diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 03f8944..2dc4beb 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -392,6 +392,7 @@
     static status_t getStreamVolumeIndex(stream_type stream, int *index);
 
     static uint32_t getStrategyForStream(stream_type stream);
+    static uint32_t getDevicesForStream(stream_type stream);
 
     static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
     static status_t registerEffect(effect_descriptor_t *desc,
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index 5afceaa..720a562 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -74,6 +74,7 @@
     virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index) = 0;
     virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) = 0;
     virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream) = 0;
+    virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream) = 0;
     virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc) = 0;
     virtual status_t registerEffect(effect_descriptor_t *desc,
                                     audio_io_handle_t output,
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 9d9b3c0..2f694ba 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -668,6 +668,13 @@
     return aps->getStrategyForStream(stream);
 }
 
+uint32_t AudioSystem::getDevicesForStream(AudioSystem::stream_type stream)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return 0;
+    return aps->getDevicesForStream(stream);
+}
+
 audio_io_handle_t AudioSystem::getOutputForEffect(effect_descriptor_t *desc)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 457f7ed..b89a278 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -49,7 +49,8 @@
     GET_OUTPUT_FOR_EFFECT,
     REGISTER_EFFECT,
     UNREGISTER_EFFECT,
-    IS_STREAM_ACTIVE
+    IS_STREAM_ACTIVE,
+    GET_DEVICES_FOR_STREAM,
 };
 
 class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
@@ -263,6 +264,15 @@
         return reply.readInt32();
     }
 
+    virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(static_cast <uint32_t>(stream));
+        remote()->transact(GET_DEVICES_FOR_STREAM, data, &reply);
+        return (uint32_t) reply.readInt32();
+    }
+
     virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc)
     {
         Parcel data, reply;
@@ -495,6 +505,14 @@
             return NO_ERROR;
         } break;
 
+        case GET_DEVICES_FOR_STREAM: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            AudioSystem::stream_type stream =
+                    static_cast <AudioSystem::stream_type>(data.readInt32());
+            reply->writeInt32(static_cast <int>(getDevicesForStream(stream)));
+            return NO_ERROR;
+        } break;
+
         case GET_OUTPUT_FOR_EFFECT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             effect_descriptor_t desc;
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index 2f5a202..e368848 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -27,11 +27,6 @@
 #include "include/ThrottledSource.h"
 #include "include/MPEG2TSExtractor.h"
 
-#include "ARTPSession.h"
-#include "APacketSource.h"
-#include "ASessionDescription.h"
-#include "UDPPusher.h"
-
 #include <binder/IPCThreadState.h>
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -58,6 +53,7 @@
 
 static int64_t kLowWaterMarkUs = 2000000ll;  // 2secs
 static int64_t kHighWaterMarkUs = 10000000ll;  // 10secs
+static int64_t kHighWaterMarkRTSPUs = 4000000ll;  // 4secs
 static const size_t kLowWaterMarkBytes = 40000;
 static const size_t kHighWaterMarkBytes = 200000;
 
@@ -463,10 +459,6 @@
         mLiveSession.clear();
     }
 
-    mRTPPusher.clear();
-    mRTCPPusher.clear();
-    mRTPSession.clear();
-
     if (mVideoSource != NULL) {
         mVideoSource->stop();
 
@@ -644,6 +636,9 @@
         LOGV("cachedDurationUs = %.2f secs, eos=%d",
              cachedDurationUs / 1E6, eos);
 
+        int64_t highWaterMarkUs =
+            (mRTSPController != NULL) ? kHighWaterMarkRTSPUs : kHighWaterMarkUs;
+
         if ((mFlags & PLAYING) && !eos
                 && (cachedDurationUs < kLowWaterMarkUs)) {
             LOGI("cache is running low (%.2f secs) , pausing.",
@@ -652,7 +647,7 @@
             pause_l();
             ensureCacheIsFetching_l();
             notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
-        } else if (eos || cachedDurationUs > kHighWaterMarkUs) {
+        } else if (eos || cachedDurationUs > highWaterMarkUs) {
             if (mFlags & CACHE_UNDERRUN) {
                 LOGI("cache has filled up (%.2f secs), resuming.",
                      cachedDurationUs / 1E6);
@@ -799,25 +794,6 @@
                 mAudioPlayer = new AudioPlayer(mAudioSink, this);
                 mAudioPlayer->setSource(mAudioSource);
 
-                // We've already started the MediaSource in order to enable
-                // the prefetcher to read its data.
-                status_t err = mAudioPlayer->start(
-                        true /* sourceAlreadyStarted */);
-
-                if (err != OK) {
-                    delete mAudioPlayer;
-                    mAudioPlayer = NULL;
-
-                    mFlags &= ~(PLAYING | FIRST_FRAME);
-
-                    if (mDecryptHandle != NULL) {
-                        mDrmManagerClient->setPlaybackStatus(mDecryptHandle,
-                                 Playback::STOP, 0);
-                    }
-
-                    return err;
-                }
-
                 mTimeSource = mAudioPlayer;
 
                 deferredAudioSeek = true;
@@ -825,8 +801,26 @@
                 mWatchForAudioSeekComplete = false;
                 mWatchForAudioEOS = true;
             }
-        } else {
-            mAudioPlayer->resume();
+        }
+
+        CHECK(!(mFlags & AUDIO_RUNNING));
+
+        if (mVideoSource == NULL) {
+            status_t err = startAudioPlayer_l();
+
+            if (err != OK) {
+                delete mAudioPlayer;
+                mAudioPlayer = NULL;
+
+                mFlags &= ~(PLAYING | FIRST_FRAME);
+
+                if (mDecryptHandle != NULL) {
+                    mDrmManagerClient->setPlaybackStatus(
+                            mDecryptHandle, Playback::STOP, 0);
+                }
+
+                return err;
+            }
         }
     }
 
@@ -858,6 +852,36 @@
     return OK;
 }
 
+status_t AwesomePlayer::startAudioPlayer_l() {
+    CHECK(!(mFlags & AUDIO_RUNNING));
+
+    if (mAudioSource == NULL || mAudioPlayer == NULL) {
+        return OK;
+    }
+
+    if (!(mFlags & AUDIOPLAYER_STARTED)) {
+        mFlags |= AUDIOPLAYER_STARTED;
+
+        // We've already started the MediaSource in order to enable
+        // the prefetcher to read its data.
+        status_t err = mAudioPlayer->start(
+                true /* sourceAlreadyStarted */);
+
+        if (err != OK) {
+            notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err);
+            return err;
+        }
+    } else {
+        mAudioPlayer->resume();
+    }
+
+    mFlags |= AUDIO_RUNNING;
+
+    mWatchForAudioEOS = true;
+
+    return OK;
+}
+
 void AwesomePlayer::notifyVideoSize_l() {
     sp<MetaData> meta = mVideoSource->getFormat();
 
@@ -959,7 +983,7 @@
 
     cancelPlayerEvents(true /* keepBufferingGoing */);
 
-    if (mAudioPlayer != NULL) {
+    if (mAudioPlayer != NULL && (mFlags & AUDIO_RUNNING)) {
         if (at_eos) {
             // If we played the audio stream to completion we
             // want to make sure that all samples remaining in the audio
@@ -968,6 +992,8 @@
         } else {
             mAudioPlayer->pause();
         }
+
+        mFlags &= ~AUDIO_RUNNING;
     }
 
     mFlags &= ~PLAYING;
@@ -1200,9 +1226,7 @@
         // requested seek time instead.
 
         mAudioPlayer->seekTo(videoTimeUs < 0 ? mSeekTimeUs : videoTimeUs);
-        mAudioPlayer->resume();
         mWatchForAudioSeekComplete = true;
-        mWatchForAudioEOS = true;
     } else if (!mSeekNotificationSent) {
         // If we're playing video only, report seek complete now,
         // otherwise audio player will notify us later.
@@ -1246,8 +1270,10 @@
             // locations, we'll "pause" the audio source, causing it to
             // stop reading input data until a subsequent seek.
 
-            if (mAudioPlayer != NULL) {
+            if (mAudioPlayer != NULL && (mFlags & AUDIO_RUNNING)) {
                 mAudioPlayer->pause();
+
+                mFlags &= ~AUDIO_RUNNING;
             }
             mAudioSource->pause();
         }
@@ -1317,6 +1343,14 @@
     bool wasSeeking = mSeeking;
     finishSeekIfNecessary(timeUs);
 
+    if (mAudioPlayer != NULL && !(mFlags & (AUDIO_RUNNING | SEEK_PREVIEW))) {
+        status_t err = startAudioPlayer_l();
+        if (err != OK) {
+            LOGE("Startung the audio player failed w/ err %d", err);
+            return;
+        }
+    }
+
     TimeSource *ts = (mFlags & AUDIO_AT_EOS) ? &mSystemTimeSource : mTimeSource;
 
     if (mFlags & FIRST_FRAME) {
@@ -1331,10 +1365,8 @@
         mTimeSourceDeltaUs = realTimeUs - mediaTimeUs;
     }
 
-    if (!wasSeeking && mRTPSession == NULL) {
+    if (!wasSeeking) {
         // Let's display the first frame after seeking right away.
-        // We'll completely ignore timestamps for gtalk videochat
-        // and we'll play incoming video as fast as we get it.
 
         int64_t nowUs = ts->getRealTimeUs() - mTimeSourceDeltaUs;
 
@@ -1594,118 +1626,6 @@
             ->setLiveSession(mLiveSession);
 
         return setDataSource_l(extractor);
-    } else if (!strncmp("rtsp://gtalk/", mUri.string(), 13)) {
-        if (mLooper == NULL) {
-            mLooper = new ALooper;
-            mLooper->setName("gtalk rtp");
-            mLooper->start(
-                    false /* runOnCallingThread */,
-                    false /* canCallJava */,
-                    PRIORITY_HIGHEST);
-        }
-
-        const char *startOfCodecString = &mUri.string()[13];
-        const char *startOfSlash1 = strchr(startOfCodecString, '/');
-        if (startOfSlash1 == NULL) {
-            return BAD_VALUE;
-        }
-        const char *startOfWidthString = &startOfSlash1[1];
-        const char *startOfSlash2 = strchr(startOfWidthString, '/');
-        if (startOfSlash2 == NULL) {
-            return BAD_VALUE;
-        }
-        const char *startOfHeightString = &startOfSlash2[1];
-
-        String8 codecString(startOfCodecString, startOfSlash1 - startOfCodecString);
-        String8 widthString(startOfWidthString, startOfSlash2 - startOfWidthString);
-        String8 heightString(startOfHeightString);
-
-#if 0
-        mRTPPusher = new UDPPusher("/data/misc/rtpout.bin", 5434);
-        mLooper->registerHandler(mRTPPusher);
-
-        mRTCPPusher = new UDPPusher("/data/misc/rtcpout.bin", 5435);
-        mLooper->registerHandler(mRTCPPusher);
-#endif
-
-        mRTPSession = new ARTPSession;
-        mLooper->registerHandler(mRTPSession);
-
-#if 0
-        // My AMR SDP
-        static const char *raw =
-            "v=0\r\n"
-            "o=- 64 233572944 IN IP4 127.0.0.0\r\n"
-            "s=QuickTime\r\n"
-            "t=0 0\r\n"
-            "a=range:npt=0-315\r\n"
-            "a=isma-compliance:2,2.0,2\r\n"
-            "m=audio 5434 RTP/AVP 97\r\n"
-            "c=IN IP4 127.0.0.1\r\n"
-            "b=AS:30\r\n"
-            "a=rtpmap:97 AMR/8000/1\r\n"
-            "a=fmtp:97 octet-align\r\n";
-#elif 1
-        String8 sdp;
-        sdp.appendFormat(
-            "v=0\r\n"
-            "o=- 64 233572944 IN IP4 127.0.0.0\r\n"
-            "s=QuickTime\r\n"
-            "t=0 0\r\n"
-            "a=range:npt=0-315\r\n"
-            "a=isma-compliance:2,2.0,2\r\n"
-            "m=video 5434 RTP/AVP 97\r\n"
-            "c=IN IP4 127.0.0.1\r\n"
-            "b=AS:30\r\n"
-            "a=rtpmap:97 %s/90000\r\n"
-            "a=cliprect:0,0,%s,%s\r\n"
-            "a=framesize:97 %s-%s\r\n",
-
-            codecString.string(),
-            heightString.string(), widthString.string(),
-            widthString.string(), heightString.string()
-            );
-        const char *raw = sdp.string();
-
-#endif
-
-        sp<ASessionDescription> desc = new ASessionDescription;
-        CHECK(desc->setTo(raw, strlen(raw)));
-
-        CHECK_EQ(mRTPSession->setup(desc), (status_t)OK);
-
-        if (mRTPPusher != NULL) {
-            mRTPPusher->start();
-        }
-
-        if (mRTCPPusher != NULL) {
-            mRTCPPusher->start();
-        }
-
-        CHECK_EQ(mRTPSession->countTracks(), 1u);
-        sp<MediaSource> source = mRTPSession->trackAt(0);
-
-#if 0
-        bool eos;
-        while (((APacketSource *)source.get())
-                ->getQueuedDuration(&eos) < 5000000ll && !eos) {
-            usleep(100000ll);
-        }
-#endif
-
-        const char *mime;
-        CHECK(source->getFormat()->findCString(kKeyMIMEType, &mime));
-
-        if (!strncasecmp("video/", mime, 6)) {
-            setVideoSource(source);
-        } else {
-            CHECK(!strncasecmp("audio/", mime, 6));
-            setAudioSource(source);
-        }
-
-        mExtractorFlags = MediaExtractor::CAN_PAUSE;
-
-        return OK;
     } else if (!strncasecmp("rtsp://", mUri.string(), 7)) {
         if (mLooper == NULL) {
             mLooper = new ALooper;
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 3021359..797e5ca 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -39,8 +39,6 @@
 
 struct ALooper;
 struct ARTSPController;
-struct ARTPSession;
-struct UDPPusher;
 
 class DrmManagerClinet;
 class DecryptHandle;
@@ -123,6 +121,9 @@
         // We're triggering a single video event to display the first frame
         // after the seekpoint.
         SEEK_PREVIEW        = 4096,
+
+        AUDIO_RUNNING       = 8192,
+        AUDIOPLAYER_STARTED = 16384,
     };
 
     mutable Mutex mLock;
@@ -204,8 +205,6 @@
 
     sp<ALooper> mLooper;
     sp<ARTSPController> mRTSPController;
-    sp<ARTPSession> mRTPSession;
-    sp<UDPPusher> mRTPPusher, mRTCPPusher;
 
     sp<LiveSession> mLiveSession;
 
@@ -260,6 +259,8 @@
     void finishSeekIfNecessary(int64_t videoTimeUs);
     void ensureCacheIsFetching_l();
 
+    status_t startAudioPlayer_l();
+
     AwesomePlayer(const AwesomePlayer &);
     AwesomePlayer &operator=(const AwesomePlayer &);
 };
diff --git a/media/libstagefright/rtsp/ARTPAssembler.cpp b/media/libstagefright/rtsp/ARTPAssembler.cpp
index 9ba2b37..a897c10 100644
--- a/media/libstagefright/rtsp/ARTPAssembler.cpp
+++ b/media/libstagefright/rtsp/ARTPAssembler.cpp
@@ -65,13 +65,9 @@
 
 // static
 void ARTPAssembler::CopyTimes(const sp<ABuffer> &to, const sp<ABuffer> &from) {
-    uint64_t ntpTime;
-    CHECK(from->meta()->findInt64("ntp-time", (int64_t *)&ntpTime));
-
     uint32_t rtpTime;
     CHECK(from->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
 
-    to->meta()->setInt64("ntp-time", ntpTime);
     to->meta()->setInt32("rtp-time", rtpTime);
 
     // Copy the seq number.
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 5a1ea5c..601f569 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -169,12 +169,6 @@
             break;
         }
 
-        case kWhatFakeTimestamps:
-        {
-            onFakeTimestamps();
-            break;
-        }
-
         default:
         {
             TRESPASS();
@@ -461,12 +455,6 @@
     buffer->setInt32Data(u16at(&data[2]));
     buffer->setRange(payloadOffset, size - payloadOffset);
 
-    if ((mFlags & kFakeTimestamps) && !source->timeEstablished()) {
-        source->timeUpdate(rtpTime, 0);
-        source->timeUpdate(rtpTime + 90000, 0x100000000ll);
-        CHECK(source->timeEstablished());
-    }
-
     source->processRTPPacket(buffer);
 
     return OK;
@@ -592,9 +580,7 @@
 
     sp<ARTPSource> source = findSource(s, id);
 
-    if ((mFlags & kFakeTimestamps) == 0) {
-        source->timeUpdate(rtpTime, ntpTime);
-    }
+    source->timeUpdate(rtpTime, ntpTime);
 
     return 0;
 }
@@ -652,27 +638,5 @@
     }
 }
 
-void ARTPConnection::fakeTimestamps() {
-    (new AMessage(kWhatFakeTimestamps, id()))->post();
-}
-
-void ARTPConnection::onFakeTimestamps() {
-    List<StreamInfo>::iterator it = mStreams.begin();
-    while (it != mStreams.end()) {
-        StreamInfo &info = *it++;
-
-        for (size_t j = 0; j < info.mSources.size(); ++j) {
-            sp<ARTPSource> source = info.mSources.valueAt(j);
-
-            if (!source->timeEstablished()) {
-                source->timeUpdate(0, 0);
-                source->timeUpdate(0 + 90000, 0x100000000ll);
-
-                mFlags |= kFakeTimestamps;
-            }
-        }
-    }
-}
-
 }  // namespace android
 
diff --git a/media/libstagefright/rtsp/ARTPConnection.h b/media/libstagefright/rtsp/ARTPConnection.h
index a17b382..edbcc35 100644
--- a/media/libstagefright/rtsp/ARTPConnection.h
+++ b/media/libstagefright/rtsp/ARTPConnection.h
@@ -29,7 +29,6 @@
 
 struct ARTPConnection : public AHandler {
     enum Flags {
-        kFakeTimestamps      = 1,
         kRegularlyRequestFIR = 2,
     };
 
@@ -51,8 +50,6 @@
     static void MakePortPair(
             int *rtpSocket, int *rtcpSocket, unsigned *rtpPort);
 
-    void fakeTimestamps();
-
 protected:
     virtual ~ARTPConnection();
     virtual void onMessageReceived(const sp<AMessage> &msg);
@@ -63,7 +60,6 @@
         kWhatRemoveStream,
         kWhatPollStreams,
         kWhatInjectPacket,
-        kWhatFakeTimestamps,
     };
 
     static const int64_t kSelectTimeoutUs;
@@ -81,7 +77,6 @@
     void onPollStreams();
     void onInjectPacket(const sp<AMessage> &msg);
     void onSendReceiverReports();
-    void onFakeTimestamps();
 
     status_t receive(StreamInfo *info, bool receiveRTP);
 
diff --git a/media/libstagefright/rtsp/ARTPSession.cpp b/media/libstagefright/rtsp/ARTPSession.cpp
index 39c6619..c6bcb12 100644
--- a/media/libstagefright/rtsp/ARTPSession.cpp
+++ b/media/libstagefright/rtsp/ARTPSession.cpp
@@ -44,9 +44,7 @@
 
     mDesc = desc;
 
-    mRTPConn = new ARTPConnection(
-            ARTPConnection::kFakeTimestamps
-                | ARTPConnection::kRegularlyRequestFIR);
+    mRTPConn = new ARTPConnection(ARTPConnection::kRegularlyRequestFIR);
 
     looper()->registerHandler(mRTPConn);
 
diff --git a/media/libstagefright/rtsp/ARTPSource.cpp b/media/libstagefright/rtsp/ARTPSource.cpp
index 5aae4e7..893a387 100644
--- a/media/libstagefright/rtsp/ARTPSource.cpp
+++ b/media/libstagefright/rtsp/ARTPSource.cpp
@@ -42,12 +42,12 @@
     : mID(id),
       mHighestSeqNumber(0),
       mNumBuffersReceived(0),
-      mNumTimes(0),
       mLastNTPTime(0),
       mLastNTPTimeUpdateUs(0),
       mIssueFIRRequests(false),
       mLastFIRRequestUs(-1),
-      mNextFIRSeqNo((rand() * 256.0) / RAND_MAX) {
+      mNextFIRSeqNo((rand() * 256.0) / RAND_MAX),
+      mNotify(notify) {
     unsigned long PT;
     AString desc;
     AString params;
@@ -80,52 +80,25 @@
 }
 
 void ARTPSource::processRTPPacket(const sp<ABuffer> &buffer) {
-    if (queuePacket(buffer)
-            && mNumTimes == 2
-            && mAssembler != NULL) {
+    if (queuePacket(buffer) && mAssembler != NULL) {
         mAssembler->onPacketReceived(this);
     }
 }
 
 void ARTPSource::timeUpdate(uint32_t rtpTime, uint64_t ntpTime) {
-    LOGV("timeUpdate");
-
     mLastNTPTime = ntpTime;
     mLastNTPTimeUpdateUs = ALooper::GetNowUs();
 
-    if (mNumTimes == 2) {
-        mNTPTime[0] = mNTPTime[1];
-        mRTPTime[0] = mRTPTime[1];
-        mNumTimes = 1;
-    }
-    mNTPTime[mNumTimes] = ntpTime;
-    mRTPTime[mNumTimes++] = rtpTime;
-
-    if (timeEstablished()) {
-        for (List<sp<ABuffer> >::iterator it = mQueue.begin();
-             it != mQueue.end(); ++it) {
-            sp<AMessage> meta = (*it)->meta();
-
-            uint32_t rtpTime;
-            CHECK(meta->findInt32("rtp-time", (int32_t *)&rtpTime));
-
-            meta->setInt64("ntp-time", RTP2NTP(rtpTime));
-        }
-    }
+    sp<AMessage> notify = mNotify->dup();
+    notify->setInt32("time-update", true);
+    notify->setInt32("rtp-time", rtpTime);
+    notify->setInt64("ntp-time", ntpTime);
+    notify->post();
 }
 
 bool ARTPSource::queuePacket(const sp<ABuffer> &buffer) {
     uint32_t seqNum = (uint32_t)buffer->int32Data();
 
-    if (mNumTimes == 2) {
-        sp<AMessage> meta = buffer->meta();
-
-        uint32_t rtpTime;
-        CHECK(meta->findInt32("rtp-time", (int32_t *)&rtpTime));
-
-        meta->setInt64("ntp-time", RTP2NTP(rtpTime));
-    }
-
     if (mNumBuffersReceived++ == 0) {
         mHighestSeqNumber = seqNum;
         mQueue.push_back(buffer);
@@ -180,14 +153,6 @@
     return true;
 }
 
-uint64_t ARTPSource::RTP2NTP(uint32_t rtpTime) const {
-    CHECK_EQ(mNumTimes, 2u);
-
-    return mNTPTime[0] + (double)(mNTPTime[1] - mNTPTime[0])
-            * ((double)rtpTime - (double)mRTPTime[0])
-            / (double)(mRTPTime[1] - mRTPTime[0]);
-}
-
 void ARTPSource::byeReceived() {
     mAssembler->onByeReceived();
 }
diff --git a/media/libstagefright/rtsp/ARTPSource.h b/media/libstagefright/rtsp/ARTPSource.h
index e62c3f1..b70f94e 100644
--- a/media/libstagefright/rtsp/ARTPSource.h
+++ b/media/libstagefright/rtsp/ARTPSource.h
@@ -46,10 +46,6 @@
     void addReceiverReport(const sp<ABuffer> &buffer);
     void addFIR(const sp<ABuffer> &buffer);
 
-    bool timeEstablished() const {
-        return mNumTimes == 2;
-    }
-
 private:
     uint32_t mID;
     uint32_t mHighestSeqNumber;
@@ -58,10 +54,6 @@
     List<sp<ABuffer> > mQueue;
     sp<ARTPAssembler> mAssembler;
 
-    size_t mNumTimes;
-    uint64_t mNTPTime[2];
-    uint32_t mRTPTime[2];
-
     uint64_t mLastNTPTime;
     int64_t mLastNTPTimeUpdateUs;
 
@@ -69,7 +61,7 @@
     int64_t mLastFIRRequestUs;
     uint8_t mNextFIRSeqNo;
 
-    uint64_t RTP2NTP(uint32_t rtpTime) const;
+    sp<AMessage> mNotify;
 
     bool queuePacket(const sp<ABuffer> &buffer);
 
diff --git a/media/libstagefright/rtsp/Android.mk b/media/libstagefright/rtsp/Android.mk
index 0bbadc1..fb42de8 100644
--- a/media/libstagefright/rtsp/Android.mk
+++ b/media/libstagefright/rtsp/Android.mk
@@ -11,13 +11,11 @@
         APacketSource.cpp           \
         ARTPAssembler.cpp           \
         ARTPConnection.cpp          \
-        ARTPSession.cpp             \
         ARTPSource.cpp              \
         ARTPWriter.cpp              \
         ARTSPConnection.cpp         \
         ARTSPController.cpp         \
         ASessionDescription.cpp     \
-        UDPPusher.cpp               \
 
 LOCAL_C_INCLUDES:= \
 	$(JNI_H_INCLUDE) \
@@ -57,4 +55,4 @@
 
 LOCAL_MODULE:= rtp_test
 
-include $(BUILD_EXECUTABLE)
+# include $(BUILD_EXECUTABLE)
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 306a9c1..ba7c1b2 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -39,9 +39,9 @@
 #include <arpa/inet.h>
 #include <sys/socket.h>
 
-// If no access units are received within 3 secs, assume that the rtp
+// If no access units are received within 5 secs, assume that the rtp
 // stream has ended and signal end of stream.
-static int64_t kAccessUnitTimeoutUs = 3000000ll;
+static int64_t kAccessUnitTimeoutUs = 5000000ll;
 
 // If no access units arrive for the first 10 secs after starting the
 // stream, assume none ever will and signal EOS or switch transports.
@@ -101,7 +101,9 @@
           mSetupTracksSuccessful(false),
           mSeekPending(false),
           mFirstAccessUnit(true),
-          mFirstAccessUnitNTP(0),
+          mNTPAnchorUs(-1),
+          mMediaAnchorUs(-1),
+          mLastMediaTimeUs(0),
           mNumAccessUnitsReceived(0),
           mCheckPending(false),
           mCheckGeneration(0),
@@ -551,7 +553,8 @@
                 mSetupTracksSuccessful = false;
                 mSeekPending = false;
                 mFirstAccessUnit = true;
-                mFirstAccessUnitNTP = 0;
+                mNTPAnchorUs = -1;
+                mMediaAnchorUs = -1;
                 mNumAccessUnitsReceived = 0;
                 mReceivedFirstRTCPPacket = false;
                 mReceivedFirstRTPPacket = false;
@@ -632,6 +635,20 @@
 
             case 'accu':
             {
+                int32_t timeUpdate;
+                if (msg->findInt32("time-update", &timeUpdate) && timeUpdate) {
+                    size_t trackIndex;
+                    CHECK(msg->findSize("track-index", &trackIndex));
+
+                    uint32_t rtpTime;
+                    uint64_t ntpTime;
+                    CHECK(msg->findInt32("rtp-time", (int32_t *)&rtpTime));
+                    CHECK(msg->findInt64("ntp-time", (int64_t *)&ntpTime));
+
+                    onTimeUpdate(trackIndex, rtpTime, ntpTime);
+                    break;
+                }
+
                 int32_t first;
                 if (msg->findInt32("first-rtcp", &first)) {
                     mReceivedFirstRTCPPacket = true;
@@ -683,51 +700,11 @@
                     break;
                 }
 
-                uint64_t ntpTime;
-                CHECK(accessUnit->meta()->findInt64(
-                            "ntp-time", (int64_t *)&ntpTime));
-
-                uint32_t rtpTime;
-                CHECK(accessUnit->meta()->findInt32(
-                            "rtp-time", (int32_t *)&rtpTime));
-
                 if (track->mNewSegment) {
                     track->mNewSegment = false;
-
-                    LOGV("first segment unit ntpTime=0x%016llx rtpTime=%u seq=%d",
-                         ntpTime, rtpTime, seqNum);
                 }
 
-                if (mFirstAccessUnit) {
-                    mDoneMsg->setInt32("result", OK);
-                    mDoneMsg->post();
-                    mDoneMsg = NULL;
-
-                    mFirstAccessUnit = false;
-                    mFirstAccessUnitNTP = ntpTime;
-                }
-
-                if (ntpTime >= mFirstAccessUnitNTP) {
-                    ntpTime -= mFirstAccessUnitNTP;
-                } else {
-                    ntpTime = 0;
-                }
-
-                int64_t timeUs = (int64_t)(ntpTime * 1E6 / (1ll << 32));
-
-                accessUnit->meta()->setInt64("timeUs", timeUs);
-
-#if 0
-                int32_t damaged;
-                if (accessUnit->meta()->findInt32("damaged", &damaged)
-                        && damaged != 0) {
-                    LOGI("ignoring damaged AU");
-                } else
-#endif
-                {
-                    TrackInfo *track = &mTracks.editItemAt(trackIndex);
-                    track->mPacketSource->queueAccessUnit(accessUnit);
-                }
+                onAccessUnitComplete(trackIndex, accessUnit);
                 break;
             }
 
@@ -778,9 +755,15 @@
             {
                 // Session is paused now.
                 for (size_t i = 0; i < mTracks.size(); ++i) {
-                    mTracks.editItemAt(i).mPacketSource->flushQueue();
+                    TrackInfo *info = &mTracks.editItemAt(i);
+
+                    info->mPacketSource->flushQueue();
+                    info->mRTPAnchor = 0;
+                    info->mNTPAnchorUs = -1;
                 }
 
+                mNTPAnchorUs = -1;
+
                 int64_t timeUs;
                 CHECK(msg->findInt64("time", &timeUs));
 
@@ -831,6 +814,11 @@
                     } else {
                         parsePlayResponse(response);
 
+                        ssize_t i = response->mHeaders.indexOfKey("rtp-info");
+                        CHECK_GE(i, 0);
+
+                        LOGV("rtp-info: %s", response->mHeaders.valueAt(i).c_str());
+
                         LOGI("seek completed.");
                     }
                 }
@@ -875,7 +863,6 @@
                         mTryFakeRTCP = true;
 
                         mReceivedFirstRTCPPacket = true;
-                        mRTPConn->fakeTimestamps();
                     } else {
                         LOGW("Never received any data, switching transports.");
 
@@ -980,7 +967,7 @@
 
             uint32_t rtpTime = strtoul(val.c_str(), &end, 10);
 
-            LOGV("track #%d: rtpTime=%u <=> ntp=%.2f", n, rtpTime, npt1);
+            LOGV("track #%d: rtpTime=%u <=> npt=%.2f", n, rtpTime, npt1);
 
             info->mPacketSource->setNormalPlayTimeMapping(
                     rtpTime, (int64_t)(npt1 * 1E6));
@@ -1003,6 +990,25 @@
     }
 
 private:
+    struct TrackInfo {
+        AString mURL;
+        int mRTPSocket;
+        int mRTCPSocket;
+        bool mUsingInterleavedTCP;
+        uint32_t mFirstSeqNumInSegment;
+        bool mNewSegment;
+
+        uint32_t mRTPAnchor;
+        int64_t mNTPAnchorUs;
+        int32_t mTimeScale;
+
+        sp<APacketSource> mPacketSource;
+
+        // Stores packets temporarily while no notion of time
+        // has been established yet.
+        List<sp<ABuffer> > mPackets;
+    };
+
     sp<ALooper> mLooper;
     sp<ALooper> mNetLooper;
     sp<ARTSPConnection> mConn;
@@ -1015,7 +1021,11 @@
     bool mSetupTracksSuccessful;
     bool mSeekPending;
     bool mFirstAccessUnit;
-    uint64_t mFirstAccessUnitNTP;
+
+    int64_t mNTPAnchorUs;
+    int64_t mMediaAnchorUs;
+    int64_t mLastMediaTimeUs;
+
     int64_t mNumAccessUnitsReceived;
     bool mCheckPending;
     int32_t mCheckGeneration;
@@ -1025,16 +1035,6 @@
     bool mReceivedFirstRTPPacket;
     bool mSeekable;
 
-    struct TrackInfo {
-        AString mURL;
-        int mRTPSocket;
-        int mRTCPSocket;
-        bool mUsingInterleavedTCP;
-        uint32_t mFirstSeqNumInSegment;
-        bool mNewSegment;
-
-        sp<APacketSource> mPacketSource;
-    };
     Vector<TrackInfo> mTracks;
 
     sp<AMessage> mDoneMsg;
@@ -1066,6 +1066,20 @@
         info->mUsingInterleavedTCP = false;
         info->mFirstSeqNumInSegment = 0;
         info->mNewSegment = true;
+        info->mRTPAnchor = 0;
+        info->mNTPAnchorUs = -1;
+
+        unsigned long PT;
+        AString formatDesc;
+        AString formatParams;
+        mSessionDesc->getFormatType(index, &PT, &formatDesc, &formatParams);
+
+        int32_t timescale;
+        int32_t numChannels;
+        ASessionDescription::ParseFormatDesc(
+                formatDesc.c_str(), &timescale, &numChannels);
+
+        info->mTimeScale = timescale;
 
         LOGV("track #%d URL=%s", mTracks.size(), trackURL.c_str());
 
@@ -1144,6 +1158,90 @@
         return true;
     }
 
+    void onTimeUpdate(int32_t trackIndex, uint32_t rtpTime, uint64_t ntpTime) {
+        LOGV("onTimeUpdate track %d, rtpTime = 0x%08x, ntpTime = 0x%016llx",
+             trackIndex, rtpTime, ntpTime);
+
+        int64_t ntpTimeUs = (int64_t)(ntpTime * 1E6 / (1ll << 32));
+
+        TrackInfo *track = &mTracks.editItemAt(trackIndex);
+
+        track->mRTPAnchor = rtpTime;
+        track->mNTPAnchorUs = ntpTimeUs;
+
+        if (mNTPAnchorUs < 0) {
+            mNTPAnchorUs = ntpTimeUs;
+            mMediaAnchorUs = mLastMediaTimeUs;
+        }
+    }
+
+    void onAccessUnitComplete(
+            int32_t trackIndex, const sp<ABuffer> &accessUnit) {
+        LOGV("onAccessUnitComplete track %d", trackIndex);
+
+        if (mFirstAccessUnit) {
+            mDoneMsg->setInt32("result", OK);
+            mDoneMsg->post();
+            mDoneMsg = NULL;
+
+            mFirstAccessUnit = false;
+        }
+
+        TrackInfo *track = &mTracks.editItemAt(trackIndex);
+
+        if (mNTPAnchorUs < 0 || mMediaAnchorUs < 0 || track->mNTPAnchorUs < 0) {
+            LOGV("storing accessUnit, no time established yet");
+            track->mPackets.push_back(accessUnit);
+            return;
+        }
+
+        while (!track->mPackets.empty()) {
+            sp<ABuffer> accessUnit = *track->mPackets.begin();
+            track->mPackets.erase(track->mPackets.begin());
+
+            if (addMediaTimestamp(trackIndex, track, accessUnit)) {
+                track->mPacketSource->queueAccessUnit(accessUnit);
+            }
+        }
+
+        if (addMediaTimestamp(trackIndex, track, accessUnit)) {
+            track->mPacketSource->queueAccessUnit(accessUnit);
+        }
+    }
+
+    bool addMediaTimestamp(
+            int32_t trackIndex, const TrackInfo *track,
+            const sp<ABuffer> &accessUnit) {
+        uint32_t rtpTime;
+        CHECK(accessUnit->meta()->findInt32(
+                    "rtp-time", (int32_t *)&rtpTime));
+
+        int64_t relRtpTimeUs =
+            (((int64_t)rtpTime - (int64_t)track->mRTPAnchor) * 1000000ll)
+                / track->mTimeScale;
+
+        int64_t ntpTimeUs = track->mNTPAnchorUs + relRtpTimeUs;
+
+        int64_t mediaTimeUs = mMediaAnchorUs + ntpTimeUs - mNTPAnchorUs;
+
+        if (mediaTimeUs > mLastMediaTimeUs) {
+            mLastMediaTimeUs = mediaTimeUs;
+        }
+
+        if (mediaTimeUs < 0) {
+            LOGV("dropping early accessUnit.");
+            return false;
+        }
+
+        LOGV("track %d rtpTime=%d mediaTimeUs = %lld us (%.2f secs)",
+             trackIndex, rtpTime, mediaTimeUs, mediaTimeUs / 1E6);
+
+        accessUnit->meta()->setInt64("timeUs", mediaTimeUs);
+
+        return true;
+    }
+
+
     DISALLOW_EVIL_CONSTRUCTORS(MyHandler);
 };
 
diff --git a/services/audioflinger/AudioPolicyManagerBase.cpp b/services/audioflinger/AudioPolicyManagerBase.cpp
index a9e8bc0..6a82a59 100644
--- a/services/audioflinger/AudioPolicyManagerBase.cpp
+++ b/services/audioflinger/AudioPolicyManagerBase.cpp
@@ -1543,6 +1543,20 @@
     return (uint32_t)getStrategy(stream);
 }
 
+uint32_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
+    uint32_t devices;
+    // By checking the range of stream before calling getStrategy, we avoid
+    // getStrategy's behavior for invalid streams.  getStrategy would do a LOGE
+    // and then return STRATEGY_MEDIA, but we want to return the empty set.
+    if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
+        devices = 0;
+    } else {
+        AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
+        devices = getDeviceForStrategy(strategy, true);
+    }
+    return devices;
+}
+
 AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(
         AudioSystem::stream_type stream) {
     // stream to strategy mapping
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 953ddac..b614c48 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -365,6 +365,14 @@
     return mpPolicyManager->getStrategyForStream(stream);
 }
 
+uint32_t AudioPolicyService::getDevicesForStream(AudioSystem::stream_type stream)
+{
+    if (mpPolicyManager == NULL) {
+        return 0;
+    }
+    return mpPolicyManager->getDevicesForStream(stream);
+}
+
 audio_io_handle_t AudioPolicyService::getOutputForEffect(effect_descriptor_t *desc)
 {
     if (mpPolicyManager == NULL) {
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 4749b8b..faad893 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -86,6 +86,7 @@
     virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index);
 
     virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream);
+    virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream);
 
     virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
     virtual status_t registerEffect(effect_descriptor_t *desc,