auto import from //branches/cupcake/...@131421

22 files changed, 242 insertions, 124 deletions

diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 6bd54ba..7437f65 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -29,8 +29,27 @@ class AudioSystem
 {
 public:
 
+    enum stream_type {
+        DEFAULT         =-1,
+        VOICE_CALL      = 0,
+        SYSTEM          = 1,
+        RING            = 2,
+        MUSIC           = 3,
+        ALARM           = 4,
+        NOTIFICATION    = 5,
+        BLUETOOTH_SCO   = 6,
+        NUM_STREAM_TYPES
+    };
+
+    enum audio_output_type {
+        AUDIO_OUTPUT_DEFAULT      =-1,
+        AUDIO_OUTPUT_HARDWARE     = 0,
+        AUDIO_OUTPUT_A2DP         = 1,
+        NUM_AUDIO_OUTPUT_TYPES
+    };
+
     enum audio_format {
-        DEFAULT = 0,
+        FORMAT_DEFAULT = 0,
         PCM_16_BIT,
         PCM_8_BIT,
         INVALID_FORMAT
@@ -96,9 +115,11 @@ public:
     static float linearToLog(int volume);
     static int logToLinear(float volume);
 
-    static status_t getOutputSamplingRate(int* samplingRate);
-    static status_t getOutputFrameCount(int* frameCount);
-    static status_t getOutputLatency(uint32_t* latency);
+    static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT);
+    static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT);
+    static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT);
+
+    static bool routedToA2dpOutput(int streamType);
     
     static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, 
         size_t* buffSize);
@@ -117,9 +138,10 @@ private:
         virtual void binderDied(const wp<IBinder>& who);
         
         // IAudioFlingerClient
-        virtual void audioOutputChanged(uint32_t frameCount, uint32_t samplingRate, uint32_t latency);
+        virtual void a2dpEnabledChanged(bool enabled);
         
     };
+    static int getOutput(int streamType);
 
     static sp<AudioFlingerClient> gAudioFlingerClient;
 
@@ -128,9 +150,10 @@ private:
     static Mutex gLock;
     static sp<IAudioFlinger> gAudioFlinger;
     static audio_error_callback gAudioErrorCallback;
-    static int gOutSamplingRate;
-    static int gOutFrameCount;
-    static uint32_t gOutLatency;
+    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
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 5b2bab9..659f5f8 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -42,19 +42,6 @@ class audio_track_cblk_t;
 class AudioTrack
 {
 public:
-
-    enum stream_type {
-        DEFAULT         =-1,
-        VOICE_CALL      = 0,
-        SYSTEM          = 1,
-        RING            = 2,
-        MUSIC           = 3,
-        ALARM           = 4,
-        NOTIFICATION    = 5,
-        BLUETOOTH_SCO   = 6,
-        NUM_STREAM_TYPES
-    };
-
     enum channel_index {
         MONO   = 0,
         LEFT   = 0,
@@ -128,7 +115,7 @@ public:
      * Parameters:
      *
      * streamType:         Select the type of audio stream this track is attached to
-     *                     (e.g. AudioTrack::MUSIC).
+     *                     (e.g. AudioSystem::MUSIC).
      * sampleRate:         Track sampling rate in Hz.
      * format:             PCM sample format (e.g AudioSystem::PCM_16_BIT for signed
      *                     16 bits per sample).
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index df601d7..6f13fe0 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -65,11 +65,11 @@ public:
     /* query the audio hardware state. This state never changes,
      * and therefore can be cached.
      */
-    virtual     uint32_t    sampleRate() const = 0;
-    virtual     int         channelCount() const = 0;
-    virtual     int         format() const = 0;
-    virtual     size_t      frameCount() const = 0;
-    virtual     uint32_t    latency() const = 0;
+    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;
 
     /* set/get the audio hardware state. This will probably be used by
      * the preference panel, mostly.
@@ -117,6 +117,9 @@ public:
     
     // force AudioFlinger thread out of standby
     virtual     void        wakeUp() = 0;
+
+    // is A2DP output enabled
+    virtual     bool        isA2dpEnabled() const = 0;
 };
 
 
diff --git a/include/media/IAudioFlingerClient.h b/include/media/IAudioFlingerClient.h
index 10c3e0f..c3deb0b 100644
--- a/include/media/IAudioFlingerClient.h
+++ b/include/media/IAudioFlingerClient.h
@@ -32,7 +32,7 @@ public:
     DECLARE_META_INTERFACE(AudioFlingerClient);
 
     // Notifies a change of audio output from/to hardware to/from A2DP.
-    virtual void audioOutputChanged(uint32_t frameCount, uint32_t samplingRate, uint32_t latency) = 0;
+    virtual void a2dpEnabledChanged(bool enabled) = 0;
 
 };
 
diff --git a/include/media/IMediaRecorder.h b/include/media/IMediaRecorder.h
index 49e45d1..0dff84e 100644
--- a/include/media/IMediaRecorder.h
+++ b/include/media/IMediaRecorder.h
@@ -38,6 +38,7 @@ public:
     virtual	status_t		setVideoEncoder(int ve) = 0;
     virtual	status_t		setAudioEncoder(int ae) = 0;
     virtual	status_t		setOutputFile(const char* path) = 0;
+    virtual	status_t		setOutputFile(int fd, int64_t offset, int64_t length) = 0;
     virtual	status_t		setVideoSize(int width, int height) = 0;
     virtual	status_t		setVideoFrameRate(int frames_per_second) = 0;
     virtual	status_t		prepare() = 0;
diff --git a/include/media/PVMediaRecorder.h b/include/media/PVMediaRecorder.h
index 5fee0d6..f795d04 100644
--- a/include/media/PVMediaRecorder.h
+++ b/include/media/PVMediaRecorder.h
@@ -43,6 +43,7 @@ public:
     status_t setCamera(const sp<ICamera>& camera);
     status_t setPreviewSurface(const sp<ISurface>& surface);
     status_t setOutputFile(const char *path);
+    status_t setOutputFile(int fd, int64_t offset, int64_t length);
     status_t prepare();
     status_t start();
     status_t stop();
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index a901d32..436e8f1 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -102,6 +102,7 @@ public:
     status_t    setVideoEncoder(int ve);
     status_t    setAudioEncoder(int ae);
     status_t    setOutputFile(const char* path);
+    status_t    setOutputFile(int fd, int64_t offset, int64_t length);
     status_t    setVideoSize(int width, int height);
     status_t    setVideoFrameRate(int frames_per_second);
     status_t    prepare();
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index a987b92..3d39181 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -128,22 +128,8 @@ status_t AudioRecord::set(
         return BAD_VALUE;
     }
 
-    size_t inputBuffSizeInBytes = -1;
-    if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &inputBuffSizeInBytes)
-        != NO_ERROR) {
-            LOGE("AudioSystem could not query the input buffer size.");
-            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);
-    
-    // We use 2* size of input buffer for ping pong use of record buffer.
-    int minFrameCount = 2 * inputBuffSizeInBytes / frameSizeInBytes;
-    LOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
+    // TODO: Get input frame count from hardware.
+    int minFrameCount = 1024*2;
 
     if (frameCount == 0) {
         frameCount = minFrameCount;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index cf91105..63dfc3b 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -20,7 +20,6 @@
 #include <utils/Log.h>
 #include <utils/IServiceManager.h>
 #include <media/AudioSystem.h>
-#include <media/AudioTrack.h>
 #include <math.h>
 
 namespace android {
@@ -31,9 +30,10 @@ sp<IAudioFlinger> AudioSystem::gAudioFlinger;
 sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
 audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
 // Cached values
-int AudioSystem::gOutSamplingRate = 0;
-int AudioSystem::gOutFrameCount = 0;
-uint32_t AudioSystem::gOutLatency = 0;
+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;
 // Cached values for recording queries
 uint32_t AudioSystem::gPrevInSamplingRate = 16000;
 int AudioSystem::gPrevInFormat = AudioSystem::PCM_16_BIT;
@@ -66,9 +66,12 @@ const sp<IAudioFlinger>& AudioSystem::get_audio_flinger()
         gAudioFlinger = interface_cast<IAudioFlinger>(binder);
         gAudioFlinger->registerClient(gAudioFlingerClient);
         // Cache frequently accessed parameters 
-        gOutFrameCount = (int)gAudioFlinger->frameCount();
-        gOutSamplingRate = (int)gAudioFlinger->sampleRate();
-        gOutLatency = gAudioFlinger->latency();
+        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;
@@ -147,7 +150,7 @@ status_t AudioSystem::getMasterMute(bool* mute)
 
 status_t AudioSystem::setStreamVolume(int stream, float value)
 {
-    if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) return BAD_VALUE;
+    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);
@@ -156,7 +159,7 @@ status_t AudioSystem::setStreamVolume(int stream, float value)
 
 status_t AudioSystem::setStreamMute(int stream, bool mute)
 {
-    if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) return BAD_VALUE;
+    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->setStreamMute(stream, mute);
@@ -165,7 +168,7 @@ status_t AudioSystem::setStreamMute(int stream, bool mute)
 
 status_t AudioSystem::getStreamVolume(int stream, float* volume)
 {
-    if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) return BAD_VALUE;
+    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);
@@ -174,7 +177,7 @@ status_t AudioSystem::getStreamVolume(int stream, float* volume)
 
 status_t AudioSystem::getStreamMute(int stream, bool* mute)
 {
-    if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) return BAD_VALUE;
+    if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     *mute = af->streamMute(stream);
@@ -252,37 +255,48 @@ int AudioSystem::logToLinear(float volume)
     return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
 }
 
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate)
+status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
 {
-    if (gOutSamplingRate == 0) {
+    int output = getOutput(streamType);
+
+    if (gOutSamplingRate[output] == 0) {
         const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
         if (af == 0) return PERMISSION_DENIED;
         // gOutSamplingRate is updated by get_audio_flinger()
     }
-    *samplingRate = gOutSamplingRate;
+    LOGV("getOutputSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, gOutSamplingRate[output]);
+    *samplingRate = gOutSamplingRate[output];
     
     return NO_ERROR;
 }
 
-status_t AudioSystem::getOutputFrameCount(int* frameCount)
+status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
 {
-    if (gOutFrameCount == 0) {
+    int output = getOutput(streamType);
+
+    if (gOutFrameCount[output] == 0) {
         const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
         if (af == 0) return PERMISSION_DENIED;
         // gOutFrameCount is updated by get_audio_flinger()
     }
-    *frameCount = gOutFrameCount;
+    LOGV("getOutputFrameCount() streamType %d, output %d, frame count %d", streamType, output, gOutFrameCount[output]);
+
+    *frameCount = gOutFrameCount[output];
     return NO_ERROR;
 }
 
-status_t AudioSystem::getOutputLatency(uint32_t* latency)
+status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
 {
-    if (gOutLatency == 0) {
+    int output = getOutput(streamType);
+
+    if (gOutLatency[output] == 0) {
         const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
         if (af == 0) return PERMISSION_DENIED;
         // gOutLatency is updated by get_audio_flinger()
-    }    
-    *latency = gOutLatency;
+    }
+    LOGV("getOutputLatency() streamType %d, output %d, latency %d", streamType, output, gOutLatency[output]);
+
+    *latency = gOutLatency[output];
     
     return NO_ERROR;
 }
@@ -315,24 +329,23 @@ status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int ch
 void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {   
     Mutex::Autolock _l(AudioSystem::gLock);
     AudioSystem::gAudioFlinger.clear();
-    AudioSystem::gOutSamplingRate = 0;
-    AudioSystem::gOutFrameCount = 0;
-    AudioSystem::gOutLatency = 0;
+
+    for (int output = 0; output < NUM_AUDIO_OUTPUT_TYPES; output++) {
+        gOutFrameCount[output] = 0;
+        gOutSamplingRate[output] = 0;
+        gOutLatency[output] = 0;
+    }
     AudioSystem::gInBuffSize = 0;
-    
+
     if (gAudioErrorCallback) {
         gAudioErrorCallback(DEAD_OBJECT);
     }
     LOGW("AudioFlinger server died!");
 }
 
-void AudioSystem::AudioFlingerClient::audioOutputChanged(uint32_t frameCount, uint32_t samplingRate, uint32_t latency) {
-
-    AudioSystem::gOutFrameCount = frameCount;
-    AudioSystem::gOutSamplingRate = samplingRate;
-    AudioSystem::gOutLatency = latency;
-
-    LOGV("AudioFlinger output changed!");
+void AudioSystem::AudioFlingerClient::a2dpEnabledChanged(bool enabled) {
+    gA2dpEnabled = enabled;        
+    LOGV("AudioFlinger A2DP enabled status changed! %d", enabled);
 }
 
 void AudioSystem::setErrorCallback(audio_error_callback cb) {
@@ -340,5 +353,31 @@ void AudioSystem::setErrorCallback(audio_error_callback cb) {
     gAudioErrorCallback = cb;
 }
 
+int AudioSystem::getOutput(int streamType)
+{  
+    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:
+    case VOICE_CALL:
+    case BLUETOOTH_SCO:
+    case SYSTEM:
+        return true;
+    default:
+        return false;
+    }
+}
+
+
+
 }; // namespace android
 
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 63b2012..1ffad46 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -128,22 +128,21 @@ status_t AudioTrack::set(
        return NO_INIT;
     }
     int afSampleRate;
-    if (AudioSystem::getOutputSamplingRate(&afSampleRate) != NO_ERROR) {
+    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
         return NO_INIT;
     }
     int afFrameCount;
-    if (AudioSystem::getOutputFrameCount(&afFrameCount) != NO_ERROR) {
+    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
         return NO_INIT;
     }
     uint32_t afLatency;
-    if (AudioSystem::getOutputLatency(&afLatency) != NO_ERROR) {
+    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
         return NO_INIT;
     }
 
-
     // handle default values first.
-    if (streamType == DEFAULT) {
-        streamType = MUSIC;
+    if (streamType == AudioSystem::DEFAULT) {
+        streamType = AudioSystem::MUSIC;
     }
     if (sampleRate == 0) {
         sampleRate = afSampleRate;
@@ -260,7 +259,7 @@ status_t AudioTrack::set(
     mMarkerPosition = 0;
     mNewPosition = 0;
     mUpdatePeriod = 0;
-    
+
     return NO_ERROR;
 }
 
@@ -434,7 +433,7 @@ void AudioTrack::setSampleRate(int rate)
 {
     int afSamplingRate;
 
-    if (AudioSystem::getOutputSamplingRate(&afSamplingRate) != NO_ERROR) {
+    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
         return;
     }
     // Resampler implementation limits input sampling rate to 2 x output sampling rate.
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 4215820..5cbb25c 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -53,7 +53,8 @@ enum {
     SET_PARAMETER,
     REGISTER_CLIENT,
     GET_INPUTBUFFERSIZE,
-    WAKE_UP
+    WAKE_UP,
+    IS_A2DP_ENABLED
 };
 
 class BpAudioFlinger : public BpInterface<IAudioFlinger>
@@ -123,42 +124,47 @@ public:
         return interface_cast<IAudioRecord>(reply.readStrongBinder());
     }
 
-    virtual uint32_t sampleRate() const
+    virtual uint32_t sampleRate(int output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(output);
         remote()->transact(SAMPLE_RATE, data, &reply);
         return reply.readInt32();
     }
 
-    virtual int channelCount() const
+    virtual int channelCount(int output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(output);
         remote()->transact(CHANNEL_COUNT, data, &reply);
         return reply.readInt32();
     }
 
-    virtual int format() const
+    virtual int format(int output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(output);
         remote()->transact(FORMAT, data, &reply);
         return reply.readInt32();
     }
 
-    virtual size_t frameCount() const
+    virtual size_t frameCount(int output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(output);
         remote()->transact(FRAME_COUNT, data, &reply);
         return reply.readInt32();
     }
 
-    virtual uint32_t latency() const
+    virtual uint32_t latency(int output) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(output);
         remote()->transact(LATENCY, data, &reply);
         return reply.readInt32();
     }
@@ -333,6 +339,14 @@ public:
         remote()->transact(WAKE_UP, data, &reply);
         return;
     }
+
+    virtual bool isA2dpEnabled() const
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        remote()->transact(IS_A2DP_ENABLED, data, &reply);
+        return (bool)reply.readInt32();
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioFlinger, "android.media.IAudioFlinger");
@@ -385,27 +399,32 @@ status_t BnAudioFlinger::onTransact(
         } break;
         case SAMPLE_RATE: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            reply->writeInt32( sampleRate() );
+            int output = data.readInt32();
+            reply->writeInt32( sampleRate(output) );
             return NO_ERROR;
         } break;
         case CHANNEL_COUNT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            reply->writeInt32( channelCount() );
+            int output = data.readInt32();
+            reply->writeInt32( channelCount(output) );
             return NO_ERROR;
         } break;
         case FORMAT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            reply->writeInt32( format() );
+            int output = data.readInt32();
+            reply->writeInt32( format(output) );
             return NO_ERROR;
         } break;
         case FRAME_COUNT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            reply->writeInt32( frameCount() );
+            int output = data.readInt32();
+            reply->writeInt32( frameCount(output) );
             return NO_ERROR;
         } break;
         case LATENCY: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            reply->writeInt32( latency() );
+            int output = data.readInt32();
+            reply->writeInt32( latency(output) );
             return NO_ERROR;
         } break;
          case SET_MASTER_VOLUME: {
@@ -519,7 +538,11 @@ status_t BnAudioFlinger::onTransact(
             wakeUp();
             return NO_ERROR;
         } break;
-
+        case IS_A2DP_ENABLED: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            reply->writeInt32( (int)isA2dpEnabled() );
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index d956266..5feb11f 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -38,13 +38,11 @@ public:
     {
     }
 
-    void audioOutputChanged(uint32_t frameCount, uint32_t samplingRate, uint32_t latency)
+    void a2dpEnabledChanged(bool enabled)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlingerClient::getInterfaceDescriptor());
-        data.writeInt32(frameCount);
-        data.writeInt32(samplingRate);
-        data.writeInt32(latency);
+        data.writeInt32((int)enabled);
         remote()->transact(AUDIO_OUTPUT_CHANGED, data, &reply);
     }
 };
@@ -65,10 +63,8 @@ status_t BnAudioFlingerClient::onTransact(
     switch(code) {
         case AUDIO_OUTPUT_CHANGED: {
             CHECK_INTERFACE(IAudioFlingerClient, data, reply);
-            uint32_t frameCount = data.readInt32();
-            uint32_t samplingRate = data.readInt32();
-            uint32_t latency = data.readInt32();
-            audioOutputChanged(frameCount, samplingRate, latency);
+            bool enabled = (bool)data.readInt32();
+            a2dpEnabledChanged(enabled);
             return NO_ERROR;
         } break;
         default:
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index 1f6d599..507d03e 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -39,7 +39,8 @@ enum {
     SET_OUTPUT_FORMAT,
     SET_VIDEO_ENCODER,
     SET_AUDIO_ENCODER,
-    SET_OUTPUT_FILE,
+    SET_OUTPUT_FILE_PATH,
+    SET_OUTPUT_FILE_FD,
     SET_VIDEO_SIZE,
     SET_VIDEO_FRAMERATE,
     SET_PREVIEW_SURFACE,
@@ -139,7 +140,18 @@ public:
         Parcel data, reply;
         data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor());
         data.writeCString(path);
-        remote()->transact(SET_OUTPUT_FILE, data, &reply);
+        remote()->transact(SET_OUTPUT_FILE_PATH, data, &reply);
+        return reply.readInt32();
+    }
+
+    status_t setOutputFile(int fd, int64_t offset, int64_t length) {
+        LOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+        Parcel data, reply;
+        data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor());
+        data.writeFileDescriptor(fd);
+        data.writeInt64(offset);
+        data.writeInt64(length);
+        remote()->transact(SET_OUTPUT_FILE_FD, data, &reply);
         return reply.readInt32();
     }
 
@@ -330,13 +342,22 @@ status_t BnMediaRecorder::onTransact(
             return NO_ERROR;
 
         } break;
-        case SET_OUTPUT_FILE: {
-            LOGV("SET_OUTPUT_FILE");
+        case SET_OUTPUT_FILE_PATH: {
+            LOGV("SET_OUTPUT_FILE_PATH");
             CHECK_INTERFACE(IMediaRecorder, data, reply);
             const char* path = data.readCString();
             reply->writeInt32(setOutputFile(path));
             return NO_ERROR;
         } break;
+        case SET_OUTPUT_FILE_FD: {
+            LOGV("SET_OUTPUT_FILE_FD");
+            CHECK_INTERFACE(IMediaRecorder, data, reply);
+            int fd = dup(data.readFileDescriptor());
+            int64_t offset = data.readInt64();
+            int64_t length = data.readInt64();
+            reply->writeInt32(setOutputFile(fd, offset, length));
+            return NO_ERROR;
+        } break;
         case SET_VIDEO_SIZE: {
             LOGV("SET_VIDEO_SIZE");
             CHECK_INTERFACE(IMediaRecorder, data, reply);
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index ead24d4..9bd75c2 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -96,7 +96,7 @@ int JetPlayer::init()
 
     // create the output AudioTrack
     mAudioTrack = new AudioTrack();
-    mAudioTrack->set(AudioTrack::MUSIC,  //TODO parametrize this
+    mAudioTrack->set(AudioSystem::MUSIC,  //TODO parametrize this
             pLibConfig->sampleRate,
             1, // format = PCM 16bits per sample,
             pLibConfig->numChannels,
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index fa36460..7fafc56 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -93,7 +93,7 @@ ToneGenerator::ToneGenerator(int streamType, float volume) {
 
     mState = TONE_IDLE;
 
-    if (AudioSystem::getOutputSamplingRate(&mSamplingRate) != NO_ERROR) {
+    if (AudioSystem::getOutputSamplingRate(&mSamplingRate, streamType) != NO_ERROR) {
         LOGE("Unable to marshal AudioFlinger");
         return;
     }
@@ -182,7 +182,7 @@ bool ToneGenerator::startTone(int toneType) {
             mLock.lock();
             if (mState == TONE_STARTING) {
                 if (mWaitCbkCond.waitRelative(mLock, seconds(1)) != NO_ERROR) {
-                    LOGE("--- timed out");
+                    LOGE("--- Immediate start timed out");
                     mState = TONE_IDLE;
                 }
             }
@@ -200,7 +200,7 @@ bool ToneGenerator::startTone(int toneType) {
             }
             LOGV("cond received");
         } else {
-            LOGE("--- timed out");
+            LOGE("--- Delayed start timed out");
             mState = TONE_IDLE;
         }
     }
@@ -235,7 +235,7 @@ void ToneGenerator::stopTone() {
         if (lStatus == NO_ERROR) {
             LOGV("track stop complete, time %d", (unsigned int)(systemTime()/1000000));
         } else {
-            LOGE("--- timed out");
+            LOGE("--- Stop timed out");
             mState = TONE_IDLE;
             mpAudioTrack->stop();
         }
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 31ff507..bd8579c 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -82,7 +82,7 @@ MediaPlayer::MediaPlayer()
     mListener = NULL;
     mCookie = NULL;
     mDuration = -1;
-    mStreamType = AudioTrack::MUSIC;
+    mStreamType = AudioSystem::MUSIC;
     mCurrentPosition = -1;
     mSeekPosition = -1;
     mCurrentState = MEDIA_PLAYER_IDLE;
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 6ee4c0d..4ab26ac 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -248,7 +248,33 @@ status_t MediaRecorder::setOutputFile(const char* path)
 
     status_t ret = mMediaRecorder->setOutputFile(path);
     if (OK != ret) {
-        LOGV("setAudioEncoder failed: %d", ret);
+        LOGV("setOutputFile failed: %d", ret);
+        mCurrentState = MEDIA_RECORDER_ERROR;
+        return UNKNOWN_ERROR;
+    }
+    mIsOutputFileSet = true;
+    return ret;
+}
+
+status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length)
+{
+    LOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+    if(mMediaRecorder == NULL) {
+        LOGE("media recorder is not initialized yet");
+        return INVALID_OPERATION;
+    }
+    if (mIsOutputFileSet) {
+        LOGE("output file has already been set");
+        return INVALID_OPERATION;
+    }
+    if (!(mCurrentState & MEDIA_RECORDER_DATASOURCE_CONFIGURED)) {
+        LOGE("setOutputFile called in an invalid state(%d)", mCurrentState);
+        return INVALID_OPERATION;
+    }
+
+    status_t ret = mMediaRecorder->setOutputFile(fd, offset, length);
+    if (OK != ret) {
+        LOGV("setOutputFile failed: %d", ret);
         mCurrentState = MEDIA_RECORDER_ERROR;
         return UNKNOWN_ERROR;
     }
@@ -306,7 +332,7 @@ status_t MediaRecorder::prepare()
         return INVALID_OPERATION;
     }
     if (!(mCurrentState & MEDIA_RECORDER_DATASOURCE_CONFIGURED)) {
-        LOGE("setVideoFrameRate called in an invalid state: %d", mCurrentState);
+        LOGE("prepare called in an invalid state: %d", mCurrentState);
         return INVALID_OPERATION;
     }
 
@@ -328,7 +354,7 @@ status_t MediaRecorder::getMaxAmplitude(int* max)
         return INVALID_OPERATION;
     }
     if (mCurrentState & MEDIA_RECORDER_ERROR) {
-        LOGE("setVideoFrameRate called in an invalid state: %d", mCurrentState);
+        LOGE("getMaxAmplitude called in an invalid state: %d", mCurrentState);
         return INVALID_OPERATION;
     }
 
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 9e366e2..97e3536 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -818,7 +818,7 @@ Exit:
 MediaPlayerService::AudioOutput::AudioOutput()
 {
     mTrack = 0;
-    mStreamType = AudioTrack::MUSIC;
+    mStreamType = AudioSystem::MUSIC;
     mLeftVolume = 1.0;
     mRightVolume = 1.0;
     mLatency = 0;
@@ -900,15 +900,15 @@ status_t MediaPlayerService::AudioOutput::open(uint32_t sampleRate, int channelC
     int afFrameCount;
     int frameCount;
 
-    if (AudioSystem::getOutputFrameCount(&afFrameCount) != NO_ERROR) {
+    if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) != NO_ERROR) {
         return NO_INIT;
     }
-    if (AudioSystem::getOutputSamplingRate(&afSampleRate) != NO_ERROR) {
+    if (AudioSystem::getOutputSamplingRate(&afSampleRate, mStreamType) != NO_ERROR) {
         return NO_INIT;
     }
 
-    frameCount = (sampleRate*afFrameCount)/afSampleRate;
-    AudioTrack *t = new AudioTrack(mStreamType, sampleRate, format, channelCount, frameCount*bufferCount);
+    frameCount = (sampleRate*afFrameCount*bufferCount)/afSampleRate;
+    AudioTrack *t = new AudioTrack(mStreamType, sampleRate, format, channelCount, frameCount);
     if ((t == 0) || (t->initCheck() != NO_ERROR)) {
         LOGE("Unable to create audio track");
         delete t;
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index f326a0e..e8ba17f 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -121,6 +121,17 @@ status_t MediaRecorderClient::setOutputFile(const char* path)
     return mRecorder->setOutputFile(path);
 }
 
+status_t MediaRecorderClient::setOutputFile(int fd, int64_t offset, int64_t length)
+{
+    LOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+    Mutex::Autolock lock(mLock);
+    if (mRecorder == NULL) {
+        LOGE("recorder is not initialized");
+        return NO_INIT;
+    }
+    return mRecorder->setOutputFile(fd, offset, length);
+}
+
 status_t MediaRecorderClient::setVideoSize(int width, int height)
 {
     LOGV("setVideoSize(%dx%d)", width, height);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.h b/media/libmediaplayerservice/MediaRecorderClient.h
index 3158017..2b80c10 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.h
+++ b/media/libmediaplayerservice/MediaRecorderClient.h
@@ -36,6 +36,7 @@ public:
     virtual     status_t        setVideoEncoder(int ve);
     virtual     status_t        setAudioEncoder(int ae);
     virtual     status_t        setOutputFile(const char* path);
+    virtual     status_t        setOutputFile(int fd, int64_t offset, int64_t length);
     virtual     status_t        setVideoSize(int width, int height);
     virtual     status_t        setVideoFrameRate(int frames_per_second);
     virtual     status_t        prepare();
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index 7ce2fab..d03caa5 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -58,7 +58,7 @@ static const S_EAS_LIB_CONFIG* pLibConfig = NULL;
 MidiFile::MidiFile() :
     mEasData(NULL), mEasHandle(NULL), mAudioBuffer(NULL),
     mPlayTime(-1), mDuration(-1), mState(EAS_STATE_ERROR),
-    mStreamType(AudioTrack::MUSIC), mLoop(false), mExit(false),
+    mStreamType(AudioSystem::MUSIC), mLoop(false), mExit(false),
     mPaused(false), mRender(false), mTid(-1)
 {
     LOGV("constructor");
diff --git a/media/libmediaplayerservice/VorbisPlayer.cpp b/media/libmediaplayerservice/VorbisPlayer.cpp
index 009d628..0ad335f 100644
--- a/media/libmediaplayerservice/VorbisPlayer.cpp
+++ b/media/libmediaplayerservice/VorbisPlayer.cpp
@@ -55,7 +55,7 @@ static status_t STATE_OPEN = 2;
 
 VorbisPlayer::VorbisPlayer() :
     mAudioBuffer(NULL), mPlayTime(-1), mDuration(-1), mState(STATE_ERROR),
-    mStreamType(AudioTrack::MUSIC), mLoop(false), mAndroidLoop(false),
+    mStreamType(AudioSystem::MUSIC), mLoop(false), mAndroidLoop(false),
     mExit(false), mPaused(false), mRender(false), mRenderTid(-1)
 {
     LOGV("constructor\n");