diff --git a/libs/audioflinger/A2dpAudioInterface.cpp b/libs/audioflinger/A2dpAudioInterface.cpp
index b6d5078..16a4f2d 100644
--- a/libs/audioflinger/A2dpAudioInterface.cpp
+++ b/libs/audioflinger/A2dpAudioInterface.cpp
@@ -71,8 +71,8 @@
 }
 
 AudioStreamIn* A2dpAudioInterface::openInputStream(
-        int format, int channelCount, uint32_t sampleRate, status_t *status,
-        AudioSystem::audio_in_acoustics acoustics)
+        int inputSource, int format, int channelCount, uint32_t sampleRate,
+        status_t *status, AudioSystem::audio_in_acoustics acoustics)
 {
     if (status)
         *status = -1;
diff --git a/libs/audioflinger/A2dpAudioInterface.h b/libs/audioflinger/A2dpAudioInterface.h
index 7901a8c..091e775 100644
--- a/libs/audioflinger/A2dpAudioInterface.h
+++ b/libs/audioflinger/A2dpAudioInterface.h
@@ -55,6 +55,7 @@
                                 status_t *status=0);
 
     virtual AudioStreamIn* openInputStream(
+                                int inputSource,
                                 int format,
                                 int channelCount,
                                 uint32_t sampleRate,
diff --git a/libs/audioflinger/AudioBufferProvider.h b/libs/audioflinger/AudioBufferProvider.h
index 1a467c7..81c5c39 100644
--- a/libs/audioflinger/AudioBufferProvider.h
+++ b/libs/audioflinger/AudioBufferProvider.h
@@ -36,6 +36,8 @@
         };
         size_t frameCount;
     };
+
+    virtual ~AudioBufferProvider() {}
     
     virtual status_t getNextBuffer(Buffer* buffer) = 0;
     virtual void releaseBuffer(Buffer* buffer) = 0;
diff --git a/libs/audioflinger/AudioDumpInterface.h b/libs/audioflinger/AudioDumpInterface.h
index 9a94102..b72c94e 100644
--- a/libs/audioflinger/AudioDumpInterface.h
+++ b/libs/audioflinger/AudioDumpInterface.h
@@ -78,9 +78,9 @@
     virtual status_t    setParameter(const char* key, const char* value)
                             {return mFinalInterface->setParameter(key, value);}
 
-    virtual AudioStreamIn* openInputStream( int format, int channelCount, uint32_t sampleRate, status_t *status,
-                                            AudioSystem::audio_in_acoustics acoustics)
-                            {return mFinalInterface->openInputStream( format, channelCount, sampleRate, status, acoustics);}
+    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 status_t    dump(int fd, const Vector<String16>& args) { return mFinalInterface->dumpState(fd, args); }
 
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index b56221f..8a19fbd 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -499,7 +499,8 @@
     }
 
 #ifdef WITH_A2DP
-    LOGD("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(), IPCThreadState::self()->getCallingPid());
+    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);
@@ -817,19 +818,22 @@
         {
             AutoMutex lock(mHardwareLock);
             if (mForcedSpeakerCount++ == 0) {
-                mRouteRestoreTime = 0;
-                mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
-                if (mForcedRoute == 0 && !(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);
+                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);
         }
@@ -890,7 +894,7 @@
             }
             LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount);
         } else {
-            LOGE("mA2dpDisableCount is already zero");
+            LOGV("mA2dpDisableCount is already zero");
         }
     }
 }
@@ -1277,7 +1281,7 @@
     status_t lStatus;
     
     // Resampler implementation limits input sampling rate to 2 x output sampling rate.
-    if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) {
+    if (sampleRate > mSampleRate*2) {
         LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
         lStatus = BAD_VALUE;
         goto Exit;
@@ -1553,7 +1557,6 @@
 AudioFlinger::MixerThread::TrackBase::TrackBase(
             const sp<MixerThread>& mixerThread,
             const sp<Client>& client,
-            int streamType,
             uint32_t sampleRate,
             int format,
             int channelCount,
@@ -1563,7 +1566,6 @@
     :   RefBase(),
         mMixerThread(mixerThread),
         mClient(client),
-        mStreamType(streamType),
         mFrameCount(0),
         mState(IDLE),
         mClientTid(-1),
@@ -1594,8 +1596,8 @@
                 new(mCblk) audio_track_cblk_t();
                 // clear all buffers
                 mCblk->frameCount = frameCount;
-                mCblk->sampleRate = (uint16_t)sampleRate;
-                mCblk->channels = (uint16_t)channelCount;
+                mCblk->sampleRate = sampleRate;
+                mCblk->channels = (uint8_t)channelCount;
                 if (sharedBuffer == 0) {
                     mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
                     memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
@@ -1618,8 +1620,8 @@
            new(mCblk) audio_track_cblk_t();
            // clear all buffers
            mCblk->frameCount = frameCount;
-           mCblk->sampleRate = (uint16_t)sampleRate;
-           mCblk->channels = (uint16_t)channelCount;
+           mCblk->sampleRate = sampleRate;
+           mCblk->channels = (uint8_t)channelCount;
            mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
            memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
            // Force underrun condition to avoid false underrun callback until first data is
@@ -1680,7 +1682,7 @@
 }
 
 int AudioFlinger::MixerThread::TrackBase::channelCount() const {
-    return mCblk->channels;
+    return (int)mCblk->channels;
 }
 
 void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
@@ -1713,12 +1715,13 @@
             int channelCount,
             int frameCount,
             const sp<IMemory>& sharedBuffer)
-    :   TrackBase(mixerThread, client, streamType, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
+    :   TrackBase(mixerThread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
 {
     mVolume[0] = 1.0f;
     mVolume[1] = 1.0f;
     mMute = false;
     mSharedBuffer = sharedBuffer;
+    mStreamType = streamType;
 }
 
 AudioFlinger::MixerThread::Track::~Track()
@@ -1902,15 +1905,15 @@
 AudioFlinger::MixerThread::RecordTrack::RecordTrack(
             const sp<MixerThread>& mixerThread,
             const sp<Client>& client,
-            int streamType,
+            int inputSource,
             uint32_t sampleRate,
             int format,
             int channelCount,
             int frameCount,
             uint32_t flags)
-    :   TrackBase(mixerThread, client, streamType, sampleRate, format,
+    :   TrackBase(mixerThread, client, sampleRate, format,
                   channelCount, frameCount, flags, 0),
-        mOverflow(false)
+        mOverflow(false), mInputSource(inputSource)
 {
 }
 
@@ -2235,7 +2238,7 @@
 
 sp<IAudioRecord> AudioFlinger::openRecord(
         pid_t pid,
-        int streamType,
+        int inputSource,
         uint32_t sampleRate,
         int format,
         int channelCount,
@@ -2258,18 +2261,12 @@
         goto Exit;
     }
 
-    if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
+    if (uint32_t(inputSource) >= AudioRecord::NUM_INPUT_SOURCES) {
         LOGE("invalid stream type");
         lStatus = BAD_VALUE;
         goto Exit;
     }
 
-    if (sampleRate > MAX_SAMPLE_RATE) {
-        LOGE("Sample rate out of range");
-        lStatus = BAD_VALUE;
-        goto Exit;
-    }
-
     if (mAudioRecordThread == 0) {
         LOGE("Audio record thread not started");
         lStatus = NO_INIT;
@@ -2301,7 +2298,7 @@
         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, streamType, sampleRate,
+        recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, inputSource, sampleRate,
                                                    format, channelCount, frameCount, flags);
     }
     if (recordTrack->getCblk() == NULL) {
@@ -2407,7 +2404,9 @@
                
                 LOGV("AudioRecordThread: loop starting");
                 if (mRecordTrack != 0) {
-                    input = mAudioHardware->openInputStream(mRecordTrack->format(), 
+                    input = mAudioHardware->openInputStream(
+                                    mRecordTrack->inputSource(),
+                                    mRecordTrack->format(), 
                                     mRecordTrack->channelCount(), 
                                     mRecordTrack->sampleRate(), 
                                     &mStartStatus,
diff --git a/libs/audioflinger/AudioFlinger.h b/libs/audioflinger/AudioFlinger.h
index c7ca9ec..8e47b29 100644
--- a/libs/audioflinger/AudioFlinger.h
+++ b/libs/audioflinger/AudioFlinger.h
@@ -139,7 +139,7 @@
     // record interface
     virtual sp<IAudioRecord> openRecord(
                                 pid_t pid,
-                                int streamType,
+                                int inputSource,
                                 uint32_t sampleRate,
                                 int format,
                                 int channelCount,
@@ -232,7 +232,6 @@
 
                                 TrackBase(const sp<MixerThread>& mixerThread,
                                         const sp<Client>& client,
-                                        int streamType,
                                         uint32_t sampleRate,
                                         int format,
                                         int channelCount,
@@ -260,10 +259,6 @@
                 return mCblk;
             }
 
-            int type() const {
-                return mStreamType;
-            }
-
             int format() const {
                 return mFormat;
             }
@@ -293,7 +288,6 @@
             sp<Client>          mClient;
             sp<IMemory>         mCblkMemory;
             audio_track_cblk_t* mCblk;
-            int                 mStreamType;
             void*               mBuffer;
             void*               mBufferEnd;
             uint32_t            mFrameCount;
@@ -328,6 +322,11 @@
                     void        mute(bool);
                     void        setVolume(float left, float right);
 
+                    int type() const {
+                        return mStreamType;
+                    }
+
+
         protected:
             friend class MixerThread;
             friend class AudioFlinger;
@@ -364,6 +363,7 @@
             int8_t              mRetryCount;
             sp<IMemory>         mSharedBuffer;
             bool                mResetDone;
+            int                 mStreamType;
         };  // end of Track
 
         // record track
@@ -371,7 +371,7 @@
         public:
                                 RecordTrack(const sp<MixerThread>& mixerThread,
                                         const sp<Client>& client,
-                                        int streamType,
+                                        int inputSource,
                                         uint32_t sampleRate,
                                         int format,
                                         int channelCount,
@@ -385,6 +385,8 @@
                     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;
@@ -397,6 +399,7 @@
             virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
 
             bool                mOverflow;
+            int                 mInputSource;
         };
 
         // playback track
diff --git a/libs/audioflinger/AudioHardwareGeneric.cpp b/libs/audioflinger/AudioHardwareGeneric.cpp
index 62beada..1e159b8 100644
--- a/libs/audioflinger/AudioHardwareGeneric.cpp
+++ b/libs/audioflinger/AudioHardwareGeneric.cpp
@@ -30,6 +30,7 @@
 #include <utils/String8.h>
 
 #include "AudioHardwareGeneric.h"
+#include <media/AudioRecord.h>
 
 namespace android {
 
@@ -93,9 +94,15 @@
 }
 
 AudioStreamIn* AudioHardwareGeneric::openInputStream(
-        int format, int channelCount, uint32_t sampleRate, status_t *status,
-        AudioSystem::audio_in_acoustics acoustics)
+        int inputSource, int format, int channelCount, 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)) {
+        return 0;
+    }
+
     AutoMutex lock(mLock);
 
     // only one input stream allowed
diff --git a/libs/audioflinger/AudioHardwareGeneric.h b/libs/audioflinger/AudioHardwareGeneric.h
index c949aa1..c89df87 100644
--- a/libs/audioflinger/AudioHardwareGeneric.h
+++ b/libs/audioflinger/AudioHardwareGeneric.h
@@ -112,6 +112,7 @@
             status_t *status=0);
 
     virtual AudioStreamIn* openInputStream(
+            int inputSource,
             int format,
             int channelCount,
             uint32_t sampleRate,
diff --git a/libs/audioflinger/AudioHardwareStub.cpp b/libs/audioflinger/AudioHardwareStub.cpp
index b13cb1c..0ab4c60 100644
--- a/libs/audioflinger/AudioHardwareStub.cpp
+++ b/libs/audioflinger/AudioHardwareStub.cpp
@@ -23,6 +23,7 @@
 #include <utils/String8.h>
 
 #include "AudioHardwareStub.h"
+#include <media/AudioRecord.h>
 
 namespace android {
 
@@ -56,9 +57,15 @@
 }
 
 AudioStreamIn* AudioHardwareStub::openInputStream(
-        int format, int channelCount, uint32_t sampleRate,
+        int inputSource, int format, int channelCount, 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)) {
+        return 0;
+    }
+
     AudioStreamInStub* in = new AudioStreamInStub();
     status_t lStatus = in->set(format, channelCount, sampleRate, acoustics);
     if (status) {
diff --git a/libs/audioflinger/AudioHardwareStub.h b/libs/audioflinger/AudioHardwareStub.h
index d406424..bf63cc5 100644
--- a/libs/audioflinger/AudioHardwareStub.h
+++ b/libs/audioflinger/AudioHardwareStub.h
@@ -78,6 +78,7 @@
                                 status_t *status=0);
 
     virtual AudioStreamIn* openInputStream(
+                                int inputSource,
                                 int format,
                                 int channelCount,
                                 uint32_t sampleRate,