_RING_VOLUME, 1);
+
+        if (mNotificationsUseRingVolume == 1) {
+            STREAM_VOLUME_ALIAS[AudioSystem.STREAM_NOTIFICATION] = AudioSystem.STREAM_RING;
+        }
         // Each stream will read its own persisted settings
 
         // Broadcast the sticky intent
@@ -307,25 +308,13 @@
         broadcastVibrateSetting(AudioManager.VIBRATE_TYPE_NOTIFICATION);
     }
 
-    private void readAudioSettings() {
-        synchronized (mSettingsLock) {
-            mMicMute = AudioSystem.isMicrophoneMuted();
-            mMode = AudioSystem.getMode();
-            for (int mode = 0; mode < AudioSystem.NUM_MODES; mode++) {
-                mRoutes[mode] = AudioSystem.getRouting(mode);
-            }
-        }
+    private void setStreamVolumeIndex(int stream, int index) {
+        AudioSystem.setStreamVolumeIndex(stream, (index + 5)/10);
     }
 
-    private void applyAudioSettings() {
-        synchronized (mSettingsLock) {
-            AudioSystem.muteMicrophone(mMicMute);
-            AudioSystem.setMode(mMode);
-            for (int mode = 0; mode < AudioSystem.NUM_MODES; mode++) {
-                AudioSystem.setRouting(mode, mRoutes[mode], AudioSystem.ROUTE_ALL);
-            }
-        }
-   }
+    private int rescaleIndex(int index, int srcStream, int dstStream) {
+        return (index * mStreamStates[dstStream].getMaxIndex() + mStreamStates[srcStream].getMaxIndex() / 2) / mStreamStates[srcStream].getMaxIndex();
+    }
 
     ///////////////////////////////////////////////////////////////////////////
     // IPC methods
@@ -354,44 +343,26 @@
         ensureValidDirection(direction);
         ensureValidStreamType(streamType);
 
-        boolean notificationsUseRingVolume = Settings.System.getInt(mContentResolver,
-                Settings.System.NOTIFICATIONS_USE_RING_VOLUME, 1) == 1;
-        if (notificationsUseRingVolume && streamType == AudioManager.STREAM_NOTIFICATION) {
-            // Redirect the volume change to the ring stream
-            streamType = AudioManager.STREAM_RING;
-        }
 
-        VolumeStreamState streamState = mStreamStates[streamType];
+        VolumeStreamState streamState = mStreamStates[STREAM_VOLUME_ALIAS[streamType]];
         final int oldIndex = streamState.mIndex;
         boolean adjustVolume = true;
 
         // If either the client forces allowing ringer modes for this adjustment,
         // or the stream type is one that is affected by ringer modes
         if ((flags & AudioManager.FLAG_ALLOW_RINGER_MODES) != 0
-                || streamType == AudioManager.STREAM_RING) {
+                || streamType == AudioSystem.STREAM_RING) {
             // Check if the ringer mode changes with this volume adjustment. If
             // it does, it will handle adjusting the volume, so we won't below
             adjustVolume = checkForRingerModeChange(oldIndex, direction);
         }
 
         if (adjustVolume && streamState.adjustIndex(direction)) {
-
-            boolean alsoUpdateNotificationVolume =  notificationsUseRingVolume &&
-                    streamType == AudioManager.STREAM_RING;
-            if (alsoUpdateNotificationVolume) {
-                mStreamStates[AudioManager.STREAM_NOTIFICATION].adjustIndex(direction);
-            }
-
             // Post message to set system volume (it in turn will post a message
             // to persist). Do not change volume if stream is muted.
             if (streamState.muteCount() == 0) {
-                sendMsg(mAudioHandler, MSG_SET_SYSTEM_VOLUME, streamType, SENDMSG_NOOP, 0, 0,
+                sendMsg(mAudioHandler, MSG_SET_SYSTEM_VOLUME, STREAM_VOLUME_ALIAS[streamType], SENDMSG_NOOP, 0, 0,
                         streamState, 0);
-
-                if (alsoUpdateNotificationVolume) {
-                    sendMsg(mAudioHandler, MSG_SET_SYSTEM_VOLUME, AudioManager.STREAM_NOTIFICATION,
-                            SENDMSG_NOOP, 0, 0, mStreamStates[AudioManager.STREAM_NOTIFICATION], 0);
-                }
             }
         }
 
@@ -404,9 +375,8 @@
     /** @see AudioManager#setStreamVolume(int, int, int) */
     public void setStreamVolume(int streamType, int index, int flags) {
         ensureValidStreamType(streamType);
-        syncRingerAndNotificationStreamVolume(streamType, index, false);
-
-        setStreamVolumeInt(streamType, index, false, true);
+        index = rescaleIndex(index * 10, streamType, STREAM_VOLUME_ALIAS[streamType]);
+        setStreamVolumeInt(STREAM_VOLUME_ALIAS[streamType], index, false, true);
 
         // UI, etc.
         mVolumePanel.postVolumeChanged(streamType, flags);
@@ -420,37 +390,12 @@
         intent.putExtra(AudioManager.EXTRA_VOLUME_STREAM_VALUE, getStreamVolume(streamType));
 
         // Currently, sending the intent only when the stream is BLUETOOTH_SCO
-        if (streamType == AudioManager.STREAM_BLUETOOTH_SCO) {
+        if (streamType == AudioSystem.STREAM_BLUETOOTH_SCO) {
             mContext.sendBroadcast(intent);
         }
     }
 
     /**
-     * Sync the STREAM_RING and STREAM_NOTIFICATION volumes if mandated by the
-     * value in Settings.
-     *
-     * @param streamType Type of the stream
-     * @param index Volume index for the stream
-     * @param force If true, set the volume even if the current and desired
-     * volume as same
-     */
-    private void syncRingerAndNotificationStreamVolume(int streamType, int index, boolean force) {
-        boolean notificationsUseRingVolume = Settings.System.getInt(mContentResolver,
-                Settings.System.NOTIFICATIONS_USE_RING_VOLUME, 1) == 1;
-        if (notificationsUseRingVolume) {
-            if (streamType == AudioManager.STREAM_NOTIFICATION) {
-                // Redirect the volume change to the ring stream
-                streamType = AudioManager.STREAM_RING;
-            }
-            if (streamType == AudioManager.STREAM_RING) {
-                // One-off to sync notification volume to ringer volume
-                setStreamVolumeInt(AudioManager.STREAM_NOTIFICATION, index, force, true);
-            }
-        }
-    }
-
-
-    /**
      * Sets the stream state's index, and posts a message to set system volume.
      * This will not call out to the UI. Assumes a valid stream type.
      *
@@ -491,13 +436,13 @@
     /** @see AudioManager#getStreamVolume(int) */
     public int getStreamVolume(int streamType) {
         ensureValidStreamType(streamType);
-        return mStreamStates[streamType].mIndex;
+        return (mStreamStates[streamType].mIndex + 5) / 10;
     }
 
     /** @see AudioManager#getStreamMaxVolume(int) */
     public int getStreamMaxVolume(int streamType) {
         ensureValidStreamType(streamType);
-        return mStreamStates[streamType].getMaxIndex();
+        return (mStreamStates[streamType].getMaxIndex() + 5) / 10;
     }
 
     /** @see AudioManager#getRingerMode() */
@@ -507,11 +452,12 @@
 
     /** @see AudioManager#setRingerMode(int) */
     public void setRingerMode(int ringerMode) {
-        if (ringerMode != mRingerMode) {
-            setRingerModeInt(ringerMode, true);
-
-            // Send sticky broadcast
-            broadcastRingerMode();
+        synchronized (mSettingsLock) {
+            if (ringerMode != mRingerMode) {
+                setRingerModeInt(ringerMode, true);
+                // Send sticky broadcast
+                broadcastRingerMode();
+            }
         }
     }
 
@@ -541,7 +487,7 @@
                 }
             }
         }
-        
+
         // Post a persist ringer mode msg
         if (persist) {
             sendMsg(mAudioHandler, MSG_PERSIST_RINGER_MODE, SHARED_MSG,
@@ -606,39 +552,28 @@
         return existingValue;
     }
 
-    /** @see AudioManager#setMicrophoneMute(boolean) */
-    public void setMicrophoneMute(boolean on) {
-        if (!checkAudioSettingsPermission("setMicrophoneMute()")) {
-            return;
-        }
-        synchronized (mSettingsLock) {
-            if (on != mMicMute) {
-                AudioSystem.muteMicrophone(on);
-                mMicMute = on;
-            }
-        }
-    }
-
-    /** @see AudioManager#isMicrophoneMute() */
-    public boolean isMicrophoneMute() {
-        return mMicMute;
-    }
-
     /** @see AudioManager#setMode(int) */
     public void setMode(int mode) {
         if (!checkAudioSettingsPermission("setMode()")) {
             return;
         }
+
+        if (mode < AudioSystem.MODE_CURRENT || mode > AudioSystem.MODE_IN_CALL) {
+            return;
+        }
+
         synchronized (mSettingsLock) {
+            if (mode == AudioSystem.MODE_CURRENT) {
+                mode = mMode;
+            }
             if (mode != mMode) {
-                if (AudioSystem.setMode(mode) == AudioSystem.AUDIO_STATUS_OK) {
+                if (AudioSystem.setPhoneState(mode) == AudioSystem.AUDIO_STATUS_OK) {
                     mMode = mode;
                 }
             }
             int streamType = getActiveStreamType(AudioManager.USE_DEFAULT_STREAM_TYPE);
-            int index = mStreamStates[streamType].mIndex;
-            syncRingerAndNotificationStreamVolume(streamType, index, true);
-            setStreamVolumeInt(streamType, index, true, true);
+            int index = mStreamStates[STREAM_VOLUME_ALIAS[streamType]].mIndex;
+            setStreamVolumeInt(STREAM_VOLUME_ALIAS[streamType], index, true, true);
         }
     }
 
@@ -647,187 +582,6 @@
         return mMode;
     }
 
-    /** @see AudioManager#setRouting(int, int, int) */
-    public void setRouting(int mode, int routes, int mask) {
-        int incallMask = 0;
-        int ringtoneMask = 0;
-        int normalMask = 0;
-
-        if (!checkAudioSettingsPermission("setRouting()")) {
-            return;
-        }
-        synchronized (mSettingsLock) {
-            // Temporary fix for issue #1713090 until audio routing is refactored in eclair release.
-            // mode AudioSystem.MODE_INVALID is used only by the following AudioManager methods:
-            // setWiredHeadsetOn(), setBluetoothA2dpOn(), setBluetoothScoOn() and setSpeakerphoneOn().
-            // If applications are using AudioManager.setRouting() that is now deprecated, the routing
-            // command will be ignored.
-            if (mode == AudioSystem.MODE_INVALID) {
-                switch (mask) {
-                case AudioSystem.ROUTE_SPEAKER:
-                    // handle setSpeakerphoneOn()
-                    if (routes != 0 && !mSpeakerIsOn) {
-                        mSpeakerIsOn = true;
-                        mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_SPEAKER;
-                        incallMask = AudioSystem.ROUTE_ALL;
-                    } else if (routes == 0 && mSpeakerIsOn) {
-                        mSpeakerIsOn = false;
-                        if (mBluetoothScoIsConnected) {
-                            mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_BLUETOOTH_SCO;
-                        } else if (mHeadsetIsConnected) {
-                            mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_HEADSET;
-                        } else {
-                            mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_EARPIECE;
-                        }
-                        incallMask = AudioSystem.ROUTE_ALL;
-                    }
-                    break;
-
-                case AudioSystem.ROUTE_BLUETOOTH_SCO:
-                    // handle setBluetoothScoOn()
-                    if (routes != 0 && !mBluetoothScoIsConnected) {
-                        mBluetoothScoIsConnected = true;
-                        mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_BLUETOOTH_SCO;
-                        mRoutes[AudioSystem.MODE_RINGTONE] = (mRoutes[AudioSystem.MODE_RINGTONE] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                              AudioSystem.ROUTE_BLUETOOTH_SCO;
-                        mRoutes[AudioSystem.MODE_NORMAL] = (mRoutes[AudioSystem.MODE_NORMAL] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                            AudioSystem.ROUTE_BLUETOOTH_SCO;
-                        incallMask = AudioSystem.ROUTE_ALL;
-                        // A2DP has higher priority than SCO headset, so headset connect/disconnect events
-                        // should not affect A2DP routing
-                        ringtoneMask = AudioSystem.ROUTE_ALL & ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        normalMask = AudioSystem.ROUTE_ALL & ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                    } else if (routes == 0 && mBluetoothScoIsConnected) {
-                        mBluetoothScoIsConnected = false;
-                        if (mHeadsetIsConnected) {
-                            mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_HEADSET;
-                            mRoutes[AudioSystem.MODE_RINGTONE] = (mRoutes[AudioSystem.MODE_RINGTONE] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                                 (AudioSystem.ROUTE_HEADSET|AudioSystem.ROUTE_SPEAKER);
-                            mRoutes[AudioSystem.MODE_NORMAL] = (mRoutes[AudioSystem.MODE_NORMAL] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                               AudioSystem.ROUTE_HEADSET;
-                        } else {
-                            if (mSpeakerIsOn) {
-                                mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_SPEAKER;
-                            } else {
-                                mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_EARPIECE;
-                            }
-                            mRoutes[AudioSystem.MODE_RINGTONE] = (mRoutes[AudioSystem.MODE_RINGTONE] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                                 AudioSystem.ROUTE_SPEAKER;
-                            mRoutes[AudioSystem.MODE_NORMAL] = (mRoutes[AudioSystem.MODE_NORMAL] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                               AudioSystem.ROUTE_SPEAKER;
-                        }
-                        incallMask = AudioSystem.ROUTE_ALL;
-                        // A2DP has higher priority than SCO headset, so headset connect/disconnect events
-                        // should not affect A2DP routing
-                        ringtoneMask = AudioSystem.ROUTE_ALL & ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        normalMask = AudioSystem.ROUTE_ALL & ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                    }
-                    break;
-
-                case AudioSystem.ROUTE_HEADSET:
-                    // handle setWiredHeadsetOn()
-                    if (routes != 0 && !mHeadsetIsConnected) {
-                        mHeadsetIsConnected = true;
-                        // do not act upon headset connection if bluetooth SCO is connected to match phone app behavior
-                        if (!mBluetoothScoIsConnected) {
-                            mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_HEADSET;
-                            mRoutes[AudioSystem.MODE_RINGTONE] = (mRoutes[AudioSystem.MODE_RINGTONE] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                                 (AudioSystem.ROUTE_HEADSET|AudioSystem.ROUTE_SPEAKER);
-                            mRoutes[AudioSystem.MODE_NORMAL] = (mRoutes[AudioSystem.MODE_NORMAL] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                               AudioSystem.ROUTE_HEADSET;
-                            incallMask = AudioSystem.ROUTE_ALL;
-                            // A2DP has higher priority than wired headset, so headset connect/disconnect events
-                            // should not affect A2DP routing
-                            ringtoneMask = AudioSystem.ROUTE_ALL & ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                            normalMask = AudioSystem.ROUTE_ALL & ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        }
-                    } else if (routes == 0 && mHeadsetIsConnected) {
-                        mHeadsetIsConnected = false;
-                        // do not act upon headset disconnection if bluetooth SCO is connected to match phone app behavior
-                        if (!mBluetoothScoIsConnected) {
-                            if (mSpeakerIsOn) {
-                                mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_SPEAKER;
-                            } else {
-                                mRoutes[AudioSystem.MODE_IN_CALL] = AudioSystem.ROUTE_EARPIECE;
-                            }
-                            mRoutes[AudioSystem.MODE_RINGTONE] = (mRoutes[AudioSystem.MODE_RINGTONE] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                                 AudioSystem.ROUTE_SPEAKER;
-                            mRoutes[AudioSystem.MODE_NORMAL] = (mRoutes[AudioSystem.MODE_NORMAL] & AudioSystem.ROUTE_BLUETOOTH_A2DP) |
-                                                               AudioSystem.ROUTE_SPEAKER;
-
-                            incallMask = AudioSystem.ROUTE_ALL;
-                            // A2DP has higher priority than wired headset, so headset connect/disconnect events
-                            // should not affect A2DP routing
-                            ringtoneMask = AudioSystem.ROUTE_ALL & ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                            normalMask = AudioSystem.ROUTE_ALL & ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        }
-                    }
-                    break;
-
-                case AudioSystem.ROUTE_BLUETOOTH_A2DP:
-                    // handle setBluetoothA2dpOn()
-                    if (routes != 0 && !mBluetoothA2dpIsConnected) {
-                        mBluetoothA2dpIsConnected = true;
-                        mRoutes[AudioSystem.MODE_RINGTONE] |= AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        mRoutes[AudioSystem.MODE_NORMAL] |= AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        // the audio flinger chooses A2DP as a higher priority,
-                        // so there is no need to disable other routes.
-                        ringtoneMask = AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        normalMask = AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                    } else if (routes == 0 && mBluetoothA2dpIsConnected) {
-                        mBluetoothA2dpIsConnected = false;
-                        mRoutes[AudioSystem.MODE_RINGTONE] &= ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        mRoutes[AudioSystem.MODE_NORMAL] &= ~AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        // the audio flinger chooses A2DP as a higher priority,
-                        // so there is no need to disable other routes.
-                        ringtoneMask = AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                        normalMask = AudioSystem.ROUTE_BLUETOOTH_A2DP;
-                    }
-                    break;
-                }
-                
-                // incallMask is != 0 means we must apply ne routing to MODE_IN_CALL mode
-                if (incallMask != 0) {
-                    AudioSystem.setRouting(AudioSystem.MODE_IN_CALL,
-                                           mRoutes[AudioSystem.MODE_IN_CALL],
-                                           incallMask);
-                }
-                // ringtoneMask is != 0 means we must apply ne routing to MODE_RINGTONE mode
-                if (ringtoneMask != 0) {
-                    AudioSystem.setRouting(AudioSystem.MODE_RINGTONE,
-                                           mRoutes[AudioSystem.MODE_RINGTONE],
-                                           ringtoneMask);
-                }
-                // normalMask is != 0 means we must apply ne routing to MODE_NORMAL mode
-                if (normalMask != 0) {
-                    AudioSystem.setRouting(AudioSystem.MODE_NORMAL,
-                                           mRoutes[AudioSystem.MODE_NORMAL],
-                                           normalMask);
-                }
-
-                int streamType = getActiveStreamType(AudioManager.USE_DEFAULT_STREAM_TYPE);
-                int index = mStreamStates[streamType].mIndex;
-                syncRingerAndNotificationStreamVolume(streamType, index, true);
-                setStreamVolumeInt(streamType, index, true, true);
-            }
-        }
-    }
-
-    /** @see AudioManager#getRouting(int) */
-    public int getRouting(int mode) {
-        return mRoutes[mode];
-    }
-
-    /** @see AudioManager#isMusicActive() */
-    public boolean isMusicActive() {
-        return AudioSystem.isMusicActive();
-    }
-
-    /** @see AudioManager#setParameter(String, String) */
-    public void setParameter(String key, String value) {
-        AudioSystem.setParameter(key, value);
-    }
-
     /** @see AudioManager#playSoundEffect(int) */
     public void playSoundEffect(int effectType) {
         sendMsg(mAudioHandler, MSG_PLAY_SOUND_EFFECT, SHARED_MSG, SENDMSG_NOOP,
@@ -930,14 +684,28 @@
             if (streamType != AudioSystem.STREAM_BLUETOOTH_SCO) {
                 String settingName = System.VOLUME_SETTINGS[streamType];
                 String lastAudibleSettingName = settingName + System.APPEND_FOR_LAST_AUDIBLE;
+                int index = Settings.System.getInt(mContentResolver,
+                                               settingName,
+                                               AudioManager.DEFAULT_STREAM_VOLUME[streamType]);
+                if (STREAM_VOLUME_ALIAS[streamType] != streamType) {
+                    index = rescaleIndex(index * 10, STREAM_VOLUME_ALIAS[streamType], streamType);
+                } else {
+                    index *= 10;
+                }
+                streamState.mIndex = streamState.getValidIndex(index);
 
-                streamState.mIndex = streamState.getValidIndex(Settings.System.getInt(mContentResolver,
-                        settingName,
-                        AudioManager.DEFAULT_STREAM_VOLUME[streamType]));
-                streamState.mLastAudibleIndex = streamState.getValidIndex(Settings.System.getInt(mContentResolver,
-                        lastAudibleSettingName,
-                        streamState.mIndex > 0 ? streamState.mIndex : AudioManager.DEFAULT_STREAM_VOLUME[streamType]));
+                index = (index + 5) / 10;
+                index = Settings.System.getInt(mContentResolver,
+                                                lastAudibleSettingName,
+                                                (index > 0) ? index : AudioManager.DEFAULT_STREAM_VOLUME[streamType]);
+                if (STREAM_VOLUME_ALIAS[streamType] != streamType) {
+                    index = rescaleIndex(index * 10, STREAM_VOLUME_ALIAS[streamType], streamType);
+                } else {
+                    index *= 10;
+                }
+                streamState.mLastAudibleIndex = streamState.getValidIndex(index);
             }
+
             // unmute stream that whas muted but is not affect by mute anymore
             if (streamState.muteCount() != 0 && !isStreamAffectedByMute(streamType)) {
                 int size = streamState.mDeathHandlers.size();
@@ -948,7 +716,7 @@
             }
             // apply stream volume
             if (streamState.muteCount() == 0) {
-                AudioSystem.setVolume(streamType, streamState.mVolumes[streamState.mIndex]);
+                setStreamVolumeIndex(streamType, streamState.mIndex);
             }
         }
 
@@ -969,7 +737,7 @@
         boolean adjustVolumeIndex = true;
         int newRingerMode = mRingerMode;
 
-        if (mRingerMode == AudioManager.RINGER_MODE_NORMAL && oldIndex == 1
+        if (mRingerMode == AudioManager.RINGER_MODE_NORMAL && (oldIndex + 5) / 10 == 1
                 && direction == AudioManager.ADJUST_LOWER) {
             newRingerMode = AudioManager.RINGER_MODE_VIBRATE;
         } else if (mRingerMode == AudioManager.RINGER_MODE_VIBRATE) {
@@ -1026,7 +794,7 @@
             Log.w(TAG, "Couldn't connect to phone service", e);
         }
 
-        if ((getRouting(AudioSystem.MODE_IN_CALL) & AudioSystem.ROUTE_BLUETOOTH_SCO) != 0) {
+        if (AudioSystem.getForceUse(AudioSystem.FOR_COMMUNICATION) == AudioSystem.FORCE_BT_SCO) {
             // Log.v(TAG, "getActiveStreamType: Forcing STREAM_BLUETOOTH_SCO...");
             return AudioSystem.STREAM_BLUETOOTH_SCO;
         } else if (isOffhook) {
@@ -1110,47 +878,36 @@
         private final String mLastAudibleVolumeIndexSettingName;
         private final int mStreamType;
 
-        private final int[] mVolumes;
+        private int mIndexMax;
         private int mIndex;
         private int mLastAudibleIndex;
         private ArrayList<VolumeDeathHandler> mDeathHandlers; //handles mute/solo requests client death
 
-        private VolumeStreamState(String settingName, int streamType, int[] volumes) {
+        private VolumeStreamState(String settingName, int streamType) {
 
             mVolumeIndexSettingName = settingName;
             mLastAudibleVolumeIndexSettingName = settingName + System.APPEND_FOR_LAST_AUDIBLE;
 
             mStreamType = streamType;
-            mVolumes = volumes;
 
             final ContentResolver cr = mContentResolver;
-            mIndex = getValidIndex(Settings.System.getInt(cr, mVolumeIndexSettingName, AudioManager.DEFAULT_STREAM_VOLUME[streamType]));
-            mLastAudibleIndex = getValidIndex(Settings.System.getInt(cr,
-                    mLastAudibleVolumeIndexSettingName, mIndex > 0 ? mIndex : AudioManager.DEFAULT_STREAM_VOLUME[streamType]));
-
-            AudioSystem.setVolume(streamType, volumes[mIndex]);
-            mDeathHandlers = new ArrayList<VolumeDeathHandler>();
-        }
-
-        /**
-         * Constructor to be used when there is no setting associated with the VolumeStreamState.
-         *
-         * @param defaultVolume Default volume of the stream to use.
-         * @param streamType Type of the stream.
-         * @param volumes Volumes levels associated with this stream.
-         */
-        private VolumeStreamState(int defaultVolume, int streamType, int[] volumes) {
-            mVolumeIndexSettingName = null;
-            mLastAudibleVolumeIndexSettingName = null;
-            mIndex = mLastAudibleIndex = defaultVolume;
-            mStreamType = streamType;
-            mVolumes = volumes;
-            AudioSystem.setVolume(mStreamType, defaultVolume);
+            mIndexMax = AudioManager.MAX_STREAM_VOLUME[streamType];
+            mIndex = Settings.System.getInt(cr,
+                                            mVolumeIndexSettingName,
+                                            AudioManager.DEFAULT_STREAM_VOLUME[streamType]);
+            mLastAudibleIndex = Settings.System.getInt(cr,
+                                                       mLastAudibleVolumeIndexSettingName,
+                                                       (mIndex > 0) ? mIndex : AudioManager.DEFAULT_STREAM_VOLUME[streamType]);
+            AudioSystem.initStreamVolume(streamType, 0, mIndexMax);
+            mIndexMax *= 10;
+            mIndex = getValidIndex(10 * mIndex);
+            mLastAudibleIndex = getValidIndex(10 * mLastAudibleIndex);
+            setStreamVolumeIndex(streamType, mIndex);
             mDeathHandlers = new ArrayList<VolumeDeathHandler>();
         }
 
         public boolean adjustIndex(int deltaIndex) {
-            return setIndex(mIndex + deltaIndex, true);
+            return setIndex(mIndex + deltaIndex * 10, true);
         }
 
         public boolean setIndex(int index, boolean lastAudible) {
@@ -1161,6 +918,13 @@
                 if (lastAudible) {
                     mLastAudibleIndex = mIndex;
                 }
+                // Apply change to all streams using this one as alias
+                int numStreamTypes = AudioSystem.getNumStreamTypes();
+                for (int streamType = numStreamTypes - 1; streamType >= 0; streamType--) {
+                    if (streamType != mStreamType && STREAM_VOLUME_ALIAS[streamType] == mStreamType) {
+                        mStreamStates[streamType].setIndex(rescaleIndex(mIndex, mStreamType, streamType), lastAudible);
+                    }
+                }
                 return true;
             } else {
                 return false;
@@ -1168,7 +932,7 @@
         }
 
         public int getMaxIndex() {
-            return mVolumes.length - 1;
+            return mIndexMax;
         }
 
         public void mute(IBinder cb, boolean state) {
@@ -1183,8 +947,8 @@
         private int getValidIndex(int index) {
             if (index < 0) {
                 return 0;
-            } else if (index >= mVolumes.length) {
-                return mVolumes.length - 1;
+            } else if (index > mIndexMax) {
+                return mIndexMax;
             }
 
             return index;
@@ -1318,8 +1082,16 @@
         private void setSystemVolume(VolumeStreamState streamState) {
 
             // Adjust volume
-            AudioSystem
-                    .setVolume(streamState.mStreamType, streamState.mVolumes[streamState.mIndex]);
+            setStreamVolumeIndex(streamState.mStreamType, streamState.mIndex);
+
+            // Apply change to all streams using this one as alias
+            int numStreamTypes = AudioSystem.getNumStreamTypes();
+            for (int streamType = numStreamTypes - 1; streamType >= 0; streamType--) {
+                if (streamType != streamState.mStreamType &&
+                    STREAM_VOLUME_ALIAS[streamType] == streamState.mStreamType) {
+                    setStreamVolumeIndex(streamType, mStreamStates[streamType].mIndex);
+                }
+            }
 
             // Post a persist volume msg
             sendMsg(mAudioHandler, MSG_PERSIST_VOLUME, streamState.mStreamType,
@@ -1327,12 +1099,10 @@
         }
 
         private void persistVolume(VolumeStreamState streamState) {
-            if (streamState.mStreamType != AudioManager.STREAM_BLUETOOTH_SCO) {
-                System.putInt(mContentResolver, streamState.mVolumeIndexSettingName,
-                        streamState.mIndex);
-                System.putInt(mContentResolver, streamState.mLastAudibleVolumeIndexSettingName,
-                        streamState.mLastAudibleIndex);
-            }
+            System.putInt(mContentResolver, streamState.mVolumeIndexSettingName,
+                    (streamState.mIndex + 5)/ 10);
+            System.putInt(mContentResolver, streamState.mLastAudibleVolumeIndexSettingName,
+                    (streamState.mLastAudibleIndex + 5) / 10);
         }
 
         private void persistRingerMode() {
@@ -1426,18 +1196,17 @@
 
                 case MSG_MEDIA_SERVER_STARTED:
                     Log.e(TAG, "Media server started.");
-                    // Restore audio routing and stream volumes
-                    applyAudioSettings();
+                    // Restore stream volumes
                     int numStreamTypes = AudioSystem.getNumStreamTypes();
                     for (int streamType = numStreamTypes - 1; streamType >= 0; streamType--) {
-                        int volume;
+                        int index;
                         VolumeStreamState streamState = mStreamStates[streamType];
                         if (streamState.muteCount() == 0) {
-                            volume = streamState.mVolumes[streamState.mIndex];
+                            index = streamState.mIndex;
                         } else {
-                            volume = streamState.mVolumes[0];
+                            index = 0;
                         }
-                        AudioSystem.setVolume(streamType, volume);
+                        setStreamVolumeIndex(streamType, index);
                     }
                     setRingerMode(mRingerMode);
                     mMediaServerOk = true;
@@ -1451,28 +1220,144 @@
     }
 
     private class SettingsObserver extends ContentObserver {
-        
+
         SettingsObserver() {
             super(new Handler());
             mContentResolver.registerContentObserver(Settings.System.getUriFor(
                 Settings.System.MODE_RINGER_STREAMS_AFFECTED), false, this);
+            mContentResolver.registerContentObserver(Settings.System.getUriFor(
+                    Settings.System.NOTIFICATIONS_USE_RING_VOLUME), false, this);
         }
 
         @Override
         public void onChange(boolean selfChange) {
             super.onChange(selfChange);
-            
-            mRingerModeAffectedStreams = Settings.System.getInt(mContentResolver,
-                    Settings.System.MODE_RINGER_STREAMS_AFFECTED,
-                    0);
+            synchronized (mSettingsLock) {
+                int ringerModeAffectedStreams = Settings.System.getInt(mContentResolver,
+                        Settings.System.MODE_RINGER_STREAMS_AFFECTED,
+                        0);
+                if (ringerModeAffectedStreams != mRingerModeAffectedStreams) {
+                    /*
+                     * Ensure all stream types that should be affected by ringer mode
+                     * are in the proper state.
+                     */
+                    mRingerModeAffectedStreams = ringerModeAffectedStreams;
+                    setRingerModeInt(getRingerMode(), false);
+                }
 
-            /*
-             * Ensure all stream types that should be affected by ringer mode
-             * are in the proper state.
-             */
-            setRingerModeInt(getRingerMode(), false);
+                int notificationsUseRingVolume = Settings.System.getInt(mContentResolver,
+                        Settings.System.NOTIFICATIONS_USE_RING_VOLUME,
+                        1);
+                if (notificationsUseRingVolume != mNotificationsUseRingVolume) {
+                    mNotificationsUseRingVolume = notificationsUseRingVolume;
+                    if (mNotificationsUseRingVolume == 1) {
+                        STREAM_VOLUME_ALIAS[AudioSystem.STREAM_NOTIFICATION] = AudioSystem.STREAM_RING;
+                    } else {
+                        STREAM_VOLUME_ALIAS[AudioSystem.STREAM_NOTIFICATION] = AudioSystem.STREAM_NOTIFICATION;
+                        // Persist notification volume volume as it was not persisted while aliased to ring volume
+                        sendMsg(mAudioHandler, MSG_PERSIST_VOLUME, AudioSystem.STREAM_NOTIFICATION,
+                                SENDMSG_REPLACE, 0, 0, mStreamStates[AudioSystem.STREAM_NOTIFICATION], PERSIST_DELAY);
+                    }
+                }
+            }
         }
-        
     }
-    
+
+    /**
+     * Receiver for misc intent broadcasts the Phone app cares about.
+     */
+    private class AudioServiceBroadcastReceiver extends BroadcastReceiver {
+        @Override
+        public void onReceive(Context context, Intent intent) {
+            String action = intent.getAction();
+
+            if (action.equals(BluetoothA2dp.SINK_STATE_CHANGED_ACTION)) {
+                int state = intent.getIntExtra(BluetoothA2dp.SINK_STATE,
+                                               BluetoothA2dp.STATE_DISCONNECTED);
+                String address = intent.getStringExtra(BluetoothIntent.ADDRESS);
+                if (state == BluetoothA2dp.STATE_DISCONNECTED) {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_BLUETOOTH_A2DP,
+                            AudioSystem.DEVICE_STATE_UNAVAILABLE,
+                            address);
+                } else if (state == BluetoothA2dp.STATE_CONNECTED){
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_BLUETOOTH_A2DP,
+                                                         AudioSystem.DEVICE_STATE_AVAILABLE,
+                                                         address);
+                }
+            } else if (action.equals(BluetoothIntent.HEADSET_AUDIO_STATE_CHANGED_ACTION)) {
+                int state = intent.getIntExtra(BluetoothIntent.HEADSET_AUDIO_STATE,
+                                               BluetoothHeadset.STATE_ERROR);
+                String address = intent.getStringExtra(BluetoothIntent.ADDRESS);
+                if (state == BluetoothHeadset.AUDIO_STATE_DISCONNECTED) {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_BLUETOOTH_SCO,
+                                                         AudioSystem.DEVICE_STATE_UNAVAILABLE,
+                                                         address);
+                } else if (state == BluetoothHeadset.AUDIO_STATE_CONNECTED) {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_BLUETOOTH_SCO,
+                                                         AudioSystem.DEVICE_STATE_AVAILABLE,
+                                                         address);
+                }
+            } else if (action.equals(Intent.ACTION_HEADSET_PLUG)) {
+                int state = intent.getIntExtra("state", 0);
+                if ((state & BIT_HEADSET) == 0 &&
+                    (mHeadsetState & BIT_HEADSET) != 0) {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_WIRED_HEADSET,
+                            AudioSystem.DEVICE_STATE_UNAVAILABLE,
+                            "");
+                } else if ((state & BIT_HEADSET) != 0 &&
+                    (mHeadsetState & BIT_HEADSET) == 0)  {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_WIRED_HEADSET,
+                            AudioSystem.DEVICE_STATE_AVAILABLE,
+                            "");
+                }
+                if ((state & BIT_HEADSET_NO_MIC) == 0 &&
+                    (mHeadsetState & BIT_HEADSET_NO_MIC) != 0) {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_WIRED_HEADPHONE,
+                            AudioSystem.DEVICE_STATE_UNAVAILABLE,
+                            "");
+                } else if ((state & BIT_HEADSET_NO_MIC) != 0 &&
+                    (mHeadsetState & BIT_HEADSET_NO_MIC) == 0)  {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_WIRED_HEADPHONE,
+                            AudioSystem.DEVICE_STATE_AVAILABLE,
+                            "");
+                }
+                if ((state & BIT_TTY) == 0 &&
+                    (mHeadsetState & BIT_TTY) != 0) {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_TTY,
+                            AudioSystem.DEVICE_STATE_UNAVAILABLE,
+                            "");
+                } else if ((state & BIT_TTY) != 0 &&
+                    (mHeadsetState & BIT_TTY) == 0)  {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_TTY,
+                            AudioSystem.DEVICE_STATE_AVAILABLE,
+                            "");
+                }
+                if ((state & BIT_FM_HEADSET) == 0 &&
+                    (mHeadsetState & BIT_FM_HEADSET) != 0) {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_FM_HEADPHONE,
+                            AudioSystem.DEVICE_STATE_UNAVAILABLE,
+                            "");
+                } else if ((state & BIT_FM_HEADSET) != 0 &&
+                    (mHeadsetState & BIT_FM_HEADSET) == 0)  {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_FM_HEADPHONE,
+                            AudioSystem.DEVICE_STATE_AVAILABLE,
+                            "");
+                }
+                if ((state & BIT_FM_SPEAKER) == 0 &&
+                    (mHeadsetState & BIT_FM_SPEAKER) != 0) {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_FM_SPEAKER,
+                            AudioSystem.DEVICE_STATE_UNAVAILABLE,
+                            "");
+                } else if ((state & BIT_FM_SPEAKER) != 0 &&
+                    (mHeadsetState & BIT_FM_SPEAKER) == 0)  {
+                    AudioSystem.setDeviceConnectionState(AudioSystem.DEVICE_OUT_FM_SPEAKER,
+                            AudioSystem.DEVICE_STATE_AVAILABLE,
+                            "");
+                }
+                mHeadsetState = state;
+            }
+        }
+    }
+
+
 }
diff --git a/media/java/android/media/AudioSystem.java b/media/java/android/media/AudioSystem.java
index 5917ab90..d587f65 100644
--- a/media/java/android/media/AudioSystem.java
+++ b/media/java/android/media/AudioSystem.java
@@ -45,38 +45,21 @@
     public static final int STREAM_NOTIFICATION = 5;
     /* @hide The audio stream for phone calls when connected on bluetooth */
     public static final int STREAM_BLUETOOTH_SCO = 6;
+    /* @hide The audio stream for enforced system sounds in certain countries (e.g camera in Japan) */
+    public static final int STREAM_SYSTEM_ENFORCED = 7;
+    /* @hide The audio stream for DTMF tones */
+    public static final int STREAM_DTMF = 8;
+    /* @hide The audio stream for text to speech (TTS) */
+    public static final int STREAM_TTS = 9;
     /**
      * @deprecated Use {@link #numStreamTypes() instead}
      */
     public static final int NUM_STREAMS = 5;
 
     // Expose only the getter method publicly so we can change it in the future
-    private static final int NUM_STREAM_TYPES = 7;
+    private static final int NUM_STREAM_TYPES = 10;
     public static final int getNumStreamTypes() { return NUM_STREAM_TYPES; }
 
-    /* max and min volume levels */
-    /* Maximum volume setting, for use with setVolume(int,int) */
-    public static final int MAX_VOLUME = 100;
-    /* Minimum volume setting, for use with setVolume(int,int) */
-    public static final int MIN_VOLUME = 0;
-
-    /*
-     * Sets the volume of a specified audio stream.
-     *
-     * param type   the stream type to set the volume of (e.g. STREAM_MUSIC)
-     * param volume the volume level to set (0-100)
-     * return command completion status see AUDIO_STATUS_OK, see AUDIO_STATUS_ERROR
-     */
-    public static native int setVolume(int type, int volume);
-
-    /*
-     * Returns the volume of a specified audio stream.
-     *
-     * param type the stream type to get the volume of (e.g. STREAM_MUSIC)
-     * return the current volume (0-100)
-     */
-    public static native int getVolume(int type);
-
     /*
      * Sets the microphone mute on or off.
      *
@@ -101,17 +84,22 @@
      *              it can route the audio appropriately.
      * return command completion status see AUDIO_STATUS_OK, see AUDIO_STATUS_ERROR
      */
-    public static native int setMode(int mode);
-
+    /** @deprecated use {@link #setPhoneState(int)} */
+    public static int setMode(int mode) {
+        return AUDIO_STATUS_ERROR;
+    }
     /*
      * Returns the current audio mode.
      *
      * return      the current audio mode (NORMAL, RINGTONE, or IN_CALL).
      *              Returns the current current audio state from the HAL.
      */
-    public static native int getMode();
+    /** @deprecated */
+    public static int getMode() {
+        return MODE_INVALID;
+    }
 
-    /* modes for setMode/getMode/setRoute/getRoute */
+    /* modes for setPhoneState */
     public static final int MODE_INVALID            = -2;
     public static final int MODE_CURRENT            = -1;
     public static final int MODE_NORMAL             = 0;
@@ -121,15 +109,20 @@
 
 
     /* Routing bits for setRouting/getRouting API */
-    public static final int ROUTE_EARPIECE          = (1 << 0);
-    public static final int ROUTE_SPEAKER           = (1 << 1);
-
+    /** @deprecated */
+    @Deprecated public static final int ROUTE_EARPIECE          = (1 << 0);
+    /** @deprecated */
+    @Deprecated public static final int ROUTE_SPEAKER           = (1 << 1);
     /** @deprecated use {@link #ROUTE_BLUETOOTH_SCO} */
     @Deprecated public static final int ROUTE_BLUETOOTH = (1 << 2);
-    public static final int ROUTE_BLUETOOTH_SCO     = (1 << 2);
-    public static final int ROUTE_HEADSET           = (1 << 3);
-    public static final int ROUTE_BLUETOOTH_A2DP    = (1 << 4);
-    public static final int ROUTE_ALL               = 0xFFFFFFFF;
+    /** @deprecated */
+    @Deprecated public static final int ROUTE_BLUETOOTH_SCO     = (1 << 2);
+    /** @deprecated */
+    @Deprecated public static final int ROUTE_HEADSET           = (1 << 3);
+    /** @deprecated */
+    @Deprecated public static final int ROUTE_BLUETOOTH_A2DP    = (1 << 4);
+    /** @deprecated */
+    @Deprecated public static final int ROUTE_ALL               = 0xFFFFFFFF;
 
     /*
      * Sets the audio routing for a specified mode
@@ -141,7 +134,10 @@
      * ROUTE_xxx types. Unset bits indicate the route should be left unchanged
      * return command completion status see AUDIO_STATUS_OK, see AUDIO_STATUS_ERROR
      */
-    public static native int setRouting(int mode, int routes, int mask);
+    /** @deprecated use {@link #setDeviceConnectionState(int,int,String)} */
+    public static int setRouting(int mode, int routes, int mask) {
+        return AUDIO_STATUS_ERROR;
+    }
 
     /*
      * Returns the current audio routing bit vector for a specified mode.
@@ -150,7 +146,10 @@
      * return an audio route bit vector that can be compared with ROUTE_xxx
      * bits
      */
-    public static native int getRouting(int mode);
+    /** @deprecated use {@link #getDeviceConnectionState(int,String)} */
+    public static int getRouting(int mode) {
+        return 0;
+    }
 
     /*
      * Checks whether any music is active.
@@ -160,17 +159,23 @@
     public static native boolean isMusicActive();
 
     /*
-     * Sets a generic audio configuration parameter. The use of these parameters
+     * Sets a group generic audio configuration parameters. The use of these parameters
      * are platform dependant, see libaudio
      *
-     * ** Temporary interface - DO NOT USE
-     *
-     * TODO: Replace with a more generic key:value get/set mechanism
-     *
-     * param key   name of parameter to set. Must not be null.
-     * param value value of parameter. Must not be null.
+     * param keyValuePairs  list of parameters key value pairs in the form:
+     *    key1=value1;key2=value2;...
      */
-    public static native void setParameter(String key, String value);
+    public static native int setParameters(String keyValuePairs);
+
+    /*
+     * Gets a group generic audio configuration parameters. The use of these parameters
+     * are platform dependant, see libaudio
+     *
+     * param keys  list of parameters
+     * return value: list of parameters key value pairs in the form:
+     *    key1=value1;key2=value2;...
+     */
+    public static native String getParameters(String keys);
 
     /*
     private final static String TAG = "audio";
@@ -220,4 +225,68 @@
             mErrorCallback.onError(error);
         }
     }
+
+    /*
+     * AudioPolicyService methods
+     */
+
+    // output devices
+    public static final int DEVICE_OUT_EARPIECE = 0x1;
+    public static final int DEVICE_OUT_SPEAKER = 0x2;
+    public static final int DEVICE_OUT_WIRED_HEADSET = 0x4;
+    public static final int DEVICE_OUT_WIRED_HEADPHONE = 0x8;
+    public static final int DEVICE_OUT_BLUETOOTH_SCO = 0x10;
+    public static final int DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20;
+    public static final int DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40;
+    public static final int DEVICE_OUT_BLUETOOTH_A2DP = 0x80;
+    public static final int DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100;
+    public static final int DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200;
+    public static final int DEVICE_OUT_AUX_DIGITAL = 0x400;
+    public static final int DEVICE_OUT_FM_HEADPHONE = 0x800;
+    public static final int DEVICE_OUT_FM_SPEAKER = 0x1000;
+    public static final int DEVICE_OUT_TTY = 0x2000;
+    public static final int DEVICE_OUT_DEFAULT = 0x8000;
+    // input devices
+    public static final int DEVICE_IN_COMMUNICATION = 0x10000;
+    public static final int DEVICE_IN_AMBIENT = 0x20000;
+    public static final int DEVICE_IN_BUILTIN_MIC1 = 0x40000;
+    public static final int DEVICE_IN_BUILTIN_MIC2 = 0x80000;
+    public static final int DEVICE_IN_MIC_ARRAY = 0x100000;
+    public static final int DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x200000;
+    public static final int DEVICE_IN_WIRED_HEADSET = 0x400000;
+    public static final int DEVICE_IN_AUX_DIGITAL = 0x800000;
+    public static final int DEVICE_IN_DEFAULT = 0x80000000;
+
+    // device states
+    public static final int DEVICE_STATE_UNAVAILABLE = 0;
+    public static final int DEVICE_STATE_AVAILABLE = 1;
+
+    // phone state
+    public static final int PHONE_STATE_OFFCALL = 0;
+    public static final int PHONE_STATE_RINGING = 1;
+    public static final int PHONE_STATE_INCALL = 2;
+
+    // config for setForceUse
+    public static final int FORCE_NONE = 0;
+    public static final int FORCE_SPEAKER = 1;
+    public static final int FORCE_HEADPHONES = 2;
+    public static final int FORCE_BT_SCO = 3;
+    public static final int FORCE_BT_A2DP = 4;
+    public static final int FORCE_WIRED_ACCESSORY = 5;
+    public static final int FORCE_DEFAULT = FORCE_NONE;
+
+    // usage for serForceUse
+    public static final int FOR_COMMUNICATION = 0;
+    public static final int FOR_MEDIA = 1;
+    public static final int FOR_RECORD = 2;
+
+    public static native int setDeviceConnectionState(int device, int state, String device_address);
+    public static native int getDeviceConnectionState(int device, String device_address);
+    public static native int setPhoneState(int state);
+    public static native int setRingerMode(int mode, int mask);
+    public static native int setForceUse(int usage, int config);
+    public static native int getForceUse(int usage);
+    public static native int initStreamVolume(int stream, int indexMin, int indexMax);
+    public static native int setStreamVolumeIndex(int stream, int index);
+    public static native int getStreamVolumeIndex(int stream);
 }
diff --git a/media/java/android/media/AudioTrack.java b/media/java/android/media/AudioTrack.java
index 5f1be9d..7fbe965 100644
--- a/media/java/android/media/AudioTrack.java
+++ b/media/java/android/media/AudioTrack.java
@@ -120,7 +120,7 @@
     public  static final int ERROR_INVALID_OPERATION               = -3;
 
     private static final int ERROR_NATIVESETUP_AUDIOSYSTEM         = -16;
-    private static final int ERROR_NATIVESETUP_INVALIDCHANNELCOUNT = -17;
+    private static final int ERROR_NATIVESETUP_INVALIDCHANNELMASK  = -17;
     private static final int ERROR_NATIVESETUP_INVALIDFORMAT       = -18;
     private static final int ERROR_NATIVESETUP_INVALIDSTREAMTYPE   = -19;
     private static final int ERROR_NATIVESETUP_NATIVEINITFAILED    = -20;
@@ -198,7 +198,7 @@
     /**
      * The current audio channel configuration.
      */
-    private int mChannelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+    private int mChannelConfiguration = AudioFormat.CHANNEL_OUT_MONO;
     /**
      * The encoding of the audio samples.
      * @see AudioFormat#ENCODING_PCM_8BIT
@@ -235,8 +235,8 @@
      * @param sampleRateInHz the sample rate expressed in Hertz. Examples of rates are (but
      *   not limited to) 44100, 22050 and 11025.
      * @param channelConfig describes the configuration of the audio channels.
-     *   See {@link AudioFormat#CHANNEL_CONFIGURATION_MONO} and
-     *   {@link AudioFormat#CHANNEL_CONFIGURATION_STEREO}
+     *   See {@link AudioFormat#CHANNEL_OUT_MONO} and
+     *   {@link AudioFormat#CHANNEL_OUT_STEREO}
      * @param audioFormat the format in which the audio data is represented.
      *   See {@link AudioFormat#ENCODING_PCM_16BIT} and
      *   {@link AudioFormat#ENCODING_PCM_8BIT}
@@ -298,7 +298,8 @@
            && (streamType != AudioManager.STREAM_RING) && (streamType != AudioManager.STREAM_SYSTEM)
            && (streamType != AudioManager.STREAM_VOICE_CALL)
            && (streamType != AudioManager.STREAM_NOTIFICATION)
-           && (streamType != AudioManager.STREAM_BLUETOOTH_SCO)) {
+           && (streamType != AudioManager.STREAM_BLUETOOTH_SCO)
+           && (streamType != AudioManager.STREAM_DTMF)) {
             throw (new IllegalArgumentException("Invalid stream type."));
         } else {
             mStreamType = streamType;
@@ -316,18 +317,18 @@
         //--------------
         // channel config
         switch (channelConfig) {
-        case AudioFormat.CHANNEL_CONFIGURATION_DEFAULT:
-        case AudioFormat.CHANNEL_CONFIGURATION_MONO:
+        case AudioFormat.CHANNEL_OUT_DEFAULT:
+        case AudioFormat.CHANNEL_OUT_MONO:
             mChannelCount = 1;
-            mChannelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+            mChannelConfiguration = AudioFormat.CHANNEL_OUT_MONO;
             break;
-        case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
+        case AudioFormat.CHANNEL_OUT_STEREO:
             mChannelCount = 2;
-            mChannelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+            mChannelConfiguration = AudioFormat.CHANNEL_OUT_STEREO;
             break;
         default:
             mChannelCount = 0;
-            mChannelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_INVALID;
+            mChannelConfiguration = AudioFormat.CHANNEL_INVALID;
             throw(new IllegalArgumentException("Unsupported channel configuration."));
         }
 
@@ -452,8 +453,8 @@
     /**
      * Returns the configured channel configuration.
 
-     * See {@link AudioFormat#CHANNEL_CONFIGURATION_MONO}
-     * and {@link AudioFormat#CHANNEL_CONFIGURATION_STEREO}.
+     * See {@link AudioFormat#CHANNEL_OUT_MONO}
+     * and {@link AudioFormat#CHANNEL_OUT_STEREO}.
      */
     public int getChannelConfiguration() {
         return mChannelConfiguration;
@@ -531,8 +532,8 @@
      * the expected frequency at which the buffer will be refilled with additional data to play. 
      * @param sampleRateInHz the sample rate expressed in Hertz.
      * @param channelConfig describes the configuration of the audio channels. 
-     *   See {@link AudioFormat#CHANNEL_CONFIGURATION_MONO} and
-     *   {@link AudioFormat#CHANNEL_CONFIGURATION_STEREO}
+     *   See {@link AudioFormat#CHANNEL_OUT_MONO} and
+     *   {@link AudioFormat#CHANNEL_OUT_STEREO}
      * @param audioFormat the format in which the audio data is represented. 
      *   See {@link AudioFormat#ENCODING_PCM_16BIT} and 
      *   {@link AudioFormat#ENCODING_PCM_8BIT}
@@ -544,10 +545,10 @@
     static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
         int channelCount = 0;
         switch(channelConfig) {
-        case AudioFormat.CHANNEL_CONFIGURATION_MONO:
+        case AudioFormat.CHANNEL_OUT_MONO:
             channelCount = 1;
             break;
-        case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
+        case AudioFormat.CHANNEL_OUT_STEREO:
             channelCount = 2;
             break;
         default:
diff --git a/media/java/android/media/IAudioService.aidl b/media/java/android/media/IAudioService.aidl
index 9a8264f..bb4252b 100644
--- a/media/java/android/media/IAudioService.aidl
+++ b/media/java/android/media/IAudioService.aidl
@@ -29,9 +29,9 @@
     
     void setStreamVolume(int streamType, int index, int flags);
     
-   	void setStreamSolo(int streamType, boolean state, IBinder cb);
+    void setStreamSolo(int streamType, boolean state, IBinder cb);
    	
-   	void setStreamMute(int streamType, boolean state, IBinder cb);
+    void setStreamMute(int streamType, boolean state, IBinder cb);
     
     int getStreamVolume(int streamType);
     
@@ -46,23 +46,11 @@
     int getVibrateSetting(int vibrateType);
     
     boolean shouldVibrate(int vibrateType);
-    
-    void setMicrophoneMute(boolean on);
-
-    boolean isMicrophoneMute();
 
     void setMode(int mode);
 
     int getMode();
 
-    void setRouting(int mode, int routes, int mask);
-
-    int getRouting(int mode);
-
-    boolean isMusicActive();
-
-    void setParameter(String key, String value);
-
     oneway void playSoundEffect(int effectType);
   
     oneway void playSoundEffectVolume(int effectType, float volume);
diff --git a/media/java/android/media/JetPlayer.java b/media/java/android/media/JetPlayer.java
index 4fb0ead..2263605 100644
--- a/media/java/android/media/JetPlayer.java
+++ b/media/java/android/media/JetPlayer.java
@@ -89,7 +89,7 @@
     // Jet rendering audio parameters
     private static final int JET_OUTPUT_RATE = 22050; // _SAMPLE_RATE_22050 in Android.mk
     private static final int JET_OUTPUT_CHANNEL_CONFIG =
-            AudioFormat.CHANNEL_CONFIGURATION_STEREO; // NUM_OUTPUT_CHANNELS=2 in Android.mk
+            AudioFormat.CHANNEL_OUT_STEREO; // NUM_OUTPUT_CHANNELS=2 in Android.mk
 
     
     //--------------------------------------------
diff --git a/media/java/android/media/ToneGenerator.java b/media/java/android/media/ToneGenerator.java
index e5ee9a3..c60a1ac 100644
--- a/media/java/android/media/ToneGenerator.java
+++ b/media/java/android/media/ToneGenerator.java
@@ -724,9 +724,9 @@
     public static final int TONE_CDMA_SIGNAL_OFF = 98;
 
     /** Maximum volume, for use with {@link #ToneGenerator(int,int)} */
-    public static final int MAX_VOLUME = AudioSystem.MAX_VOLUME;
+    public static final int MAX_VOLUME = 100;
     /** Minimum volume setting, for use with {@link #ToneGenerator(int,int)} */
-    public static final int MIN_VOLUME = AudioSystem.MIN_VOLUME;
+    public static final int MIN_VOLUME = 0;
 
 
     /**
diff --git a/media/jni/soundpool/SoundPool.cpp b/media/jni/soundpool/SoundPool.cpp
index 1fdecdd..0d07abe 100644
--- a/media/jni/soundpool/SoundPool.cpp
+++ b/media/jni/soundpool/SoundPool.cpp
@@ -527,13 +527,14 @@
     // wrong audio audio buffer size  (mAudioBufferSize)
     unsigned long toggle = mToggle ^ 1;
     void *userData = (void *)((unsigned long)this | toggle);
+    uint32_t channels = (numChannels == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO;
     
 #ifdef USE_SHARED_MEM_BUFFER
     newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
-            numChannels, sample->getIMemory(), 0, callback, userData);
+            channels, sample->getIMemory(), 0, callback, userData);
 #else
     newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
-            numChannels, frameCount, 0, callback, userData, bufferFrames);
+            channels, frameCount, 0, callback, userData, bufferFrames);
 #endif
     if (newTrack->initCheck() != NO_ERROR) {
         LOGE("Error creating AudioTrack");
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 07c81f7..9d442c3 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -20,7 +20,8 @@
     mediametadataretriever.cpp \
     ToneGenerator.cpp \
     JetPlayer.cpp \
-    IOMX.cpp
+    IOMX.cpp \
+ 	IAudioPolicyService.cpp
 
 LOCAL_SHARED_LIBRARIES := \
 	libui libcutils libutils libbinder libsonivox
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 0a6f4f7..5e35564 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -28,6 +28,7 @@
 
 #include <media/AudioSystem.h>
 #include <media/AudioRecord.h>
+#include <media/mediarecorder.h>
 
 #include <binder/IServiceManager.h>
 #include <utils/Log.h>
@@ -45,7 +46,7 @@
 // ---------------------------------------------------------------------------
 
 AudioRecord::AudioRecord()
-    : mStatus(NO_INIT)
+    : mStatus(NO_INIT), mInput(0)
 {
 }
 
@@ -53,15 +54,15 @@
         int inputSource,
         uint32_t sampleRate,
         int format,
-        int channelCount,
+        uint32_t channels,
         int frameCount,
         uint32_t flags,
         callback_t cbf,
         void* user,
         int notificationFrames)
-    : mStatus(NO_INIT)
+    : mStatus(NO_INIT), mInput(0)
 {
-    mStatus = set(inputSource, sampleRate, format, channelCount,
+    mStatus = set(inputSource, sampleRate, format, channels,
             frameCount, flags, cbf, user, notificationFrames);
 }
 
@@ -78,6 +79,7 @@
         }
         mAudioRecord.clear();
         IPCThreadState::self()->flushCommands();
+        AudioSystem::releaseInput(mInput);
     }
 }
 
@@ -85,7 +87,7 @@
         int inputSource,
         uint32_t sampleRate,
         int format,
-        int channelCount,
+        uint32_t channels,
         int frameCount,
         uint32_t flags,
         callback_t cbf,
@@ -94,7 +96,7 @@
         bool threadCanCallJava)
 {
 
-    LOGV("set(): sampleRate %d, channelCount %d, frameCount %d",sampleRate, channelCount, frameCount);
+    LOGV("set(): sampleRate %d, channels %d, frameCount %d",sampleRate, channels, frameCount);
     if (mAudioRecord != 0) {
         return INVALID_OPERATION;
     }
@@ -104,8 +106,8 @@
         return NO_INIT;
     }
 
-    if (inputSource == DEFAULT_INPUT) {
-        inputSource = MIC_INPUT;
+    if (inputSource == AUDIO_SOURCE_DEFAULT) {
+        inputSource = AUDIO_SOURCE_MIC;
     }
 
     if (sampleRate == 0) {
@@ -115,15 +117,21 @@
     if (format == 0) {
         format = AudioSystem::PCM_16_BIT;
     }
-    if (channelCount == 0) {
-        channelCount = 1;
-    }
-
     // validate parameters
-    if (format != AudioSystem::PCM_16_BIT) {
+    if (!AudioSystem::isValidFormat(format)) {
+        LOGE("Invalid format");
         return BAD_VALUE;
     }
-    if (channelCount != 1 && channelCount != 2) {
+
+    if (!AudioSystem::isInputChannel(channels)) {
+        return BAD_VALUE;
+    }
+    int channelCount = AudioSystem::popCount(channels);
+
+    mInput = AudioSystem::getInput(inputSource,
+                                    sampleRate, format, channels, (AudioSystem::audio_in_acoustics)flags);
+    if (mInput == 0) {
+        LOGE("Could not get audio output for stream type %d", inputSource);
         return BAD_VALUE;
     }
 
@@ -132,14 +140,22 @@
     if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &inputBuffSizeInBytes)
             != NO_ERROR) {
         LOGE("AudioSystem could not query the input buffer size.");
-        return NO_INIT;    
+        return NO_INIT;
     }
+
     if (inputBuffSizeInBytes == 0) {
         LOGE("Recording parameters are not supported: sampleRate %d, channelCount %d, format %d",
             sampleRate, channelCount, format);
         return BAD_VALUE;
     }
+
     int frameSizeInBytes = channelCount * (format == AudioSystem::PCM_16_BIT ? 2 : 1);
+    if (AudioSystem::isLinearPCM(format)) {
+        frameSizeInBytes = channelCount * (format == AudioSystem::PCM_16_BIT ? sizeof(int16_t) : sizeof(int8_t));
+    } else {
+        frameSizeInBytes = sizeof(int8_t);
+    }
+
 
     // We use 2* size of input buffer for ping pong use of record buffer.
     int minFrameCount = 2 * inputBuffSizeInBytes / frameSizeInBytes;
@@ -157,11 +173,11 @@
 
     // open record channel
     status_t status;
-    sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), inputSource,
+    sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), mInput,
                                                        sampleRate, format,
                                                        channelCount,
                                                        frameCount,
-                                                       ((uint16_t)flags) << 16, 
+                                                       ((uint16_t)flags) << 16,
                                                        &status);
     if (record == 0) {
         LOGE("AudioFlinger could not create record track, status: %d", status);
@@ -188,7 +204,7 @@
     mFormat = format;
     // Update buffer size in case it has been limited by AudioFlinger during track creation
     mFrameCount = mCblk->frameCount;
-    mChannelCount = channelCount;
+    mChannelCount = (uint8_t)channelCount;
     mActive = 0;
     mCbf = cbf;
     mNotificationFrames = notificationFrames;
@@ -234,7 +250,11 @@
 
 int AudioRecord::frameSize() const
 {
-    return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+    if (AudioSystem::isLinearPCM(mFormat)) {
+        return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+    } else {
+        return sizeof(uint8_t);
+    }
 }
 
 int AudioRecord::inputSource() const
@@ -262,15 +282,18 @@
      }
 
     if (android_atomic_or(1, &mActive) == 0) {
-        mNewPosition = mCblk->user + mUpdatePeriod;
-        mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
-        mCblk->waitTimeMs = 0;
-        if (t != 0) {
-           t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
-        } else {
-            setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+        ret = AudioSystem::startInput(mInput);
+        if (ret == NO_ERROR) {
+            mNewPosition = mCblk->user + mUpdatePeriod;
+            mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+            mCblk->waitTimeMs = 0;
+            if (t != 0) {
+               t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
+            } else {
+                setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+            }
+            ret = mAudioRecord->start();
         }
-        ret = mAudioRecord->start();
     }
 
     if (t != 0) {
@@ -301,6 +324,7 @@
         } else {
             setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
         }
+        AudioSystem::stopInput(mInput);
     }
 
     if (t != 0) {
@@ -421,7 +445,7 @@
         "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
 
     cblk->waitTimeMs = 0;
-    
+
     if (framesReq > framesReady) {
         framesReq = framesReady;
     }
@@ -437,7 +461,7 @@
     audioBuffer->channelCount= mChannelCount;
     audioBuffer->format      = mFormat;
     audioBuffer->frameCount  = framesReq;
-    audioBuffer->size        = framesReq*mChannelCount*sizeof(int16_t);
+    audioBuffer->size        = framesReq*cblk->frameSize;
     audioBuffer->raw         = (int8_t*)cblk->buffer(u);
     active = mActive;
     return active ? status_t(NO_ERROR) : status_t(STOPPED);
@@ -468,7 +492,7 @@
 
     do {
 
-        audioBuffer.frameCount = userSize/mChannelCount/sizeof(int16_t);
+        audioBuffer.frameCount = userSize/frameSize();
 
         // Calling obtainBuffer() with a negative wait count causes
         // an (almost) infinite wait time.
@@ -519,8 +543,8 @@
 
     do {
         audioBuffer.frameCount = frames;
-        // Calling obtainBuffer() with a wait count of 1 
-        // limits wait time to WAIT_PERIOD_MS. This prevents from being 
+        // Calling obtainBuffer() with a wait count of 1
+        // limits wait time to WAIT_PERIOD_MS. This prevents from being
         // stuck here not being able to handle timed events (position, markers).
         status_t err = obtainBuffer(&audioBuffer, 1);
         if (err < NO_ERROR) {
@@ -548,14 +572,14 @@
         if (readSize > reqSize) readSize = reqSize;
 
         audioBuffer.size = readSize;
-        audioBuffer.frameCount = readSize/mChannelCount/sizeof(int16_t);
+        audioBuffer.frameCount = readSize/frameSize();
         frames -= audioBuffer.frameCount;
 
         releaseBuffer(&audioBuffer);
 
     } while (frames);
 
-    
+
     // Manage overrun callback
     if (mActive && (mCblk->framesAvailable_l() == 0)) {
         LOGV("Overrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 86d0542..1fc1024 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -20,8 +20,18 @@
 #include <utils/Log.h>
 #include <binder/IServiceManager.h>
 #include <media/AudioSystem.h>
+#include <media/IAudioPolicyService.h>
 #include <math.h>
 
+// ----------------------------------------------------------------------------
+// the sim build doesn't have gettid
+
+#ifndef HAVE_GETTID
+# define gettid getpid
+#endif
+
+// ----------------------------------------------------------------------------
+
 namespace android {
 
 // client singleton for AudioFlinger binder interface
@@ -30,10 +40,9 @@
 sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
 audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
 // Cached values
-int AudioSystem::gOutSamplingRate[NUM_AUDIO_OUTPUT_TYPES];
-int AudioSystem::gOutFrameCount[NUM_AUDIO_OUTPUT_TYPES];
-uint32_t AudioSystem::gOutLatency[NUM_AUDIO_OUTPUT_TYPES];
-bool AudioSystem::gA2dpEnabled;
+DefaultKeyedVector<int, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
+DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
+
 // Cached values for recording queries
 uint32_t AudioSystem::gPrevInSamplingRate = 16000;
 int AudioSystem::gPrevInFormat = AudioSystem::PCM_16_BIT;
@@ -65,44 +74,12 @@
         binder->linkToDeath(gAudioFlingerClient);
         gAudioFlinger = interface_cast<IAudioFlinger>(binder);
         gAudioFlinger->registerClient(gAudioFlingerClient);
-        // Cache frequently accessed parameters 
-        for (int output = 0; output < NUM_AUDIO_OUTPUT_TYPES; output++) {
-            gOutFrameCount[output] = (int)gAudioFlinger->frameCount(output);
-            gOutSamplingRate[output] = (int)gAudioFlinger->sampleRate(output);
-            gOutLatency[output] = gAudioFlinger->latency(output);
-        }
-        gA2dpEnabled = gAudioFlinger->isA2dpEnabled();
     }
     LOGE_IF(gAudioFlinger==0, "no AudioFlinger!?");
+
     return gAudioFlinger;
 }
 
-// routing helper functions
-status_t AudioSystem::speakerphone(bool state) {
-    uint32_t routes = state ? ROUTE_SPEAKER : ROUTE_EARPIECE;
-    return setRouting(MODE_IN_CALL, routes, ROUTE_ALL);
-}
-
-status_t AudioSystem::isSpeakerphoneOn(bool* state) {
-    uint32_t routes = 0;
-    status_t s = getRouting(MODE_IN_CALL, &routes);
-    *state = !!(routes & ROUTE_SPEAKER);
-    return s;
-}
-
-status_t AudioSystem::bluetoothSco(bool state) {
-    uint32_t mask = ROUTE_BLUETOOTH_SCO;
-    uint32_t routes = state ? mask : ROUTE_EARPIECE;
-    return setRouting(MODE_IN_CALL, routes, ROUTE_ALL);
-}
-
-status_t AudioSystem::isBluetoothScoOn(bool* state) {
-    uint32_t routes = 0;
-    status_t s = getRouting(MODE_IN_CALL, &routes);
-    *state = !!(routes & ROUTE_BLUETOOTH_SCO);
-    return s;
-}
-
 status_t AudioSystem::muteMicrophone(bool state) {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
@@ -148,12 +125,12 @@
     return NO_ERROR;
 }
 
-status_t AudioSystem::setStreamVolume(int stream, float value)
+status_t AudioSystem::setStreamVolume(int stream, float value, void *output)
 {
     if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
-    af->setStreamVolume(stream, value);
+    af->setStreamVolume(stream, value, output);
     return NO_ERROR;
 }
 
@@ -166,12 +143,12 @@
     return NO_ERROR;
 }
 
-status_t AudioSystem::getStreamVolume(int stream, float* volume)
+status_t AudioSystem::getStreamVolume(int stream, float* volume, void *output)
 {
     if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
-    *volume = af->streamVolume(stream);
+    *volume = af->streamVolume(stream, output);
     return NO_ERROR;
 }
 
@@ -192,29 +169,6 @@
     return af->setMode(mode);
 }
 
-status_t AudioSystem::getMode(int* mode)
-{
-    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-    if (af == 0) return PERMISSION_DENIED;
-    *mode = af->getMode();
-    return NO_ERROR;
-}
-
-status_t AudioSystem::setRouting(int mode, uint32_t routes, uint32_t mask)
-{
-    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-    if (af == 0) return PERMISSION_DENIED;
-    return af->setRouting(mode, routes, mask);
-}
-
-status_t AudioSystem::getRouting(int mode, uint32_t* routes)
-{
-    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-    if (af == 0) return PERMISSION_DENIED;
-    uint32_t r = af->getRouting(mode);
-    *routes = r;
-    return NO_ERROR;
-}
 
 status_t AudioSystem::isMusicActive(bool* state) {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -223,12 +177,20 @@
     return NO_ERROR;
 }
 
-// Temporary interface, do not use
-// TODO: Replace with a more generic key:value get/set mechanism
-status_t AudioSystem::setParameter(const char* key, const char* value) {
+
+status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
-    return af->setParameter(key, value);
+    return af->setParameters(ioHandle, keyValuePairs);
+}
+
+String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys) {
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    String8 result = String8("");
+    if (af == 0) return result;
+
+    result = af->getParameters(ioHandle, keys);
+    return result;
 }
 
 // convert volume steps to natural log scale
@@ -257,55 +219,108 @@
 
 status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
 {
-    int output = getOutput(streamType);
-    
-    if (output == NUM_AUDIO_OUTPUT_TYPES) return PERMISSION_DENIED;
+    OutputDescriptor *outputDesc;
+    audio_io_handle_t output;
 
-    // gOutSamplingRate[] is updated by getOutput() which calls get_audio_flinger()
-    LOGV("getOutputSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, gOutSamplingRate[output]);
-    
-    *samplingRate = gOutSamplingRate[output];
-    
+    if (streamType == DEFAULT) {
+        streamType = MUSIC;
+    }
+
+    output = getOutput((stream_type)streamType);
+    if (output == 0) {
+        return PERMISSION_DENIED;
+    }
+
+    gLock.lock();
+    outputDesc = AudioSystem::gOutputs.valueFor(output);
+    if (outputDesc == 0) {
+        LOGV("getOutputSamplingRate() no output descriptor for output %p in gOutputs", output);
+        gLock.unlock();
+        const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+        if (af == 0) return PERMISSION_DENIED;
+        *samplingRate = af->sampleRate(output);
+    } else {
+        LOGV("getOutputSamplingRate() reading from output desc");
+        *samplingRate = outputDesc->samplingRate;
+        gLock.unlock();
+    }
+
+    LOGV("getOutputSamplingRate() streamType %d, output %p, sampling rate %d", streamType, output, *samplingRate);
+
     return NO_ERROR;
 }
 
 status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
 {
-    int output = getOutput(streamType);
+    OutputDescriptor *outputDesc;
+    audio_io_handle_t output;
 
-    if (output == NUM_AUDIO_OUTPUT_TYPES) return PERMISSION_DENIED;
+    if (streamType == DEFAULT) {
+        streamType = MUSIC;
+    }
 
-    // gOutFrameCount[] is updated by getOutput() which calls get_audio_flinger()
-    LOGV("getOutputFrameCount() streamType %d, output %d, frame count %d", streamType, output, gOutFrameCount[output]);
+    output = getOutput((stream_type)streamType);
+    if (output == 0) {
+        return PERMISSION_DENIED;
+    }
 
-    *frameCount = gOutFrameCount[output];
-    
+    gLock.lock();
+    outputDesc = AudioSystem::gOutputs.valueFor(output);
+    if (outputDesc == 0) {
+        gLock.unlock();
+        const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+        if (af == 0) return PERMISSION_DENIED;
+        *frameCount = af->frameCount(output);
+    } else {
+        *frameCount = outputDesc->frameCount;
+        gLock.unlock();
+    }
+
+    LOGV("getOutputFrameCount() streamType %d, output %p, frameCount %d", streamType, output, *frameCount);
+
     return NO_ERROR;
 }
 
 status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
 {
-    int output = getOutput(streamType);
+    OutputDescriptor *outputDesc;
+    audio_io_handle_t output;
 
-    if (output == NUM_AUDIO_OUTPUT_TYPES) return PERMISSION_DENIED;
+    if (streamType == DEFAULT) {
+        streamType = MUSIC;
+    }
 
-    // gOutLatency[] is updated by getOutput() which calls get_audio_flinger()
-    LOGV("getOutputLatency() streamType %d, output %d, latency %d", streamType, output, gOutLatency[output]);
+    output = getOutput((stream_type)streamType);
+    if (output == 0) {
+        return PERMISSION_DENIED;
+    }
 
-    *latency = gOutLatency[output];
-    
+    gLock.lock();
+    outputDesc = AudioSystem::gOutputs.valueFor(output);
+    if (outputDesc == 0) {
+        gLock.unlock();
+        const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+        if (af == 0) return PERMISSION_DENIED;
+        *latency = af->latency(output);
+    } else {
+        *latency = outputDesc->latency;
+        gLock.unlock();
+    }
+
+    LOGV("getOutputLatency() streamType %d, output %p, latency %d", streamType, output, *latency);
+
     return NO_ERROR;
 }
 
-status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount, 
+status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
     size_t* buffSize)
 {
     // Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
-    if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat) 
+    if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
         || (channelCount != gPrevInChannelCount)) {
         // save the request params
         gPrevInSamplingRate = sampleRate;
-        gPrevInFormat = format; 
+        gPrevInFormat = format;
         gPrevInChannelCount = channelCount;
 
         gInBuffSize = 0;
@@ -314,24 +329,18 @@
             return PERMISSION_DENIED;
         }
         gInBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
-    } 
+    }
     *buffSize = gInBuffSize;
-    
+
     return NO_ERROR;
 }
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {   
+void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {
     Mutex::Autolock _l(AudioSystem::gLock);
-    AudioSystem::gAudioFlinger.clear();
 
-    for (int output = 0; output < NUM_AUDIO_OUTPUT_TYPES; output++) {
-        gOutFrameCount[output] = 0;
-        gOutSamplingRate[output] = 0;
-        gOutLatency[output] = 0;
-    }
-    AudioSystem::gInBuffSize = 0;
+    AudioSystem::gAudioFlinger.clear();
 
     if (gAudioErrorCallback) {
         gAudioErrorCallback(DEAD_OBJECT);
@@ -339,33 +348,83 @@
     LOGW("AudioFlinger server died!");
 }
 
-void AudioSystem::AudioFlingerClient::a2dpEnabledChanged(bool enabled) {
-    gA2dpEnabled = enabled;        
-    LOGV("AudioFlinger A2DP enabled status changed! %d", enabled);
+void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, void *param1, void *param2) {
+    LOGV("ioConfigChanged() event %d", event);
+    audio_io_handle_t ioHandle = (audio_io_handle_t)param1;
+    OutputDescriptor *desc;
+    uint32_t stream;
+
+    if (param1 == 0) return;
+
+    Mutex::Autolock _l(AudioSystem::gLock);
+
+    switch (event) {
+    case STREAM_CONFIG_CHANGED:
+        if (param2 == 0) break;
+        stream = *(uint32_t *)param2;
+        LOGV("ioConfigChanged() STREAM_CONFIG_CHANGED stream %d, output %p", stream, ioHandle);
+        if (gStreamOutputMap.indexOfKey(stream) >= 0) {
+            gStreamOutputMap.replaceValueFor(stream, ioHandle);
+        }
+        break;
+    case OUTPUT_OPENED: {
+        if (gOutputs.indexOfKey(ioHandle) >= 0) {
+            LOGV("ioConfigChanged() opening already existing output! %p", ioHandle);
+            break;
+        }
+        if (param2 == 0) break;
+        desc = (OutputDescriptor *)param2;
+
+        OutputDescriptor *outputDesc =  new OutputDescriptor(*desc);
+        gOutputs.add(ioHandle, outputDesc);
+        LOGV("ioConfigChanged() new output samplingRate %d, format %d channels %d frameCount %d latency %d",
+                outputDesc->samplingRate, outputDesc->format, outputDesc->channels, outputDesc->frameCount, outputDesc->latency);
+        } break;
+    case OUTPUT_CLOSED: {
+        if (gOutputs.indexOfKey(ioHandle) < 0) {
+            LOGW("ioConfigChanged() closing unknow output! %p", ioHandle);
+            break;
+        }
+        LOGV("ioConfigChanged() output %p closed", ioHandle);
+
+        gOutputs.removeItem(ioHandle);
+        for (int i = gStreamOutputMap.size() - 1; i >= 0 ; i--) {
+            if (gStreamOutputMap.valueAt(i) == ioHandle) {
+                gStreamOutputMap.removeItemsAt(i);
+            }
+        }
+        } break;
+
+    case OUTPUT_CONFIG_CHANGED: {
+        int index = gOutputs.indexOfKey(ioHandle);
+        if (index < 0) {
+            LOGW("ioConfigChanged() modifying unknow output! %p", ioHandle);
+            break;
+        }
+        if (param2 == 0) break;
+        desc = (OutputDescriptor *)param2;
+
+        LOGV("ioConfigChanged() new config for output %p samplingRate %d, format %d channels %d frameCount %d latency %d",
+                ioHandle, desc->samplingRate, desc->format,
+                desc->channels, desc->frameCount, desc->latency);
+        OutputDescriptor *outputDesc = gOutputs.valueAt(index);
+        delete outputDesc;
+        outputDesc =  new OutputDescriptor(*desc);
+        gOutputs.replaceValueFor(ioHandle, outputDesc);
+    } break;
+    case INPUT_OPENED:
+    case INPUT_CLOSED:
+    case INPUT_CONFIG_CHANGED:
+        break;
+
+    }
 }
 
 void AudioSystem::setErrorCallback(audio_error_callback cb) {
-    Mutex::Autolock _l(AudioSystem::gLock);
+    Mutex::Autolock _l(gLock);
     gAudioErrorCallback = cb;
 }
 
-int AudioSystem::getOutput(int streamType)
-{   
-    // make sure that gA2dpEnabled is valid by calling get_audio_flinger() which in turn 
-    // will call gAudioFlinger->isA2dpEnabled()
-    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-    if (af == 0) return NUM_AUDIO_OUTPUT_TYPES;
-
-    if (streamType == DEFAULT) {
-        streamType = MUSIC;
-    }
-    if (gA2dpEnabled && routedToA2dpOutput(streamType)) {
-        return AUDIO_OUTPUT_A2DP;
-    } else {
-        return AUDIO_OUTPUT_HARDWARE;
-    }
-}
-
 bool AudioSystem::routedToA2dpOutput(int streamType) {
     switch(streamType) {
     case MUSIC:
@@ -379,6 +438,451 @@
 }
 
 
+// client singleton for AudioPolicyService binder interface
+sp<IAudioPolicyService> AudioSystem::gAudioPolicyService;
+sp<AudioSystem::AudioPolicyServiceClient> AudioSystem::gAudioPolicyServiceClient;
+
+
+// establish binder interface to AudioFlinger service
+const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
+{
+    gLock.lock();
+    if (gAudioPolicyService.get() == 0) {
+        sp<IServiceManager> sm = defaultServiceManager();
+        sp<IBinder> binder;
+        do {
+            binder = sm->getService(String16("media.audio_policy"));
+            if (binder != 0)
+                break;
+            LOGW("AudioPolicyService not published, waiting...");
+            usleep(500000); // 0.5 s
+        } while(true);
+        if (gAudioPolicyServiceClient == NULL) {
+            gAudioPolicyServiceClient = new AudioPolicyServiceClient();
+        }
+        binder->linkToDeath(gAudioPolicyServiceClient);
+        gAudioPolicyService = interface_cast<IAudioPolicyService>(binder);
+        gLock.unlock();
+    } else {
+        gLock.unlock();
+    }
+    return gAudioPolicyService;
+}
+
+status_t AudioSystem::setDeviceConnectionState(audio_devices device,
+                                                  device_connection_state state,
+                                                  const char *device_address)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+
+    return aps->setDeviceConnectionState(device, state, device_address);
+}
+
+AudioSystem::device_connection_state AudioSystem::getDeviceConnectionState(audio_devices device,
+                                                  const char *device_address)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return DEVICE_STATE_UNAVAILABLE;
+
+    return aps->getDeviceConnectionState(device, device_address);
+}
+
+status_t AudioSystem::setPhoneState(int state)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+
+    return aps->setPhoneState(state);
+}
+
+status_t AudioSystem::setRingerMode(uint32_t mode, uint32_t mask)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->setRingerMode(mode, mask);
+}
+
+status_t AudioSystem::setForceUse(force_use usage, forced_config config)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->setForceUse(usage, config);
+}
+
+AudioSystem::forced_config AudioSystem::getForceUse(force_use usage)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return FORCE_NONE;
+    return aps->getForceUse(usage);
+}
+
+
+audio_io_handle_t AudioSystem::getOutput(stream_type stream,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    output_flags flags)
+{
+    audio_io_handle_t output = NULL;
+    if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0) {
+        Mutex::Autolock _l(gLock);
+        output = AudioSystem::gStreamOutputMap.valueFor(stream);
+        LOGV_IF((output != NULL), "getOutput() read %p from cache for stream %d", output, stream);
+    }
+    if (output == NULL) {
+        const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+        if (aps == 0) return NULL;
+        output = aps->getOutput(stream, samplingRate, format, channels, flags);
+        if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0) {
+            Mutex::Autolock _l(gLock);
+            AudioSystem::gStreamOutputMap.add(stream, output);
+        }
+    }
+    return output;
+}
+
+status_t AudioSystem::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->startOutput(output, stream);
+}
+
+status_t AudioSystem::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->stopOutput(output, stream);
+}
+
+void AudioSystem::releaseOutput(audio_io_handle_t output)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return;
+    aps->releaseOutput(output);
+}
+
+audio_io_handle_t AudioSystem::getInput(int inputSource,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    audio_in_acoustics acoustics)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return NULL;
+    return aps->getInput(inputSource, samplingRate, format, channels, acoustics);
+}
+
+status_t AudioSystem::startInput(audio_io_handle_t input)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->startInput(input);
+}
+
+status_t AudioSystem::stopInput(audio_io_handle_t input)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->stopInput(input);
+}
+
+void AudioSystem::releaseInput(audio_io_handle_t input)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return;
+    aps->releaseInput(input);
+}
+
+status_t AudioSystem::initStreamVolume(stream_type stream,
+                                    int indexMin,
+                                    int indexMax)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->initStreamVolume(stream, indexMin, indexMax);
+}
+
+status_t AudioSystem::setStreamVolumeIndex(stream_type stream, int index)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->setStreamVolumeIndex(stream, index);
+}
+
+status_t AudioSystem::getStreamVolumeIndex(stream_type stream, int *index)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->getStreamVolumeIndex(stream, index);
+}
+
+// ---------------------------------------------------------------------------
+
+void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who) {
+    Mutex::Autolock _l(AudioSystem::gLock);
+    AudioSystem::gAudioPolicyService.clear();
+
+    LOGW("AudioPolicyService server died!");
+}
+
+// ---------------------------------------------------------------------------
+
+
+// use emulated popcount optimization
+// http://www.df.lth.se/~john_e/gems/gem002d.html
+uint32_t AudioSystem::popCount(uint32_t u)
+{
+    u = ((u&0x55555555) + ((u>>1)&0x55555555));
+    u = ((u&0x33333333) + ((u>>2)&0x33333333));
+    u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
+    u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
+    u = ( u&0x0000ffff) + (u>>16);
+    return u;
+}
+
+bool AudioSystem::isOutputDevice(audio_devices device)
+{
+    if ((popCount(device) == 1 ) &&
+        ((device & ~AudioSystem::DEVICE_OUT_ALL) == 0)) {
+        return true;
+    } else {
+        return false;
+    }
+}
+
+bool AudioSystem::isInputDevice(audio_devices device)
+{
+    if ((popCount(device) == 1 ) &&
+        ((device & ~AudioSystem::DEVICE_IN_ALL) == 0)) {
+        return true;
+    } else {
+        return false;
+    }
+}
+
+bool AudioSystem::isA2dpDevice(audio_devices device)
+{
+    if ((popCount(device) == 1 ) &&
+        (device & (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
+                   AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+                   AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER))) {
+        return true;
+    } else {
+        return false;
+    }
+}
+
+bool AudioSystem::isBluetoothScoDevice(audio_devices device)
+{
+    if ((popCount(device) == 1 ) &&
+        (device & (AudioSystem::DEVICE_OUT_BLUETOOTH_SCO |
+                   AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
+                   AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT))) {
+        return true;
+    } else {
+        return false;
+    }
+}
+
+bool AudioSystem::isLowVisibility(stream_type stream)
+{
+    if (stream == AudioSystem::SYSTEM || stream == AudioSystem::NOTIFICATION) {
+        return true;
+    } else {
+        return false;
+    }
+}
+
+bool AudioSystem::isInputChannel(uint32_t channel)
+{
+    if ((channel & ~AudioSystem::CHANNEL_IN_ALL) == 0) {
+        return true;
+    } else {
+        return false;
+    }
+}
+
+bool AudioSystem::isOutputChannel(uint32_t channel)
+{
+    if ((channel & ~AudioSystem::CHANNEL_OUT_ALL) == 0) {
+        return true;
+    } else {
+        return false;
+    }
+}
+
+bool AudioSystem::isValidFormat(uint32_t format)
+{
+    switch (format & MAIN_FORMAT_MASK) {
+    case         PCM:
+    case         MP3:
+    case         AMR_NB:
+    case         AMR_WB:
+    case         AAC:
+    case         HE_AAC_V1:
+    case         HE_AAC_V2:
+    case         VORBIS:
+        return true;
+    default:
+        return false;
+    }
+}
+
+bool AudioSystem::isLinearPCM(uint32_t format)
+{
+    switch (format) {
+    case         PCM_16_BIT:
+    case         PCM_8_BIT:
+        return true;
+    default:
+        return false;
+    }
+}
+
+//------------------------- AudioParameter class implementation ---------------
+
+const char *AudioParameter::keyRouting = "routing";
+const char *AudioParameter::keySamplingRate = "sampling_rate";
+const char *AudioParameter::keyFormat = "format";
+const char *AudioParameter::keyChannels = "channels";
+const char *AudioParameter::keyFrameCount = "frame_count";
+
+AudioParameter::AudioParameter(const String8& keyValuePairs)
+{
+    char *str = new char[keyValuePairs.length()+1];
+    mKeyValuePairs = keyValuePairs;
+
+    strcpy(str, keyValuePairs.string());
+    char *pair = strtok(str, ";");
+    while (pair != NULL) {
+        if (strlen(pair) != 0) {
+            size_t eqIdx = strcspn(pair, "=");
+            String8 key = String8(pair, eqIdx);
+            String8 value;
+            if (eqIdx == strlen(pair)) {
+                value = String8("");
+            } else {
+                value = String8(pair + eqIdx + 1);
+            }
+            if (mParameters.indexOfKey(key) < 0) {
+                mParameters.add(key, value);
+            } else {
+                mParameters.replaceValueFor(key, value);
+            }
+        } else {
+            LOGV("AudioParameter() cstor empty key value pair");
+        }
+        pair = strtok(NULL, ";");
+    }
+
+    delete[] str;
+}
+
+AudioParameter::~AudioParameter()
+{
+    mParameters.clear();
+}
+
+String8 AudioParameter::toString()
+{
+    String8 str = String8("");
+
+    size_t size = mParameters.size();
+    for (size_t i = 0; i < size; i++) {
+        str += mParameters.keyAt(i);
+        str += "=";
+        str += mParameters.valueAt(i);
+        if (i < (size - 1)) str += ";";
+    }
+    return str;
+}
+
+status_t AudioParameter::add(const String8& key, const String8& value)
+{
+    if (mParameters.indexOfKey(key) < 0) {
+        mParameters.add(key, value);
+        return NO_ERROR;
+    } else {
+        mParameters.replaceValueFor(key, value);
+        return ALREADY_EXISTS;
+    }
+}
+
+status_t AudioParameter::addInt(const String8& key, const int value)
+{
+    char str[12];
+    if (snprintf(str, 12, "%d", value) > 0) {
+        String8 str8 = String8(str);
+        return add(key, str8);
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t AudioParameter::addFloat(const String8& key, const float value)
+{
+    char str[23];
+    if (snprintf(str, 23, "%.10f", value) > 0) {
+        String8 str8 = String8(str);
+        return add(key, str8);
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t AudioParameter::remove(const String8& key)
+{
+    if (mParameters.indexOfKey(key) >= 0) {
+        mParameters.removeItem(key);
+        return NO_ERROR;
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t AudioParameter::get(const String8& key, String8& value)
+{
+    if (mParameters.indexOfKey(key) >= 0) {
+        value = mParameters.valueFor(key);
+        return NO_ERROR;
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+status_t AudioParameter::getInt(const String8& key, int& value)
+{
+    String8 str8;
+    status_t result = get(key, str8);
+    value = 0;
+    if (result == NO_ERROR) {
+        int val;
+        if (sscanf(str8.string(), "%d", &val) == 1) {
+            value = val;
+        } else {
+            result = INVALID_OPERATION;
+        }
+    }
+    return result;
+}
+
+status_t AudioParameter::getFloat(const String8& key, float& value)
+{
+    String8 str8;
+    status_t result = get(key, str8);
+    value = 0;
+    if (result == NO_ERROR) {
+        float val;
+        if (sscanf(str8.string(), "%f", &val) == 1) {
+            value = val;
+        } else {
+            result = INVALID_OPERATION;
+        }
+    }
+    return result;
+}
 
 }; // namespace android
 
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7b9eda7..b147d25 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -54,7 +54,7 @@
         int streamType,
         uint32_t sampleRate,
         int format,
-        int channelCount,
+        int channels,
         int frameCount,
         uint32_t flags,
         callback_t cbf,
@@ -62,7 +62,7 @@
         int notificationFrames)
     : mStatus(NO_INIT)
 {
-    mStatus = set(streamType, sampleRate, format, channelCount,
+    mStatus = set(streamType, sampleRate, format, channels,
             frameCount, flags, cbf, user, notificationFrames, 0);
 }
 
@@ -70,7 +70,7 @@
         int streamType,
         uint32_t sampleRate,
         int format,
-        int channelCount,
+        int channels,
         const sp<IMemory>& sharedBuffer,
         uint32_t flags,
         callback_t cbf,
@@ -78,7 +78,7 @@
         int notificationFrames)
     : mStatus(NO_INIT)
 {
-    mStatus = set(streamType, sampleRate, format, channelCount,
+    mStatus = set(streamType, sampleRate, format, channels,
             0, flags, cbf, user, notificationFrames, sharedBuffer);
 }
 
@@ -97,6 +97,7 @@
         }
         mAudioTrack.clear();
         IPCThreadState::self()->flushCommands();
+        AudioSystem::releaseOutput(getOutput());
     }
 }
 
@@ -104,7 +105,7 @@
         int streamType,
         uint32_t sampleRate,
         int format,
-        int channelCount,
+        int channels,
         int frameCount,
         uint32_t flags,
         callback_t cbf,
@@ -150,63 +151,84 @@
     if (format == 0) {
         format = AudioSystem::PCM_16_BIT;
     }
-    if (channelCount == 0) {
-        channelCount = 2;
+    if (channels == 0) {
+        channels = AudioSystem::CHANNEL_OUT_STEREO;
     }
 
     // validate parameters
-    if (((format != AudioSystem::PCM_8_BIT) || sharedBuffer != 0) &&
-        (format != AudioSystem::PCM_16_BIT)) {
+    if (!AudioSystem::isValidFormat(format)) {
         LOGE("Invalid format");
         return BAD_VALUE;
     }
-    if (channelCount != 1 && channelCount != 2) {
-        LOGE("Invalid channel number");
+
+    // force direct flag if format is not linear PCM
+    if (!AudioSystem::isLinearPCM(format)) {
+        flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
+    }
+
+    if (!AudioSystem::isOutputChannel(channels)) {
+        LOGE("Invalid channel mask");
+        return BAD_VALUE;
+    }
+    uint32_t channelCount = AudioSystem::popCount(channels);
+
+    audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
+            sampleRate, format, channels, (AudioSystem::output_flags)flags);
+
+    if (output == 0) {
+        LOGE("Could not get audio output for stream type %d", streamType);
         return BAD_VALUE;
     }
 
-    // Ensure that buffer depth covers at least audio hardware latency
-    uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
-    if (minBufCount < 2) minBufCount = 2;
-
-    // When playing from shared buffer, playback will start even if last audioflinger
-    // block is partly filled.
-    if (sharedBuffer != 0 && minBufCount > 1) {
-        minBufCount--;
-    }
-
-    int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
-
-    if (sharedBuffer == 0) {
-        if (frameCount == 0) {
-            frameCount = minFrameCount;
-        }
-        if (notificationFrames == 0) {
-            notificationFrames = frameCount/2;
-        }
-        // Make sure that application is notified with sufficient margin
-        // before underrun
-        if (notificationFrames > frameCount/2) {
-            notificationFrames = frameCount/2;
+    if (!AudioSystem::isLinearPCM(format)) {
+        if (sharedBuffer != 0) {
+            frameCount = sharedBuffer->size();
         }
     } else {
-        // Ensure that buffer alignment matches channelcount
-        if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
-            LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
-            return BAD_VALUE;
-        }
-        frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
-    }
+        // Ensure that buffer depth covers at least audio hardware latency
+        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
+        if (minBufCount < 2) minBufCount = 2;
 
-    if (frameCount < minFrameCount) {
-      LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
-      return BAD_VALUE;
+        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+
+        if (sharedBuffer == 0) {
+            if (frameCount == 0) {
+                frameCount = minFrameCount;
+            }
+            if (notificationFrames == 0) {
+                notificationFrames = frameCount/2;
+            }
+            // Make sure that application is notified with sufficient margin
+            // before underrun
+            if (notificationFrames > frameCount/2) {
+                notificationFrames = frameCount/2;
+            }
+            if (frameCount < minFrameCount) {
+              LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
+              return BAD_VALUE;
+            }
+        } else {
+            // Ensure that buffer alignment matches channelcount
+            if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
+                LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
+                return BAD_VALUE;
+            }
+            frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
+        }
     }
 
     // create the track
     status_t status;
     sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
-                streamType, sampleRate, format, channelCount, frameCount, flags, sharedBuffer, &status);
+                                                      streamType,
+                                                      sampleRate,
+                                                      format,
+                                                      channelCount,
+                                                      frameCount,
+                                                      ((uint16_t)flags) << 16,
+                                                      sharedBuffer,
+                                                      output,
+                                                      &status);
 
     if (track == 0) {
         LOGE("AudioFlinger could not create track, status: %d", status);
@@ -245,6 +267,7 @@
     mVolume[RIGHT] = 1.0f;
     mStreamType = streamType;
     mFormat = format;
+    mChannels = channels;
     mChannelCount = channelCount;
     mSharedBuffer = sharedBuffer;
     mMuted = false;
@@ -259,6 +282,7 @@
     mMarkerReached = false;
     mNewPosition = 0;
     mUpdatePeriod = 0;
+    mFlags = flags;
 
     return NO_ERROR;
 }
@@ -297,7 +321,11 @@
 
 int AudioTrack::frameSize() const
 {
-    return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+    if (AudioSystem::isLinearPCM(mFormat)) {
+        return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+    } else {
+        return sizeof(uint8_t);
+    }
 }
 
 sp<IMemory>& AudioTrack::sharedBuffer()
@@ -323,6 +351,7 @@
      }
 
     if (android_atomic_or(1, &mActive) == 0) {
+        AudioSystem::startOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
         mNewPosition = mCblk->server + mUpdatePeriod;
         mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
         mCblk->waitTimeMs = 0;
@@ -367,6 +396,7 @@
         } else {
             setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
         }
+        AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
     }
 
     if (t != 0) {
@@ -382,12 +412,12 @@
 void AudioTrack::flush()
 {
     LOGV("flush");
-    
+
     // clear playback marker and periodic update counter
     mMarkerPosition = 0;
     mMarkerReached = false;
     mUpdatePeriod = 0;
-    
+
 
     if (!mActive) {
         mAudioTrack->flush();
@@ -403,6 +433,7 @@
     if (android_atomic_and(~1, &mActive) == 1) {
         mActive = 0;
         mAudioTrack->pause();
+        AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
     }
 }
 
@@ -455,7 +486,6 @@
 {
     audio_track_cblk_t* cblk = mCblk;
 
-
     Mutex::Autolock _l(cblk->lock);
 
     if (loopCount == 0) {
@@ -476,7 +506,7 @@
         LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
             loopStart, loopEnd, mFrameCount);
         return BAD_VALUE;
-    }   
+    }
 
     cblk->loopStart = loopStart;
     cblk->loopEnd = loopEnd;
@@ -555,7 +585,7 @@
 
     mCblk->server = position;
     mCblk->forceReady = 1;
-    
+
     return NO_ERROR;
 }
 
@@ -571,7 +601,7 @@
 status_t AudioTrack::reload()
 {
     if (!stopped()) return INVALID_OPERATION;
-    
+
     flush();
 
     mCblk->stepUser(mFrameCount);
@@ -579,6 +609,12 @@
     return NO_ERROR;
 }
 
+audio_io_handle_t AudioTrack::getOutput()
+{
+    return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType,
+            mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
+}
+
 // -------------------------------------------------------------------------
 
 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -608,7 +644,7 @@
                 return WOULD_BLOCK;
             timeout = 0;
             result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
-            if (__builtin_expect(result!=NO_ERROR, false)) { 
+            if (__builtin_expect(result!=NO_ERROR, false)) {
                 cblk->waitTimeMs += waitTimeMs;
                 if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
                     // timing out when a loop has been set and we have already written upto loop end
@@ -616,7 +652,7 @@
                     if (cblk->user < cblk->loopEnd) {
                         LOGW(   "obtainBuffer timed out (is the CPU pegged?) %p "
                                 "user=%08x, server=%08x", this, cblk->user, cblk->server);
-                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) 
+                        //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
                         cblk->lock.unlock();
                         mAudioTrack->start();
                         cblk->lock.lock();
@@ -624,7 +660,7 @@
                     }
                     cblk->waitTimeMs = 0;
                 }
-                
+
                 if (--waitCount == 0) {
                     return TIMED_OUT;
                 }
@@ -636,7 +672,7 @@
     }
 
     cblk->waitTimeMs = 0;
-    
+
     if (framesReq > framesAvail) {
         framesReq = framesAvail;
     }
@@ -653,12 +689,16 @@
         "but didn't need to be locked. We recovered, but "
         "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
 
-    audioBuffer->flags       = mMuted ? Buffer::MUTE : 0;
-    audioBuffer->channelCount= mChannelCount;
-    audioBuffer->format      = AudioSystem::PCM_16_BIT;
-    audioBuffer->frameCount  = framesReq;
-    audioBuffer->size = framesReq*mChannelCount*sizeof(int16_t);
-    audioBuffer->raw         = (int8_t *)cblk->buffer(u);
+    audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
+    audioBuffer->channelCount = mChannelCount;
+    audioBuffer->frameCount = framesReq;
+    audioBuffer->size = framesReq * cblk->frameSize;
+    if (AudioSystem::isLinearPCM(mFormat)) {
+        audioBuffer->format = AudioSystem::PCM_16_BIT;
+    } else {
+        audioBuffer->format = mFormat;
+    }
+    audioBuffer->raw = (int8_t *)cblk->buffer(u);
     active = mActive;
     return active ? status_t(NO_ERROR) : status_t(STOPPED);
 }
@@ -690,10 +730,8 @@
     Buffer audioBuffer;
 
     do {
-        audioBuffer.frameCount = userSize/mChannelCount;
-        if (mFormat == AudioSystem::PCM_16_BIT) {
-            audioBuffer.frameCount >>= 1;
-        }
+        audioBuffer.frameCount = userSize/frameSize();
+
         // Calling obtainBuffer() with a negative wait count causes
         // an (almost) infinite wait time.
         status_t err = obtainBuffer(&audioBuffer, -1);
@@ -705,6 +743,7 @@
         }
 
         size_t toWrite;
+
         if (mFormat == AudioSystem::PCM_8_BIT) {
             // Divide capacity by 2 to take expansion into account
             toWrite = audioBuffer.size>>1;
@@ -742,13 +781,13 @@
         if (mCblk->flowControlFlag == 0) {
             mCbf(EVENT_UNDERRUN, mUserData, 0);
             if (mCblk->server == mCblk->frameCount) {
-                mCbf(EVENT_BUFFER_END, mUserData, 0);                
+                mCbf(EVENT_BUFFER_END, mUserData, 0);
             }
             mCblk->flowControlFlag = 1;
             if (mSharedBuffer != 0) return false;
         }
     }
-    
+
     // Manage loop end callback
     while (mLoopCount > mCblk->loopCount) {
         int loopCount = -1;
@@ -767,7 +806,7 @@
     }
 
     // Manage new position callback
-    if(mUpdatePeriod > 0) {
+    if (mUpdatePeriod > 0) {
         while (mCblk->server >= mNewPosition) {
             mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
             mNewPosition += mUpdatePeriod;
@@ -784,10 +823,10 @@
     do {
 
         audioBuffer.frameCount = frames;
-        
-        // Calling obtainBuffer() with a wait count of 1 
-        // limits wait time to WAIT_PERIOD_MS. This prevents from being 
-        // stuck here not being able to handle timed events (position, markers, loops). 
+
+        // Calling obtainBuffer() with a wait count of 1
+        // limits wait time to WAIT_PERIOD_MS. This prevents from being
+        // stuck here not being able to handle timed events (position, markers, loops).
         status_t err = obtainBuffer(&audioBuffer, 1);
         if (err < NO_ERROR) {
             if (err != TIMED_OUT) {
@@ -832,7 +871,11 @@
         }
 
         audioBuffer.size = writtenSize;
-        audioBuffer.frameCount = writtenSize/mChannelCount/sizeof(int16_t);
+        // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
+        // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sampel size of
+        // 16 bit.
+        audioBuffer.frameCount = writtenSize/mCblk->frameSize;
+
         frames -= audioBuffer.frameCount;
 
         releaseBuffer(&audioBuffer);
@@ -949,7 +992,7 @@
         // we switch to normal obtainBuffer() timeout period
         if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
             bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1;
-        }        
+        }
         // It is possible that we receive a flush()
         // while the mixer is processing a block: in this case,
         // stepServer() is called After the flush() has reset u & s and
@@ -981,7 +1024,7 @@
 
 void* audio_track_cblk_t::buffer(uint32_t offset) const
 {
-    return (int16_t *)this->buffers + (offset-userBase)*this->channels;
+    return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
 }
 
 uint32_t audio_track_cblk_t::framesAvailable()
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 6fc0cb7..9385367 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -16,6 +16,7 @@
 */
 
 #define LOG_TAG "IAudioFlinger"
+//#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
 #include <stdint.h>
@@ -44,17 +45,21 @@
     STREAM_VOLUME,
     STREAM_MUTE,
     SET_MODE,
-    GET_MODE,
-    SET_ROUTING,
-    GET_ROUTING,
     SET_MIC_MUTE,
     GET_MIC_MUTE,
     IS_MUSIC_ACTIVE,
-    SET_PARAMETER,
+    SET_PARAMETERS,
+    GET_PARAMETERS,
     REGISTER_CLIENT,
     GET_INPUTBUFFERSIZE,
-    WAKE_UP,
-    IS_A2DP_ENABLED
+    OPEN_OUTPUT,
+    OPEN_DUPLICATE_OUTPUT,
+    CLOSE_OUTPUT,
+    SUSPEND_OUTPUT,
+    RESTORE_OUTPUT,
+    OPEN_INPUT,
+    CLOSE_INPUT,
+    SET_STREAM_OUTPUT
 };
 
 class BpAudioFlinger : public BpInterface<IAudioFlinger>
@@ -74,6 +79,7 @@
                                 int frameCount,
                                 uint32_t flags,
                                 const sp<IMemory>& sharedBuffer,
+                                void *output,
                                 status_t *status)
     {
         Parcel data, reply;
@@ -86,6 +92,7 @@
         data.writeInt32(frameCount);
         data.writeInt32(flags);
         data.writeStrongBinder(sharedBuffer->asBinder());
+        data.write(&output, sizeof(void *));
         status_t lStatus = remote()->transact(CREATE_TRACK, data, &reply);
         if (lStatus != NO_ERROR) {
             LOGE("createTrack error: %s", strerror(-lStatus));
@@ -99,7 +106,7 @@
 
     virtual sp<IAudioRecord> openRecord(
                                 pid_t pid,
-                                int inputSource,
+                                void *input,
                                 uint32_t sampleRate,
                                 int format,
                                 int channelCount,
@@ -110,7 +117,7 @@
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32(pid);
-        data.writeInt32(inputSource);
+        data.write(&input, sizeof(void *));
         data.writeInt32(sampleRate);
         data.writeInt32(format);
         data.writeInt32(channelCount);
@@ -124,47 +131,47 @@
         return interface_cast<IAudioRecord>(reply.readStrongBinder());
     }
 
-    virtual uint32_t sampleRate(int output) const
+    virtual uint32_t sampleRate(void *output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32(output);
+        data.write(&output, sizeof(void *));
         remote()->transact(SAMPLE_RATE, data, &reply);
         return reply.readInt32();
     }
 
-    virtual int channelCount(int output) const
+    virtual int channelCount(void *output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32(output);
+        data.write(&output, sizeof(void *));
         remote()->transact(CHANNEL_COUNT, data, &reply);
         return reply.readInt32();
     }
 
-    virtual int format(int output) const
+    virtual int format(void *output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32(output);
+        data.write(&output, sizeof(void *));
         remote()->transact(FORMAT, data, &reply);
         return reply.readInt32();
     }
 
-    virtual size_t frameCount(int output) const
+    virtual size_t frameCount(void *output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32(output);
+        data.write(&output, sizeof(void *));
         remote()->transact(FRAME_COUNT, data, &reply);
         return reply.readInt32();
     }
 
-    virtual uint32_t latency(int output) const
+    virtual uint32_t latency(void *output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32(output);
+        data.write(&output, sizeof(void *));
         remote()->transact(LATENCY, data, &reply);
         return reply.readInt32();
     }
@@ -203,12 +210,13 @@
         return reply.readInt32();
     }
 
-    virtual status_t setStreamVolume(int stream, float value)
+    virtual status_t setStreamVolume(int stream, float value, void *output)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32(stream);
         data.writeFloat(value);
+        data.write(&output, sizeof(void *));
         remote()->transact(SET_STREAM_VOLUME, data, &reply);
         return reply.readInt32();
     }
@@ -223,11 +231,12 @@
         return reply.readInt32();
     }
 
-    virtual float streamVolume(int stream) const
+    virtual float streamVolume(int stream, void *output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32(stream);
+        data.write(&output, sizeof(void *));
         remote()->transact(STREAM_VOLUME, data, &reply);
         return reply.readFloat();
     }
@@ -241,26 +250,6 @@
         return reply.readInt32();
     }
 
-    virtual status_t setRouting(int mode, uint32_t routes, uint32_t mask)
-    {
-        Parcel data, reply;
-        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32(mode);
-        data.writeInt32(routes);
-        data.writeInt32(mask);
-        remote()->transact(SET_ROUTING, data, &reply);
-        return reply.readInt32();
-    }
-
-    virtual uint32_t getRouting(int mode) const
-    {
-        Parcel data, reply;
-        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32(mode);
-        remote()->transact(GET_ROUTING, data, &reply);
-        return reply.readInt32();
-    }
-
     virtual status_t setMode(int mode)
     {
         Parcel data, reply;
@@ -270,14 +259,6 @@
         return reply.readInt32();
     }
 
-    virtual int getMode() const
-    {
-        Parcel data, reply;
-        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        remote()->transact(GET_MODE, data, &reply);
-        return reply.readInt32();
-    }
-
     virtual status_t setMicMute(bool state)
     {
         Parcel data, reply;
@@ -303,16 +284,26 @@
         return reply.readInt32();
     }
 
-    virtual status_t setParameter(const char* key, const char* value)
+    virtual status_t setParameters(void *ioHandle, const String8& keyValuePairs)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeCString(key);
-        data.writeCString(value);
-        remote()->transact(SET_PARAMETER, data, &reply);
+        data.write(&ioHandle, sizeof(void *));
+        data.writeString8(keyValuePairs);
+        remote()->transact(SET_PARAMETERS, data, &reply);
         return reply.readInt32();
     }
-    
+
+    virtual String8 getParameters(void *ioHandle, const String8& keys)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.write(&ioHandle, sizeof(void *));
+        data.writeString8(keys);
+        remote()->transact(GET_PARAMETERS, data, &reply);
+        return reply.readString8();
+    }
+
     virtual void registerClient(const sp<IAudioFlingerClient>& client)
     {
         Parcel data, reply;
@@ -320,7 +311,7 @@
         data.writeStrongBinder(client->asBinder());
         remote()->transact(REGISTER_CLIENT, data, &reply);
     }
-    
+
     virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
     {
         Parcel data, reply;
@@ -331,21 +322,133 @@
         remote()->transact(GET_INPUTBUFFERSIZE, data, &reply);
         return reply.readInt32();
     }
-    
-    virtual void wakeUp()
+
+    virtual void *openOutput(uint32_t *pDevices,
+                            uint32_t *pSamplingRate,
+                            uint32_t *pFormat,
+                            uint32_t *pChannels,
+                            uint32_t *pLatencyMs,
+                            uint32_t flags)
     {
         Parcel data, reply;
+        uint32_t devices = pDevices ? *pDevices : 0;
+        uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+        uint32_t format = pFormat ? *pFormat : 0;
+        uint32_t channels = pChannels ? *pChannels : 0;
+        uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
+
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        remote()->transact(WAKE_UP, data, &reply, IBinder::FLAG_ONEWAY);
-        return;
+        data.writeInt32(devices);
+        data.writeInt32(samplingRate);
+        data.writeInt32(format);
+        data.writeInt32(channels);
+        data.writeInt32(latency);
+        data.writeInt32(flags);
+        remote()->transact(OPEN_OUTPUT, data, &reply);
+        void *output;
+        reply.read(&output, sizeof(void *));
+        LOGV("openOutput() returned output, %p", output);
+        devices = reply.readInt32();
+        if (pDevices) *pDevices = devices;
+        samplingRate = reply.readInt32();
+        if (pSamplingRate) *pSamplingRate = samplingRate;
+        format = reply.readInt32();
+        if (pFormat) *pFormat = format;
+        channels = reply.readInt32();
+        if (pChannels) *pChannels = channels;
+        latency = reply.readInt32();
+        if (pLatencyMs) *pLatencyMs = latency;
+        return output;
     }
 
-    virtual bool isA2dpEnabled() const
+    virtual void *openDuplicateOutput(void *output1, void *output2)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        remote()->transact(IS_A2DP_ENABLED, data, &reply);
-        return (bool)reply.readInt32();
+        data.write(&output1, sizeof(void *));
+        data.write(&output2, sizeof(void *));
+        remote()->transact(OPEN_DUPLICATE_OUTPUT, data, &reply);
+        void *output;
+        reply.read(&output, sizeof(void *));
+        return output;
+    }
+
+    virtual status_t closeOutput(void *output)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.write(&output, sizeof(void *));
+        remote()->transact(CLOSE_OUTPUT, data, &reply);
+        return reply.readInt32();
+    }
+
+    virtual status_t suspendOutput(void *output)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.write(&output, sizeof(void *));
+        remote()->transact(SUSPEND_OUTPUT, data, &reply);
+        return reply.readInt32();
+    }
+
+    virtual status_t restoreOutput(void *output)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.write(&output, sizeof(void *));
+        remote()->transact(RESTORE_OUTPUT, data, &reply);
+        return reply.readInt32();
+    }
+
+    virtual void *openInput(uint32_t *pDevices,
+                                        uint32_t *pSamplingRate,
+                                        uint32_t *pFormat,
+                                        uint32_t *pChannels,
+                                        uint32_t acoustics)
+    {
+        Parcel data, reply;
+        uint32_t devices = pDevices ? *pDevices : 0;
+        uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+        uint32_t format = pFormat ? *pFormat : 0;
+        uint32_t channels = pChannels ? *pChannels : 0;
+
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(devices);
+        data.writeInt32(samplingRate);
+        data.writeInt32(format);
+        data.writeInt32(channels);
+        data.writeInt32(acoustics);
+        remote()->transact(OPEN_INPUT, data, &reply);
+        void *input;
+        reply.read(&input, sizeof(void *));
+        devices = reply.readInt32();
+        if (pDevices) *pDevices = devices;
+        samplingRate = reply.readInt32();
+        if (pSamplingRate) *pSamplingRate = samplingRate;
+        format = reply.readInt32();
+        if (pFormat) *pFormat = format;
+        channels = reply.readInt32();
+        if (pChannels) *pChannels = channels;
+        return input;
+    }
+
+    virtual status_t closeInput(void *input)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.write(&input, sizeof(void *));
+        remote()->transact(CLOSE_INPUT, data, &reply);
+        return reply.readInt32();
+    }
+
+    virtual status_t setStreamOutput(uint32_t stream, void *output)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(stream);
+        data.write(&output, sizeof(void *));
+        remote()->transact(SET_STREAM_OUTPUT, data, &reply);
+        return reply.readInt32();
     }
 };
 
@@ -367,10 +470,12 @@
             size_t bufferCount = data.readInt32();
             uint32_t flags = data.readInt32();
             sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
+            void *output;
+            data.read(&output, sizeof(void *));
             status_t status;
             sp<IAudioTrack> track = createTrack(pid,
                     streamType, sampleRate, format,
-                    channelCount, bufferCount, flags, buffer, &status);
+                    channelCount, bufferCount, flags, buffer, output, &status);
             reply->writeInt32(status);
             reply->writeStrongBinder(track->asBinder());
             return NO_ERROR;
@@ -378,14 +483,15 @@
         case OPEN_RECORD: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             pid_t pid = data.readInt32();
-            int inputSource = data.readInt32();
+            void *input;
+            data.read(&input, sizeof(void *));
             uint32_t sampleRate = data.readInt32();
             int format = data.readInt32();
             int channelCount = data.readInt32();
             size_t bufferCount = data.readInt32();
             uint32_t flags = data.readInt32();
             status_t status;
-            sp<IAudioRecord> record = openRecord(pid, inputSource,
+            sp<IAudioRecord> record = openRecord(pid, input,
                     sampleRate, format, channelCount, bufferCount, flags, &status);
             reply->writeInt32(status);
             reply->writeStrongBinder(record->asBinder());
@@ -393,31 +499,36 @@
         } break;
         case SAMPLE_RATE: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int output = data.readInt32();
+            void *output;
+            data.read(&output, sizeof(void *));
             reply->writeInt32( sampleRate(output) );
             return NO_ERROR;
         } break;
         case CHANNEL_COUNT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int output = data.readInt32();
+            void *output;
+            data.read(&output, sizeof(void *));
             reply->writeInt32( channelCount(output) );
             return NO_ERROR;
         } break;
         case FORMAT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int output = data.readInt32();
+            void *output;
+            data.read(&output, sizeof(void *));
             reply->writeInt32( format(output) );
             return NO_ERROR;
         } break;
         case FRAME_COUNT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int output = data.readInt32();
+            void *output;
+            data.read(&output, sizeof(void *));
             reply->writeInt32( frameCount(output) );
             return NO_ERROR;
         } break;
         case LATENCY: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int output = data.readInt32();
+            void *output;
+            data.read(&output, sizeof(void *));
             reply->writeInt32( latency(output) );
             return NO_ERROR;
         } break;
@@ -444,7 +555,10 @@
         case SET_STREAM_VOLUME: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             int stream = data.readInt32();
-            reply->writeInt32( setStreamVolume(stream, data.readFloat()) );
+            float volume = data.readFloat();
+            void *output;
+            data.read(&output, sizeof(void *));
+            reply->writeInt32( setStreamVolume(stream, volume, output) );
             return NO_ERROR;
         } break;
         case SET_STREAM_MUTE: {
@@ -456,7 +570,9 @@
         case STREAM_VOLUME: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             int stream = data.readInt32();
-            reply->writeFloat( streamVolume(stream) );
+            void *output;
+            data.read(&output, sizeof(void *));
+            reply->writeFloat( streamVolume(stream, output) );
             return NO_ERROR;
         } break;
         case STREAM_MUTE: {
@@ -465,31 +581,12 @@
             reply->writeInt32( streamMute(stream) );
             return NO_ERROR;
         } break;
-        case SET_ROUTING: {
-            CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int mode = data.readInt32();
-            uint32_t routes = data.readInt32();
-            uint32_t mask = data.readInt32();
-            reply->writeInt32( setRouting(mode, routes, mask) );
-            return NO_ERROR;
-        } break;
-        case GET_ROUTING: {
-            CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int mode = data.readInt32();
-            reply->writeInt32( getRouting(mode) );
-            return NO_ERROR;
-        } break;
         case SET_MODE: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             int mode = data.readInt32();
             reply->writeInt32( setMode(mode) );
             return NO_ERROR;
         } break;
-        case GET_MODE: {
-            CHECK_INTERFACE(IAudioFlinger, data, reply);
-            reply->writeInt32( getMode() );
-            return NO_ERROR;
-        } break;
         case SET_MIC_MUTE: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             int state = data.readInt32();
@@ -506,13 +603,23 @@
             reply->writeInt32( isMusicActive() );
             return NO_ERROR;
         } break;
-        case SET_PARAMETER: {
+        case SET_PARAMETERS: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            const char *key = data.readCString();
-            const char *value = data.readCString();
-            reply->writeInt32( setParameter(key, value) );
+            void *ioHandle;
+            data.read(&ioHandle, sizeof(void *));
+            String8 keyValuePairs(data.readString8());
+            reply->writeInt32(setParameters(ioHandle, keyValuePairs));
             return NO_ERROR;
-        } break;
+         } break;
+        case GET_PARAMETERS: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            void *ioHandle;
+            data.read(&ioHandle, sizeof(void *));
+            String8 keys(data.readString8());
+            reply->writeString8(getParameters(ioHandle, keys));
+            return NO_ERROR;
+         } break;
+
         case REGISTER_CLIENT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient>(data.readStrongBinder());
@@ -527,14 +634,93 @@
             reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) );
             return NO_ERROR;
         } break;
-        case WAKE_UP: {
+        case OPEN_OUTPUT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            wakeUp();
+            uint32_t devices = data.readInt32();
+            uint32_t samplingRate = data.readInt32();
+            uint32_t format = data.readInt32();
+            uint32_t channels = data.readInt32();
+            uint32_t latency = data.readInt32();
+            uint32_t flags = data.readInt32();
+            void *output = openOutput(&devices,
+                                     &samplingRate,
+                                     &format,
+                                     &channels,
+                                     &latency,
+                                     flags);
+            LOGV("OPEN_OUTPUT output, %p", output);
+            reply->write(&output, sizeof(void *));
+            reply->writeInt32(devices);
+            reply->writeInt32(samplingRate);
+            reply->writeInt32(format);
+            reply->writeInt32(channels);
+            reply->writeInt32(latency);
             return NO_ERROR;
         } break;
-        case IS_A2DP_ENABLED: {
+        case OPEN_DUPLICATE_OUTPUT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            reply->writeInt32( (int)isA2dpEnabled() );
+            void *output1;
+            void *output2;
+            data.read(&output1, sizeof(void *));
+            data.read(&output2, sizeof(void *));
+            void *output = openDuplicateOutput(output1, output2);
+            reply->write(&output, sizeof(void *));
+            return NO_ERROR;
+        } break;
+        case CLOSE_OUTPUT: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            void *output;
+            data.read(&output, sizeof(void *));
+            reply->writeInt32(closeOutput(output));
+            return NO_ERROR;
+        } break;
+        case SUSPEND_OUTPUT: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            void *output;
+            data.read(&output, sizeof(void *));
+            reply->writeInt32(suspendOutput(output));
+            return NO_ERROR;
+        } break;
+        case RESTORE_OUTPUT: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            void *output;
+            data.read(&output, sizeof(void *));
+            reply->writeInt32(restoreOutput(output));
+            return NO_ERROR;
+        } break;
+        case OPEN_INPUT: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            uint32_t devices = data.readInt32();
+            uint32_t samplingRate = data.readInt32();
+            uint32_t format = data.readInt32();
+            uint32_t channels = data.readInt32();
+            uint32_t acoutics = data.readInt32();
+
+            void *input = openInput(&devices,
+                                     &samplingRate,
+                                     &format,
+                                     &channels,
+                                     acoutics);
+            reply->write(&input, sizeof(void *));
+            reply->writeInt32(devices);
+            reply->writeInt32(samplingRate);
+            reply->writeInt32(format);
+            reply->writeInt32(channels);
+            return NO_ERROR;
+        } break;
+        case CLOSE_INPUT: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            void *input;
+            data.read(&input, sizeof(void *));
+            reply->writeInt32(closeInput(input));
+            return NO_ERROR;
+        } break;
+        case SET_STREAM_OUTPUT: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            void *output;
+            uint32_t stream = data.readInt32();
+            data.read(&output, sizeof(void *));
+            reply->writeInt32(setStreamOutput(stream, output));
             return NO_ERROR;
         } break;
         default:
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index 75699b4..eaae977 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -23,11 +23,12 @@
 #include <binder/Parcel.h>
 
 #include <media/IAudioFlingerClient.h>
+#include <media/AudioSystem.h>
 
 namespace android {
 
 enum {
-    AUDIO_OUTPUT_CHANGED = IBinder::FIRST_CALL_TRANSACTION
+    IO_CONFIG_CHANGED = IBinder::FIRST_CALL_TRANSACTION
 };
 
 class BpAudioFlingerClient : public BpInterface<IAudioFlingerClient>
@@ -38,12 +39,25 @@
     {
     }
 
-    void a2dpEnabledChanged(bool enabled)
+    void ioConfigChanged(int event, void *param1, void *param2)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlingerClient::getInterfaceDescriptor());
-        data.writeInt32((int)enabled);
-        remote()->transact(AUDIO_OUTPUT_CHANGED, data, &reply, IBinder::FLAG_ONEWAY);
+        data.writeInt32(event);
+        data.write(&param1, sizeof(void *));
+        if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
+            uint32_t stream = *(uint32_t *)param2;
+            LOGV("ioConfigChanged stream %d", stream);
+            data.writeInt32(stream);
+        } else if (event != AudioSystem::OUTPUT_CLOSED && event != AudioSystem::INPUT_CLOSED) {
+            AudioSystem::OutputDescriptor *desc = (AudioSystem::OutputDescriptor *)param2;
+            data.writeInt32(desc->samplingRate);
+            data.writeInt32(desc->format);
+            data.writeInt32(desc->channels);
+            data.writeInt32(desc->frameCount);
+            data.writeInt32(desc->latency);
+        }
+        remote()->transact(IO_CONFIG_CHANGED, data, &reply, IBinder::FLAG_ONEWAY);
     }
 };
 
@@ -55,10 +69,27 @@
     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
     switch(code) {
-        case AUDIO_OUTPUT_CHANGED: {
+    case IO_CONFIG_CHANGED: {
             CHECK_INTERFACE(IAudioFlingerClient, data, reply);
-            bool enabled = (bool)data.readInt32();
-            a2dpEnabledChanged(enabled);
+            int event = data.readInt32();
+            void *param1;
+            void *param2 = 0;
+            AudioSystem::OutputDescriptor desc;
+            uint32_t stream;
+            data.read(&param1, sizeof(void *));
+            if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
+                stream = data.readInt32();
+                param2 = &stream;
+                LOGV("STREAM_CONFIG_CHANGED stream %d", stream);
+            } else if (event != AudioSystem::OUTPUT_CLOSED && event != AudioSystem::INPUT_CLOSED) {
+                desc.samplingRate = data.readInt32();
+                desc.format = data.readInt32();
+                desc.channels = data.readInt32();
+                desc.frameCount = data.readInt32();
+                desc.latency = data.readInt32();
+                param2 = &desc;
+            }
+            ioConfigChanged(event, param1, param2);
             return NO_ERROR;
         } break;
         default:
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
new file mode 100644
index 0000000..0d8a329
--- /dev/null
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -0,0 +1,423 @@
+/*
+**
+** Copyright 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 "IAudioPolicyService"
+#include <utils/Log.h>
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <binder/Parcel.h>
+
+#include <media/IAudioPolicyService.h>
+
+namespace android {
+
+enum {
+    SET_DEVICE_CONNECTION_STATE = IBinder::FIRST_CALL_TRANSACTION,
+    GET_DEVICE_CONNECTION_STATE,
+    SET_PHONE_STATE,
+    SET_RINGER_MODE,
+    SET_FORCE_USE,
+    GET_FORCE_USE,
+    GET_OUTPUT,
+    START_OUTPUT,
+    STOP_OUTPUT,
+    RELEASE_OUTPUT,
+    GET_INPUT,
+    START_INPUT,
+    STOP_INPUT,
+    RELEASE_INPUT,
+    INIT_STREAM_VOLUME,
+    SET_STREAM_VOLUME,
+    GET_STREAM_VOLUME
+};
+
+class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
+{
+public:
+    BpAudioPolicyService(const sp<IBinder>& impl)
+        : BpInterface<IAudioPolicyService>(impl)
+    {
+    }
+
+    virtual status_t setDeviceConnectionState(
+                                    AudioSystem::audio_devices device,
+                                    AudioSystem::device_connection_state state,
+                                    const char *device_address)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(device));
+        data.writeInt32(static_cast <uint32_t>(state));
+        data.writeCString(device_address);
+        remote()->transact(SET_DEVICE_CONNECTION_STATE, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual AudioSystem::device_connection_state getDeviceConnectionState(
+                                    AudioSystem::audio_devices device,
+                                    const char *device_address)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(device));
+        data.writeCString(device_address);
+        remote()->transact(GET_DEVICE_CONNECTION_STATE, data, &reply);
+        return static_cast <AudioSystem::device_connection_state>(reply.readInt32());
+    }
+
+    virtual status_t setPhoneState(int state)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(state);
+        remote()->transact(SET_PHONE_STATE, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual status_t setRingerMode(uint32_t mode, uint32_t mask)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(mode);
+        data.writeInt32(mask);
+        remote()->transact(SET_RINGER_MODE, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual status_t setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(usage));
+        data.writeInt32(static_cast <uint32_t>(config));
+        remote()->transact(SET_FORCE_USE, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(usage));
+        remote()->transact(GET_FORCE_USE, data, &reply);
+        return static_cast <AudioSystem::forced_config> (reply.readInt32());
+    }
+
+    virtual audio_io_handle_t getOutput(
+                                        AudioSystem::stream_type stream,
+                                        uint32_t samplingRate,
+                                        uint32_t format,
+                                        uint32_t channels,
+                                        AudioSystem::output_flags flags)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(stream));
+        data.writeInt32(samplingRate);
+        data.writeInt32(static_cast <uint32_t>(format));
+        data.writeInt32(channels);
+        data.writeInt32(static_cast <uint32_t>(flags));
+        remote()->transact(GET_OUTPUT, data, &reply);
+        audio_io_handle_t output;
+        reply.read(&output, sizeof(audio_io_handle_t));
+        return output;
+    }
+
+    virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.write(&output, sizeof(audio_io_handle_t));
+        data.writeInt32(stream);
+        remote()->transact(START_OUTPUT, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.write(&output, sizeof(audio_io_handle_t));
+        data.writeInt32(stream);
+        remote()->transact(STOP_OUTPUT, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual void releaseOutput(audio_io_handle_t output)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.write(&output, sizeof(audio_io_handle_t));
+        remote()->transact(RELEASE_OUTPUT, data, &reply);
+    }
+
+    virtual audio_io_handle_t getInput(
+                                    int inputSource,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    AudioSystem::audio_in_acoustics acoustics)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(inputSource);
+        data.writeInt32(samplingRate);
+        data.writeInt32(static_cast <uint32_t>(format));
+        data.writeInt32(channels);
+        data.writeInt32(static_cast <uint32_t>(acoustics));
+        remote()->transact(GET_INPUT, data, &reply);
+        audio_io_handle_t input;
+        reply.read(&input, sizeof(audio_io_handle_t));
+        return input;
+    }
+
+    virtual status_t startInput(audio_io_handle_t input)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.write(&input, sizeof(audio_io_handle_t));
+        remote()->transact(START_INPUT, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual status_t stopInput(audio_io_handle_t input)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.write(&input, sizeof(audio_io_handle_t));
+        remote()->transact(STOP_INPUT, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual void releaseInput(audio_io_handle_t input)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.write(&input, sizeof(audio_io_handle_t));
+        remote()->transact(RELEASE_INPUT, data, &reply);
+    }
+
+    virtual status_t initStreamVolume(AudioSystem::stream_type stream,
+                                    int indexMin,
+                                    int indexMax)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(stream));
+        data.writeInt32(indexMin);
+        data.writeInt32(indexMax);
+        remote()->transact(INIT_STREAM_VOLUME, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(stream));
+        data.writeInt32(index);
+        remote()->transact(SET_STREAM_VOLUME, data, &reply);
+        return static_cast <status_t> (reply.readInt32());
+    }
+
+    virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(stream));
+        remote()->transact(GET_STREAM_VOLUME, data, &reply);
+        int lIndex = reply.readInt32();
+        if (index) *index = lIndex;
+        return static_cast <status_t> (reply.readInt32());
+    }
+};
+
+IMPLEMENT_META_INTERFACE(AudioPolicyService, "android.media.IAudioPolicyService");
+
+// ----------------------------------------------------------------------
+
+
+status_t BnAudioPolicyService::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    switch(code) {
+        case SET_DEVICE_CONNECTION_STATE: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::audio_devices device = static_cast <AudioSystem::audio_devices>(data.readInt32());
+            AudioSystem::device_connection_state state = static_cast <AudioSystem::device_connection_state>(data.readInt32());
+            const char *device_address = data.readCString();
+            reply->writeInt32(static_cast <uint32_t>(setDeviceConnectionState(device, state, device_address)));
+            return NO_ERROR;
+        } break;
+
+        case GET_DEVICE_CONNECTION_STATE: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::audio_devices device = static_cast <AudioSystem::audio_devices>(data.readInt32());
+            const char *device_address = data.readCString();
+            reply->writeInt32(static_cast <uint32_t>(getDeviceConnectionState(device, device_address)));
+            return NO_ERROR;
+        } break;
+
+        case SET_PHONE_STATE: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            reply->writeInt32(static_cast <uint32_t>(setPhoneState(data.readInt32())));
+            return NO_ERROR;
+        } break;
+
+        case SET_RINGER_MODE: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            uint32_t mode = data.readInt32();
+            uint32_t mask = data.readInt32();
+            reply->writeInt32(static_cast <uint32_t>(setRingerMode(mode, mask)));
+            return NO_ERROR;
+        } break;
+
+        case SET_FORCE_USE: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::force_use usage = static_cast <AudioSystem::force_use>(data.readInt32());
+            AudioSystem::forced_config config = static_cast <AudioSystem::forced_config>(data.readInt32());
+            reply->writeInt32(static_cast <uint32_t>(setForceUse(usage, config)));
+            return NO_ERROR;
+        } break;
+
+        case GET_FORCE_USE: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::force_use usage = static_cast <AudioSystem::force_use>(data.readInt32());
+            reply->writeInt32(static_cast <uint32_t>(getForceUse(usage)));
+            return NO_ERROR;
+        } break;
+
+        case GET_OUTPUT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::stream_type stream = static_cast <AudioSystem::stream_type>(data.readInt32());
+            uint32_t samplingRate = data.readInt32();
+            uint32_t format = data.readInt32();
+            uint32_t channels = data.readInt32();
+            AudioSystem::output_flags flags = static_cast <AudioSystem::output_flags>(data.readInt32());
+
+            audio_io_handle_t output = getOutput(stream,
+                                                 samplingRate,
+                                                 format,
+                                                 channels,
+                                                 flags);
+            reply->write(&output, sizeof(audio_io_handle_t));
+            return NO_ERROR;
+        } break;
+
+        case START_OUTPUT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_io_handle_t output;
+            data.read(&output, sizeof(audio_io_handle_t));
+            uint32_t stream = data.readInt32();
+            reply->writeInt32(static_cast <uint32_t>(startOutput(output, (AudioSystem::stream_type)stream)));
+            return NO_ERROR;
+        } break;
+
+        case STOP_OUTPUT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_io_handle_t output;
+            data.read(&output, sizeof(audio_io_handle_t));
+            uint32_t stream = data.readInt32();
+            reply->writeInt32(static_cast <uint32_t>(stopOutput(output, (AudioSystem::stream_type)stream)));
+            return NO_ERROR;
+        } break;
+
+        case RELEASE_OUTPUT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_io_handle_t output;
+            data.read(&output, sizeof(audio_io_handle_t));
+            releaseOutput(output);
+            return NO_ERROR;
+        } break;
+
+        case GET_INPUT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            int inputSource = data.readInt32();
+            uint32_t samplingRate = data.readInt32();
+            uint32_t format = data.readInt32();
+            uint32_t channels = data.readInt32();
+            AudioSystem::audio_in_acoustics acoustics = static_cast <AudioSystem::audio_in_acoustics>(data.readInt32());
+            audio_io_handle_t input = getInput(inputSource,
+                                               samplingRate,
+                                               format,
+                                               channels,
+                                               acoustics);
+            reply->write(&input, sizeof(audio_io_handle_t));
+            return NO_ERROR;
+        } break;
+
+        case START_INPUT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_io_handle_t input;
+            data.read(&input, sizeof(audio_io_handle_t));
+            reply->writeInt32(static_cast <uint32_t>(startInput(input)));
+            return NO_ERROR;
+        } break;
+
+        case STOP_INPUT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_io_handle_t input;
+            data.read(&input, sizeof(audio_io_handle_t));
+            reply->writeInt32(static_cast <uint32_t>(stopInput(input)));
+            return NO_ERROR;
+        } break;
+
+        case RELEASE_INPUT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_io_handle_t input;
+            data.read(&input, sizeof(audio_io_handle_t));
+            releaseInput(input);
+            return NO_ERROR;
+        } break;
+
+        case INIT_STREAM_VOLUME: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::stream_type stream = static_cast <AudioSystem::stream_type>(data.readInt32());
+            int indexMin = data.readInt32();
+            int indexMax = data.readInt32();
+            reply->writeInt32(static_cast <uint32_t>(initStreamVolume(stream, indexMin,indexMax)));
+            return NO_ERROR;
+        } break;
+
+        case SET_STREAM_VOLUME: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::stream_type stream = static_cast <AudioSystem::stream_type>(data.readInt32());
+            int index = data.readInt32();
+            reply->writeInt32(static_cast <uint32_t>(setStreamVolumeIndex(stream, index)));
+            return NO_ERROR;
+        } break;
+
+        case GET_STREAM_VOLUME: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::stream_type stream = static_cast <AudioSystem::stream_type>(data.readInt32());
+            int index;
+            status_t status = getStreamVolumeIndex(stream, &index);
+            reply->writeInt32(index);
+            reply->writeInt32(static_cast <uint32_t>(status));
+            return NO_ERROR;
+        } break;
+
+        default:
+            return BBinder::onTransact(code, data, reply, flags);
+    }
+}
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index 586aacb..ee9e1d8 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -99,7 +99,7 @@
     mAudioTrack->set(AudioSystem::MUSIC,  //TODO parametrize this
             pLibConfig->sampleRate,
             1, // format = PCM 16bits per sample,
-            pLibConfig->numChannels,
+            (pLibConfig->numChannels == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,
             mTrackBufferSize,
             0);
 
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 5435da7..3ea64ae 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -1001,7 +1001,7 @@
 
    // Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
     mpAudioTrack
-            = new AudioTrack(mStreamType, 0, AudioSystem::PCM_16_BIT, 1, 0, 0, audioCallback, this, 0);
+            = new AudioTrack(mStreamType, 0, AudioSystem::PCM_16_BIT, AudioSystem::CHANNEL_OUT_MONO, 0, 0, audioCallback, this, 0);
 
     if (mpAudioTrack == 0) {
         LOGE("AudioTrack allocation failed");
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 77f7434..1d960c5 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1310,11 +1310,21 @@
     AudioTrack *t;
     if (mCallback != NULL) {
         t = new AudioTrack(
-                mStreamType, sampleRate, format, channelCount, frameCount,
-                0 /* flags */, CallbackWrapper, this);
+                mStreamType,
+                sampleRate,
+                format,
+                (channelCount == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,
+                frameCount,
+                0 /* flags */,
+                CallbackWrapper,
+                this);
     } else {
         t = new AudioTrack(
-                mStreamType, sampleRate, format, channelCount, frameCount);
+                mStreamType,
+                sampleRate,
+                format,
+                (channelCount == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,
+                frameCount);
     }
 
     if ((t == 0) || (t->initCheck() != NO_ERROR)) {
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index fbea0d4..7094cfa 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -28,6 +28,7 @@
 #include <AudioFlinger.h>
 #include <CameraService.h>
 #include <MediaPlayerService.h>
+#include <AudioPolicyService.h>
 #include <private/android_filesystem_config.h>
 
 using namespace android;
@@ -40,6 +41,7 @@
     AudioFlinger::instantiate();
     MediaPlayerService::instantiate();
     CameraService::instantiate();
+    AudioPolicyService::instantiate();
     ProcessState::self()->startThreadPool();
     IPCThreadState::self()->joinThreadPool();
 }
diff --git a/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/functional/MediaAudioTrackTest.java b/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/functional/MediaAudioTrackTest.java
index aefedc3..cea3a5a 100644
--- a/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/functional/MediaAudioTrackTest.java
+++ b/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/functional/MediaAudioTrackTest.java
@@ -140,7 +140,7 @@
         
         TestResults res = constructorTestMultiSampleRate(
                 AudioManager.STREAM_MUSIC, AudioTrack.MODE_STREAM, 
-                    AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT,
+                    AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT,
                 AudioTrack.STATE_INITIALIZED);
 
         assertTrue("testConstructorMono16MusicStream: " + res.mResultLog, res.mResult);
@@ -153,7 +153,7 @@
         
         TestResults res = constructorTestMultiSampleRate(
                 AudioManager.STREAM_MUSIC, AudioTrack.MODE_STREAM, 
-                    AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT,
+                    AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT,
                 AudioTrack.STATE_INITIALIZED);
 
         assertTrue("testConstructorStereo16MusicStream: " + res.mResultLog, res.mResult);
@@ -166,7 +166,7 @@
         
         TestResults res = constructorTestMultiSampleRate(
                 AudioManager.STREAM_MUSIC, AudioTrack.MODE_STATIC, 
-                    AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT,
+                    AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT,
                 AudioTrack.STATE_NO_STATIC_DATA);
 
         assertTrue("testConstructorMono16MusicStatic: " + res.mResultLog, res.mResult);
@@ -179,7 +179,7 @@
         
         TestResults res = constructorTestMultiSampleRate(
                 AudioManager.STREAM_MUSIC, AudioTrack.MODE_STATIC, 
-                    AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT,
+                    AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT,
                 AudioTrack.STATE_NO_STATIC_DATA);
 
         assertTrue("testConstructorStereo16MusicStatic: " + res.mResultLog, res.mResult);
@@ -196,7 +196,7 @@
         
         TestResults res = constructorTestMultiSampleRate(
                 AudioManager.STREAM_MUSIC, AudioTrack.MODE_STREAM, 
-                    AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT,
+                    AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_8BIT,
                 AudioTrack.STATE_INITIALIZED);
 
         assertTrue("testConstructorMono8MusicStream: " + res.mResultLog, res.mResult);
@@ -208,7 +208,7 @@
         
         TestResults res = constructorTestMultiSampleRate(
                 AudioManager.STREAM_MUSIC, AudioTrack.MODE_STREAM, 
-                    AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_8BIT,
+                    AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_8BIT,
                 AudioTrack.STATE_INITIALIZED);
 
         assertTrue("testConstructorStereo8MusicStream: " + res.mResultLog, res.mResult);
@@ -220,7 +220,7 @@
         
         TestResults res = constructorTestMultiSampleRate(
                 AudioManager.STREAM_MUSIC, AudioTrack.MODE_STATIC, 
-                    AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT,
+                    AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_8BIT,
                 AudioTrack.STATE_NO_STATIC_DATA);
 
         assertTrue("testConstructorMono8MusicStatic: " + res.mResultLog, res.mResult);
@@ -232,7 +232,7 @@
         
         TestResults res = constructorTestMultiSampleRate(
                 AudioManager.STREAM_MUSIC, AudioTrack.MODE_STATIC, 
-                    AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_8BIT,
+                    AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_8BIT,
                 AudioTrack.STATE_NO_STATIC_DATA);
 
         assertTrue("testConstructorStereo8MusicStatic: " + res.mResultLog, res.mResult);
@@ -248,15 +248,15 @@
     public void testConstructorStreamType() throws Exception {
         // constants for test
         final int TYPE_TEST_SR = 22050;
-        final int TYPE_TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TYPE_TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TYPE_TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TYPE_TEST_MODE = AudioTrack.MODE_STREAM;
         final int[] STREAM_TYPES = { AudioManager.STREAM_ALARM, AudioManager.STREAM_BLUETOOTH_SCO, 
                 AudioManager.STREAM_MUSIC, AudioManager.STREAM_NOTIFICATION,
                 AudioManager.STREAM_RING, AudioManager.STREAM_SYSTEM, 
-                AudioManager.STREAM_VOICE_CALL };
+                AudioManager.STREAM_VOICE_CALL, AudioManager.STREAM_DTMF, };
         final String[] STREAM_NAMES = { "STREAM_ALARM", "STREAM_BLUETOOTH_SCO", "STREAM_MUSIC",
-                "STREAM_NOTIFICATION", "STREAM_RING", "STREAM_SYSTEM", "STREAM_VOICE_CALL" };
+                "STREAM_NOTIFICATION", "STREAM_RING", "STREAM_SYSTEM", "STREAM_VOICE_CALL", "STREAM_DTMF" };
         
         boolean localTestRes = true;
         AudioTrack track = null;
@@ -303,7 +303,7 @@
         // constants for test
         final String TEST_NAME = "testPlaybackHeadPositionAfterInit";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -324,7 +324,7 @@
         // constants for test
         final String TEST_NAME = "testPlaybackHeadPositionIncrease";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -352,7 +352,7 @@
         // constants for test
         final String TEST_NAME = "testPlaybackHeadPositionAfterFlush";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -382,7 +382,7 @@
         // constants for test
         final String TEST_NAME = "testPlaybackHeadPositionAfterStop";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -413,7 +413,7 @@
         // constants for test
         final String TEST_NAME = "testPlaybackHeadPositionAfterPause";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -448,7 +448,7 @@
         // constants for test
         final String TEST_NAME = "testSetStereoVolumeMax";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -474,7 +474,7 @@
         // constants for test
         final String TEST_NAME = "testSetStereoVolumeMin";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -500,7 +500,7 @@
         // constants for test
         final String TEST_NAME = "testSetStereoVolumeMid";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -526,7 +526,7 @@
         // constants for test
         final String TEST_NAME = "testSetPlaybackRate";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -552,7 +552,7 @@
         // constants for test
         final String TEST_NAME = "testSetPlaybackRateZero";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -574,7 +574,7 @@
         // constants for test
         final String TEST_NAME = "testSetPlaybackRateTwiceOutputSR";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -601,7 +601,7 @@
         // constants for test
         final String TEST_NAME = "testSetGetPlaybackRate";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -628,7 +628,7 @@
         // constants for test
         final String TEST_NAME = "testSetPlaybackRateUninit";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STATIC;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -655,7 +655,7 @@
         // constants for test
         final String TEST_NAME = "testSetPlaybackHeadPositionPlaying";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -682,7 +682,7 @@
         // constants for test
         final String TEST_NAME = "testSetPlaybackHeadPositionStopped";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -710,7 +710,7 @@
         // constants for test
         final String TEST_NAME = "testSetPlaybackHeadPositionPaused";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -738,7 +738,7 @@
         // constants for test
         final String TEST_NAME = "testSetPlaybackHeadPositionTooFar";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -770,7 +770,7 @@
         // constants for test
         final String TEST_NAME = "testSetLoopPointsStream";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -794,7 +794,7 @@
         // constants for test
         final String TEST_NAME = "testSetLoopPointsStartAfterEnd";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STATIC;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -818,7 +818,7 @@
         // constants for test
         final String TEST_NAME = "testSetLoopPointsSuccess";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STATIC;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -842,7 +842,7 @@
         // constants for test
         final String TEST_NAME = "testSetLoopPointsLoopTooLong";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STATIC;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -868,7 +868,7 @@
         // constants for test
         final String TEST_NAME = "testSetLoopPointsStartTooFar";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STATIC;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -896,7 +896,7 @@
         // constants for test
         final String TEST_NAME = "testSetLoopPointsEndTooFar";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STATIC;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -929,7 +929,7 @@
         // constants for test
         final String TEST_NAME = "testWriteByteOffsetTooBig";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -953,7 +953,7 @@
         // constants for test
         final String TEST_NAME = "testWriteShortOffsetTooBig";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -977,7 +977,7 @@
         // constants for test
         final String TEST_NAME = "testWriteByteSizeTooBig";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1001,7 +1001,7 @@
         // constants for test
         final String TEST_NAME = "testWriteShortSizeTooBig";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1025,7 +1025,7 @@
         // constants for test
         final String TEST_NAME = "testWriteByteNegativeOffset";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1049,7 +1049,7 @@
         // constants for test
         final String TEST_NAME = "testWriteShortNegativeOffset";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1073,7 +1073,7 @@
         // constants for test
         final String TEST_NAME = "testWriteByteNegativeSize";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1097,7 +1097,7 @@
         // constants for test
         final String TEST_NAME = "testWriteShortNegativeSize";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1121,7 +1121,7 @@
         // constants for test
         final String TEST_NAME = "testWriteByte";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1145,7 +1145,7 @@
         // constants for test
         final String TEST_NAME = "testWriteShort";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1169,7 +1169,7 @@
         // constants for test
         final String TEST_NAME = "testWriteByte8bit";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_8BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1193,7 +1193,7 @@
         // constants for test
         final String TEST_NAME = "testWriteShort8bit";
         final int TEST_SR = 22050;
-        final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+        final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
         final int TEST_FORMAT = AudioFormat.ENCODING_PCM_8BIT;
         final int TEST_MODE = AudioTrack.MODE_STREAM;
         final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1221,7 +1221,7 @@
       // constant for test
       final String TEST_NAME = "testGetMinBufferSizeTooLowSR";
       final int TEST_SR = 3999;
-      final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+      final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
       final int TEST_FORMAT = AudioFormat.ENCODING_PCM_8BIT;
       final int TEST_MODE = AudioTrack.MODE_STREAM;
       final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
@@ -1238,7 +1238,7 @@
       // constant for testg
       final String TEST_NAME = "testGetMinBufferSizeTooHighSR";
       final int TEST_SR = 48001;
-      final int TEST_CONF = AudioFormat.CHANNEL_CONFIGURATION_MONO;
+      final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO;
       final int TEST_FORMAT = AudioFormat.ENCODING_PCM_8BIT;
       final int TEST_MODE = AudioTrack.MODE_STREAM;
       final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
diff --git a/packages/SettingsProvider/src/com/android/providers/settings/DatabaseHelper.java b/packages/SettingsProvider/src/com/android/providers/settings/DatabaseHelper.java
index 2b888e4..f00fd39 100644
--- a/packages/SettingsProvider/src/com/android/providers/settings/DatabaseHelper.java
+++ b/packages/SettingsProvider/src/com/android/providers/settings/DatabaseHelper.java
@@ -64,7 +64,7 @@
 
     private static final String TAG = "SettingsProvider";
     private static final String DATABASE_NAME = "settings.db";
-    private static final int DATABASE_VERSION = 36;
+    private static final int DATABASE_VERSION = 37;
 
     private Context mContext;
 
@@ -414,7 +414,27 @@
             }
             upgradeVersion = 36;
         }
-        
+        if (upgradeVersion == 36) {
+           // This upgrade adds the STREAM_SYSTEM_ENFORCED type to the list of
+            // types affected by ringer modes (silent, vibrate, etc.)
+            db.beginTransaction();
+            try {
+                db.execSQL("DELETE FROM system WHERE name='"
+                        + Settings.System.MODE_RINGER_STREAMS_AFFECTED + "'");
+                int newValue = (1 << AudioManager.STREAM_RING)
+                        | (1 << AudioManager.STREAM_NOTIFICATION)
+                        | (1 << AudioManager.STREAM_SYSTEM)
+                        | (1 << AudioManager.STREAM_SYSTEM_ENFORCED);
+                db.execSQL("INSERT INTO system ('name', 'value') values ('"
+                        + Settings.System.MODE_RINGER_STREAMS_AFFECTED + "', '"
+                        + String.valueOf(newValue) + "')");
+                db.setTransactionSuccessful();
+            } finally {
+                db.endTransaction();
+            }
+            upgradeVersion = 36;
+        }
+
         if (upgradeVersion != currentVersion) {
             Log.w(TAG, "Got stuck trying to upgrade from version " + upgradeVersion
                     + ", must wipe the settings provider");
@@ -575,7 +595,7 @@
         // By default, only the ring/notification and system streams are affected
         loadSetting(stmt, Settings.System.MODE_RINGER_STREAMS_AFFECTED,
                 (1 << AudioManager.STREAM_RING) | (1 << AudioManager.STREAM_NOTIFICATION) |
-                (1 << AudioManager.STREAM_SYSTEM));
+                (1 << AudioManager.STREAM_SYSTEM) | (1 << AudioManager.STREAM_SYSTEM_ENFORCED));
 
         loadSetting(stmt, Settings.System.MUTE_STREAMS_AFFECTED,
                 ((1 << AudioManager.STREAM_MUSIC) |
diff --git a/packages/TtsService/jni/android_tts_SynthProxy.cpp b/packages/TtsService/jni/android_tts_SynthProxy.cpp
index 99d7723..a4090cf 100644
--- a/packages/TtsService/jni/android_tts_SynthProxy.cpp
+++ b/packages/TtsService/jni/android_tts_SynthProxy.cpp
@@ -71,7 +71,7 @@
         AudioTrack*               mAudioOut;
         AudioSystem::stream_type  mStreamType;
         uint32_t                  mSampleRate;
-        AudioSystem::audio_format mAudFormat;
+        uint32_t                  mAudFormat;
         int                       mNbChannels;
         int8_t *                  mBuffer;
         size_t                    mBufferSize;
@@ -118,7 +118,6 @@
             mSampleRate = rate;
             mAudFormat  = format;
             mNbChannels = channel;
-
             mStreamType = streamType;
 
             // retrieve system properties to ensure successful creation of the
@@ -139,7 +138,8 @@
             if (minBufCount < 2) minBufCount = 2;
             int minFrameCount = (afFrameCount * rate * minBufCount)/afSampleRate;
 
-            mAudioOut = new AudioTrack(mStreamType, rate, format, channel,
+            mAudioOut = new AudioTrack(mStreamType, rate, format,
+                    (channel == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,
                     minFrameCount > 4096 ? minFrameCount : 4096,
                     0, 0, 0, 0); // not using an AudioTrack callback
 
@@ -182,7 +182,7 @@
  * Directly speaks using AudioTrack or write to file
  */
 static tts_callback_status ttsSynthDoneCB(void *& userdata, uint32_t rate,
-                           AudioSystem::audio_format format, int channel,
+                           uint32_t format, int channel,
                            int8_t *&wav, size_t &bufferSize, tts_synth_status status) {
     //LOGV("ttsSynthDoneCallback: %d bytes", bufferSize);
 
@@ -202,7 +202,7 @@
         }
 
         if (bufferSize > 0) {
-            prepAudioTrack(pJniData, pForAfter->streamType, rate, format, channel);
+            prepAudioTrack(pJniData, pForAfter->streamType, rate, (AudioSystem::audio_format)format, channel);
             if (pJniData->mAudioOut) {
                 pJniData->mAudioOut->write(wav, bufferSize);
                 memset(wav, 0, bufferSize);
diff --git a/services/java/com/android/server/HeadsetObserver.java b/services/java/com/android/server/HeadsetObserver.java
index 9b0a2d4..bee3108 100644
--- a/services/java/com/android/server/HeadsetObserver.java
+++ b/services/java/com/android/server/HeadsetObserver.java
@@ -41,10 +41,16 @@
     private static final String HEADSET_STATE_PATH = "/sys/class/switch/h2w/state";
     private static final String HEADSET_NAME_PATH = "/sys/class/switch/h2w/name";
 
+    private static final int BIT_HEADSET = (1 << 0);
+    private static final int BIT_HEADSET_NO_MIC = (1 << 1);
+    private static final int BIT_TTY = (1 << 2);
+    private static final int BIT_FM_HEADSET = (1 << 3);
+    private static final int BIT_FM_SPEAKER = (1 << 4);
+
     private int mHeadsetState;
+    private int mPrevHeadsetState;
     private String mHeadsetName;
-    private boolean mAudioRouteNeedsUpdate;
-    private AudioManager mAudioManager;
+    private boolean mPendingIntent;
 
     private final Context mContext;
     private final WakeLock mWakeLock;  // held while there is a pending route change
@@ -76,6 +82,7 @@
 
         String newName = mHeadsetName;
         int newState = mHeadsetState;
+        mPrevHeadsetState = mHeadsetState;
         try {
             FileReader file = new FileReader(HEADSET_STATE_PATH);
             int len = file.read(buffer, 0, 1024);
@@ -91,20 +98,25 @@
             Log.e(TAG, "" , e);
         }
 
-        mAudioManager = (AudioManager) mContext.getSystemService(Context.AUDIO_SERVICE);
         update(newName, newState);
     }
 
     private synchronized final void update(String newName, int newState) {
         if (newName != mHeadsetName || newState != mHeadsetState) {
-            boolean isUnplug = (newState == 0 && mHeadsetState == 1);
+            boolean isUnplug = false;
+            if ( (mHeadsetState & BIT_HEADSET) > 0 || (mHeadsetState & BIT_HEADSET_NO_MIC) > 0) {
+                if ((newState & BIT_HEADSET) == 0 && (newState & BIT_HEADSET_NO_MIC) == 0)
+                    isUnplug = true;
+            }
             mHeadsetName = newName;
+            mPrevHeadsetState = mHeadsetState;
             mHeadsetState = newState;
-            mAudioRouteNeedsUpdate = true;
-
-            sendIntent(isUnplug);
+            mPendingIntent = true;
 
             if (isUnplug) {
+                Intent intent = new Intent(AudioManager.ACTION_AUDIO_BECOMING_NOISY);
+                mContext.sendBroadcast(intent);
+
                 // It can take hundreds of ms flush the audio pipeline after
                 // apps pause audio playback, but audio route changes are
                 // immediate, so delay the route change by 1000ms.
@@ -113,12 +125,13 @@
                 mWakeLock.acquire();
                 mHandler.sendEmptyMessageDelayed(0, 1000);
             } else {
-                updateAudioRoute();
+                sendIntent();
+                mPendingIntent = false;
             }
         }
     }
 
-    private synchronized final void sendIntent(boolean isUnplug) {
+    private synchronized final void sendIntent() {
         //  Pack up the values and broadcast them to everyone
         Intent intent = new Intent(Intent.ACTION_HEADSET_PLUG);
         intent.addFlags(Intent.FLAG_RECEIVER_REGISTERED_ONLY);
@@ -128,24 +141,15 @@
 
         // TODO: Should we require a permission?
         ActivityManagerNative.broadcastStickyIntent(intent, null);
-
-        if (isUnplug) {
-            intent = new Intent(AudioManager.ACTION_AUDIO_BECOMING_NOISY);
-            mContext.sendBroadcast(intent);
-        }
-    }
-
-    private synchronized final void updateAudioRoute() {
-        if (mAudioRouteNeedsUpdate) {
-            mAudioManager.setWiredHeadsetOn(mHeadsetState == 1);
-            mAudioRouteNeedsUpdate = false;
-        }
     }
 
     private final Handler mHandler = new Handler() {
         @Override
         public void handleMessage(Message msg) {
-            updateAudioRoute();
+            if (mPendingIntent) {
+                sendIntent();
+                mPendingIntent = false;
+            }
             mWakeLock.release();
         }
     };