diff --git a/media/libmediaplayerservice/nuplayer/Android.bp b/media/libmediaplayerservice/nuplayer/Android.bp
index 32c97cf..33621be 100644
--- a/media/libmediaplayerservice/nuplayer/Android.bp
+++ b/media/libmediaplayerservice/nuplayer/Android.bp
@@ -14,6 +14,7 @@
         "NuPlayerRenderer.cpp",
         "NuPlayerStreamListener.cpp",
         "RTSPSource.cpp",
+        "RTPSource.cpp",
         "StreamingSource.cpp",
     ],
 
diff --git a/media/libmediaplayerservice/nuplayer/RTPSource.cpp b/media/libmediaplayerservice/nuplayer/RTPSource.cpp
new file mode 100644
index 0000000..de1f8a1
--- /dev/null
+++ b/media/libmediaplayerservice/nuplayer/RTPSource.cpp
@@ -0,0 +1,708 @@
+/*
+ * Copyright (C) 2010 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_NDEBUG 0
+#define LOG_TAG "RTPSource"
+#include <utils/Log.h>
+
+#include "RTPSource.h"
+
+
+
+
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <string.h>
+
+namespace android {
+
+const int64_t kNearEOSTimeoutUs = 2000000ll; // 2 secs
+static int32_t kMaxAllowedStaleAccessUnits = 20;
+
+NuPlayer::RTPSource::RTPSource(
+        const sp<AMessage> &notify,
+        const String8& rtpParams)
+    : Source(notify),
+      mRTPParams(rtpParams),
+      mFlags(0),
+      mState(DISCONNECTED),
+      mFinalResult(OK),
+      mBuffering(false),
+      mInPreparationPhase(true),
+      mRTPConn(new ARTPConnection),
+      mEOSTimeoutAudio(0),
+      mEOSTimeoutVideo(0) {
+      ALOGD("RTPSource initialized with rtpParams=%s", rtpParams.string());
+}
+
+NuPlayer::RTPSource::~RTPSource() {
+    if (mLooper != NULL) {
+        mLooper->unregisterHandler(id());
+        mLooper->unregisterHandler(mRTPConn->id());
+        mLooper->stop();
+    }
+}
+
+status_t NuPlayer::RTPSource::getBufferingSettings(
+            BufferingSettings* buffering /* nonnull */) {
+    Mutex::Autolock _l(mBufferingSettingsLock);
+    *buffering = mBufferingSettings;
+    return OK;
+}
+
+status_t NuPlayer::RTPSource::setBufferingSettings(const BufferingSettings& buffering) {
+    Mutex::Autolock _l(mBufferingSettingsLock);
+    mBufferingSettings = buffering;
+    return OK;
+}
+
+void NuPlayer::RTPSource::prepareAsync() {
+    if (mLooper == NULL) {
+        mLooper = new ALooper;
+        mLooper->setName("rtp");
+        mLooper->start();
+
+        mLooper->registerHandler(this);
+        mLooper->registerHandler(mRTPConn);
+    }
+
+    setParameters(mRTPParams);
+
+    TrackInfo *info = NULL;
+    unsigned i;
+    for (i = 0; i < mTracks.size(); i++) {
+        info = &mTracks.editItemAt(i);
+
+        if (info == NULL)
+            break;
+
+        AString sdp;
+        ASessionDescription::SDPStringFactory(sdp, info->mLocalIp,
+                info->mIsAudio, info->mLocalPort, info->mPayloadType, info->mAS, info->mCodecName,
+                NULL, info->mWidth, info->mHeight);
+        ALOGD("RTPSource SDP =>\n%s", sdp.c_str());
+
+        sp<ASessionDescription> desc = new ASessionDescription;
+        bool isValidSdp = desc->setTo(sdp.c_str(), sdp.size());
+        ALOGV("RTPSource isValidSdp => %d", isValidSdp);
+
+        int sockRtp, sockRtcp;
+        ARTPConnection::MakeRTPSocketPair(&sockRtp, &sockRtcp, info->mLocalIp, info->mRemoteIp,
+                info->mLocalPort, info->mRemotePort);
+
+        sp<AMessage> notify = new AMessage('accu', this);
+
+        ALOGV("RTPSource addStream. track-index=%d", i);
+        notify->setSize("trackIndex", i);
+        // index(i) should be started from 1. 0 is reserved for [root]
+        mRTPConn->addStream(sockRtp, sockRtcp, desc, i + 1, notify, false);
+
+        info->mRTPSocket = sockRtp;
+        info->mRTCPSocket = sockRtcp;
+        info->mFirstSeqNumInSegment = 0;
+        info->mNewSegment = true;
+        info->mAllowedStaleAccessUnits = kMaxAllowedStaleAccessUnits;
+        info->mRTPAnchor = 0;
+        info->mNTPAnchorUs = -1;
+        info->mNormalPlayTimeRTP = 0;
+        info->mNormalPlayTimeUs = 0ll;
+
+        // index(i) should be started from 1. 0 is reserved for [root]
+        info->mPacketSource = new APacketSource(desc, i + 1);
+
+        int32_t timeScale;
+        sp<MetaData> format = getTrackFormat(i, &timeScale);
+        sp<AnotherPacketSource> source = new AnotherPacketSource(format);
+
+        if (info->mIsAudio) {
+            mAudioTrack = source;
+        } else {
+            mVideoTrack = source;
+        }
+
+        info->mSource = source;
+    }
+
+    CHECK_EQ(mState, (int)DISCONNECTED);
+    mState = CONNECTING;
+
+    if (mInPreparationPhase) {
+        mInPreparationPhase = false;
+        notifyPrepared();
+    }
+}
+
+void NuPlayer::RTPSource::start() {
+}
+
+void NuPlayer::RTPSource::pause() {
+    mState = PAUSED;
+}
+
+void NuPlayer::RTPSource::resume() {
+    mState = CONNECTING;
+}
+
+void NuPlayer::RTPSource::stop() {
+    if (mLooper == NULL) {
+        return;
+    }
+    sp<AMessage> msg = new AMessage(kWhatDisconnect, this);
+
+    sp<AMessage> dummy;
+    msg->postAndAwaitResponse(&dummy);
+}
+
+status_t NuPlayer::RTPSource::feedMoreTSData() {
+    Mutex::Autolock _l(mBufferingLock);
+    return mFinalResult;
+}
+
+sp<MetaData> NuPlayer::RTPSource::getFormatMeta(bool audio) {
+    sp<AnotherPacketSource> source = getSource(audio);
+
+    if (source == NULL) {
+        return NULL;
+    }
+
+    return source->getFormat();
+}
+
+bool NuPlayer::RTPSource::haveSufficientDataOnAllTracks() {
+    // We're going to buffer at least 2 secs worth data on all tracks before
+    // starting playback (both at startup and after a seek).
+
+    static const int64_t kMinDurationUs = 2000000ll;
+
+    int64_t mediaDurationUs = 0;
+    getDuration(&mediaDurationUs);
+    if ((mAudioTrack != NULL && mAudioTrack->isFinished(mediaDurationUs))
+            || (mVideoTrack != NULL && mVideoTrack->isFinished(mediaDurationUs))) {
+        return true;
+    }
+
+    status_t err;
+    int64_t durationUs;
+    if (mAudioTrack != NULL
+            && (durationUs = mAudioTrack->getBufferedDurationUs(&err))
+                    < kMinDurationUs
+            && err == OK) {
+        ALOGV("audio track doesn't have enough data yet. (%.2f secs buffered)",
+              durationUs / 1E6);
+        return false;
+    }
+
+    if (mVideoTrack != NULL
+            && (durationUs = mVideoTrack->getBufferedDurationUs(&err))
+                    < kMinDurationUs
+            && err == OK) {
+        ALOGV("video track doesn't have enough data yet. (%.2f secs buffered)",
+              durationUs / 1E6);
+        return false;
+    }
+
+    return true;
+}
+
+status_t NuPlayer::RTPSource::dequeueAccessUnit(
+        bool audio, sp<ABuffer> *accessUnit) {
+
+    sp<AnotherPacketSource> source = getSource(audio);
+
+    if (mState == PAUSED) {
+        ALOGV("-EWOULDBLOCK");
+        return -EWOULDBLOCK;
+    }
+
+    status_t finalResult;
+    if (!source->hasBufferAvailable(&finalResult)) {
+        if (finalResult == OK) {
+            int64_t mediaDurationUs = 0;
+            getDuration(&mediaDurationUs);
+            sp<AnotherPacketSource> otherSource = getSource(!audio);
+            status_t otherFinalResult;
+
+            // If other source already signaled EOS, this source should also signal EOS
+            if (otherSource != NULL &&
+                    !otherSource->hasBufferAvailable(&otherFinalResult) &&
+                    otherFinalResult == ERROR_END_OF_STREAM) {
+                source->signalEOS(ERROR_END_OF_STREAM);
+                return ERROR_END_OF_STREAM;
+            }
+
+            // If this source has detected near end, give it some time to retrieve more
+            // data before signaling EOS
+            if (source->isFinished(mediaDurationUs)) {
+                int64_t eosTimeout = audio ? mEOSTimeoutAudio : mEOSTimeoutVideo;
+                if (eosTimeout == 0) {
+                    setEOSTimeout(audio, ALooper::GetNowUs());
+                } else if ((ALooper::GetNowUs() - eosTimeout) > kNearEOSTimeoutUs) {
+                    setEOSTimeout(audio, 0);
+                    source->signalEOS(ERROR_END_OF_STREAM);
+                    return ERROR_END_OF_STREAM;
+                }
+                return -EWOULDBLOCK;
+            }
+
+            if (!(otherSource != NULL && otherSource->isFinished(mediaDurationUs))) {
+                // We should not enter buffering mode
+                // if any of the sources already have detected EOS.
+                // TODO: needs to be checked whether below line is needed or not.
+                // startBufferingIfNecessary();
+            }
+
+            return -EWOULDBLOCK;
+        }
+        return finalResult;
+    }
+
+    setEOSTimeout(audio, 0);
+
+    return source->dequeueAccessUnit(accessUnit);
+}
+
+sp<AnotherPacketSource> NuPlayer::RTPSource::getSource(bool audio) {
+    return audio ? mAudioTrack : mVideoTrack;
+}
+
+void NuPlayer::RTPSource::setEOSTimeout(bool audio, int64_t timeout) {
+    if (audio) {
+        mEOSTimeoutAudio = timeout;
+    } else {
+        mEOSTimeoutVideo = timeout;
+    }
+}
+
+status_t NuPlayer::RTPSource::getDuration(int64_t *durationUs) {
+    *durationUs = 0ll;
+
+    int64_t audioDurationUs;
+    if (mAudioTrack != NULL
+            && mAudioTrack->getFormat()->findInt64(
+                kKeyDuration, &audioDurationUs)
+            && audioDurationUs > *durationUs) {
+        *durationUs = audioDurationUs;
+    }
+
+    int64_t videoDurationUs;
+    if (mVideoTrack != NULL
+            && mVideoTrack->getFormat()->findInt64(
+                kKeyDuration, &videoDurationUs)
+            && videoDurationUs > *durationUs) {
+        *durationUs = videoDurationUs;
+    }
+
+    return OK;
+}
+
+status_t NuPlayer::RTPSource::seekTo(int64_t seekTimeUs, MediaPlayerSeekMode mode) {
+    ALOGV("RTPSource::seekTo=%d, mode=%d", (int)seekTimeUs, mode);
+    return OK;
+}
+
+void NuPlayer::RTPSource::schedulePollBuffering() {
+    sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
+    msg->post(1000000ll); // 1 second intervals
+}
+
+void NuPlayer::RTPSource::onPollBuffering() {
+    schedulePollBuffering();
+}
+
+void NuPlayer::RTPSource::onMessageReceived(const sp<AMessage> &msg) {
+    ALOGV("onMessageReceived =%d", msg->what());
+
+    switch (msg->what()) {
+        case kWhatAccessUnitComplete:
+        {
+            if (mState == CONNECTING) {
+                mState = CONNECTED;
+            }
+
+            int32_t timeUpdate;
+            //"time-update" raised from ARTPConnection::parseSR()
+            if (msg->findInt32("time-update", &timeUpdate) && timeUpdate) {
+                size_t trackIndex;
+                CHECK(msg->findSize("trackIndex", &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 firstRTCP;
+            if (msg->findInt32("first-rtcp", &firstRTCP)) {
+                // There won't be an access unit here, it's just a notification
+                // that the data communication worked since we got the first
+                // rtcp packet.
+                ALOGV("first-rtcp");
+                break;
+            }
+
+            size_t trackIndex;
+            CHECK(msg->findSize("trackIndex", &trackIndex));
+
+            sp<ABuffer> accessUnit;
+            if (msg->findBuffer("access-unit", &accessUnit) == false) {
+                break;
+            }
+
+            int32_t damaged;
+            if (accessUnit->meta()->findInt32("damaged", &damaged)
+                    && damaged) {
+                ALOGD("dropping damaged access unit.");
+                break;
+            }
+
+            TrackInfo *info = &mTracks.editItemAt(trackIndex);
+
+            sp<AnotherPacketSource> source = info->mSource;
+            if (source != NULL) {
+                uint32_t rtpTime;
+                CHECK(accessUnit->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+
+                /* AnotherPacketSource make an assertion if there is no ntp provided
+                   RTPSource should provide ntpUs all the times.
+                if (!info->mNPTMappingValid) {
+                    // This is a live stream, we didn't receive any normal
+                    // playtime mapping. We won't map to npt time.
+                    source->queueAccessUnit(accessUnit);
+                    break;
+                }
+                */
+
+                int64_t nptUs =
+                    ((double)rtpTime - (double)info->mRTPTime)
+                        / info->mTimeScale
+                        * 1000000ll
+                        + info->mNormalPlaytimeUs;
+
+                accessUnit->meta()->setInt64("timeUs", nptUs);
+
+                source->queueAccessUnit(accessUnit);
+            }
+
+            break;
+        }
+        case kWhatDisconnect:
+        {
+            sp<AReplyToken> replyID;
+            CHECK(msg->senderAwaitsResponse(&replyID));
+
+            for (size_t i = 0; i < mTracks.size(); ++i) {
+                TrackInfo *info = &mTracks.editItemAt(i);
+
+                if (info->mIsAudio) {
+                    mAudioTrack->signalEOS(ERROR_END_OF_STREAM);
+                    mAudioTrack = NULL;
+                    ALOGV("mAudioTrack disconnected");
+                } else {
+                    mVideoTrack->signalEOS(ERROR_END_OF_STREAM);
+                    mVideoTrack = NULL;
+                    ALOGV("mVideoTrack disconnected");
+                }
+
+                mRTPConn->removeStream(info->mRTPSocket, info->mRTCPSocket);
+                close(info->mRTPSocket);
+                close(info->mRTCPSocket);
+            }
+
+            mTracks.clear();
+            mFirstAccessUnit = true;
+            mAllTracksHaveTime = false;
+            mNTPAnchorUs = -1;
+            mMediaAnchorUs = -1;
+            mLastMediaTimeUs = -1;
+            mNumAccessUnitsReceived = 0;
+            mReceivedFirstRTCPPacket = false;
+            mReceivedFirstRTPPacket = false;
+            mPausing = false;
+            mPauseGeneration = 0;
+
+            (new AMessage)->postReply(replyID);
+
+            break;
+        }
+        case kWhatPollBuffering:
+            break;
+        default:
+            TRESPASS();
+    }
+}
+
+void NuPlayer::RTPSource::onTimeUpdate(int32_t trackIndex, uint32_t rtpTime, uint64_t ntpTime) {
+    ALOGV("onTimeUpdate track %d, rtpTime = 0x%08x, ntpTime = %#016llx",
+         trackIndex, rtpTime, (long long)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;
+    }
+
+    if (!mAllTracksHaveTime) {
+        bool allTracksHaveTime = (mTracks.size() > 0);
+        for (size_t i = 0; i < mTracks.size(); ++i) {
+            TrackInfo *track = &mTracks.editItemAt(i);
+            if (track->mNTPAnchorUs < 0) {
+                allTracksHaveTime = false;
+                break;
+            }
+        }
+        if (allTracksHaveTime) {
+            mAllTracksHaveTime = true;
+            ALOGI("Time now established for all tracks.");
+        }
+    }
+    if (mAllTracksHaveTime && dataReceivedOnAllChannels()) {
+        // Time is now established, lets start timestamping immediately
+        for (size_t i = 0; i < mTracks.size(); ++i) {
+            TrackInfo *trackInfo = &mTracks.editItemAt(i);
+            while (!trackInfo->mPackets.empty()) {
+                sp<ABuffer> accessUnit = *trackInfo->mPackets.begin();
+                trackInfo->mPackets.erase(trackInfo->mPackets.begin());
+
+                if (addMediaTimestamp(i, trackInfo, accessUnit)) {
+                    postQueueAccessUnit(i, accessUnit);
+                }
+            }
+        }
+    }
+}
+
+bool NuPlayer::RTPSource::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) {
+        ALOGV("dropping early accessUnit.");
+        return false;
+    }
+
+    ALOGV("track %d rtpTime=%u mediaTimeUs = %lld us (%.2f secs)",
+            trackIndex, rtpTime, (long long)mediaTimeUs, mediaTimeUs / 1E6);
+
+    accessUnit->meta()->setInt64("timeUs", mediaTimeUs);
+
+    return true;
+}
+
+bool NuPlayer::RTPSource::dataReceivedOnAllChannels() {
+    TrackInfo *track;
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        track = &mTracks.editItemAt(i);
+        if (track->mPackets.empty()) {
+            return false;
+        }
+    }
+    return true;
+}
+
+void NuPlayer::RTPSource::postQueueAccessUnit(
+        size_t trackIndex, const sp<ABuffer> &accessUnit) {
+    sp<AMessage> msg = new AMessage(kWhatAccessUnit, this);
+    msg->setInt32("what", kWhatAccessUnit);
+    msg->setSize("trackIndex", trackIndex);
+    msg->setBuffer("accessUnit", accessUnit);
+    msg->post();
+}
+
+void NuPlayer::RTPSource::postQueueEOS(size_t trackIndex, status_t finalResult) {
+    sp<AMessage> msg = new AMessage(kWhatEOS, this);
+    msg->setInt32("what", kWhatEOS);
+    msg->setSize("trackIndex", trackIndex);
+    msg->setInt32("finalResult", finalResult);
+    msg->post();
+}
+
+sp<MetaData> NuPlayer::RTPSource::getTrackFormat(size_t index, int32_t *timeScale) {
+    CHECK_GE(index, 0u);
+    CHECK_LT(index, mTracks.size());
+
+    const TrackInfo &info = mTracks.itemAt(index);
+
+    *timeScale = info.mTimeScale;
+
+    return info.mPacketSource->getFormat();
+}
+
+void NuPlayer::RTPSource::onConnected() {
+    ALOGV("onConnected");
+    mState = CONNECTED;
+}
+
+void NuPlayer::RTPSource::onDisconnected(const sp<AMessage> &msg) {
+    if (mState == DISCONNECTED) {
+        return;
+    }
+
+    status_t err;
+    CHECK(msg->findInt32("result", &err));
+    CHECK_NE(err, (status_t)OK);
+
+//    mLooper->unregisterHandler(mHandler->id());
+//    mHandler.clear();
+
+    if (mState == CONNECTING) {
+        // We're still in the preparation phase, signal that it
+        // failed.
+        notifyPrepared(err);
+    }
+
+    mState = DISCONNECTED;
+//    setError(err);
+
+}
+
+status_t NuPlayer::RTPSource::setParameter(const String8 &key, const String8 &value) {
+    ALOGV("setParameter: key (%s) => value (%s)", key.string(), value.string());
+
+    bool isAudioKey = key.contains("audio");
+    TrackInfo *info = NULL;
+    for (unsigned i = 0; i < mTracks.size(); ++i) {
+        info = &mTracks.editItemAt(i);
+        if (info != NULL && info->mIsAudio == isAudioKey) {
+            ALOGV("setParameter: %s track (%d) found", isAudioKey ? "audio" : "video" , i);
+            break;
+        }
+    }
+
+    if (info == NULL) {
+        TrackInfo newTrackInfo;
+        newTrackInfo.mIsAudio = isAudioKey;
+        mTracks.push(newTrackInfo);
+        info = &mTracks.editTop();
+    }
+
+    if (key == "rtp-param-mime-type") {
+        info->mMimeType = value;
+
+        const char *mime = value.string();
+        const char *delimiter = strchr(mime, '/');
+        info->mCodecName = (delimiter + 1);
+
+        ALOGV("rtp-param-mime-type: mMimeType (%s) => mCodecName (%s)",
+            info->mMimeType.string(), info->mCodecName.string());
+    } else if (key == "video-param-decoder-profile") {
+        info->mCodecProfile = atoi(value);
+    } else if (key == "video-param-decoder-level") {
+        info->mCodecLevel = atoi(value);
+    } else if (key == "video-param-width") {
+        info->mWidth = atoi(value);
+    } else if (key == "video-param-height") {
+        info->mHeight = atoi(value);
+    } else if (key == "rtp-param-local-ip") {
+        info->mLocalIp = value;
+    } else if (key == "rtp-param-local-port") {
+        info->mLocalPort = atoi(value);
+    } else if (key == "rtp-param-remote-ip") {
+        info->mRemoteIp = value;
+    } else if (key == "rtp-param-remote-port") {
+        info->mRemotePort = atoi(value);
+    } else if (key == "rtp-param-payload-type") {
+        info->mPayloadType = atoi(value);
+    } else if (key == "rtp-param-as") {
+        //AS means guaranteed bit rate that negotiated from sdp.
+        info->mAS = atoi(value);
+    } else if (key == "rtp-param-rtp-timeout") {
+    } else if (key == "rtp-param-rtcp-timeout") {
+    } else if (key == "rtp-param-time-scale") {
+    }
+
+    return OK;
+}
+
+status_t NuPlayer::RTPSource::setParameters(const String8 &params) {
+    ALOGV("setParameters: %s", params.string());
+    const char *cparams = params.string();
+    const char *key_start = cparams;
+    for (;;) {
+        const char *equal_pos = strchr(key_start, '=');
+        if (equal_pos == NULL) {
+            ALOGE("Parameters %s miss a value", cparams);
+            return BAD_VALUE;
+        }
+        String8 key(key_start, equal_pos - key_start);
+        TrimString(&key);
+        if (key.length() == 0) {
+            ALOGE("Parameters %s contains an empty key", cparams);
+            return BAD_VALUE;
+        }
+        const char *value_start = equal_pos + 1;
+        const char *semicolon_pos = strchr(value_start, ';');
+        String8 value;
+        if (semicolon_pos == NULL) {
+            value.setTo(value_start);
+        } else {
+            value.setTo(value_start, semicolon_pos - value_start);
+        }
+        if (setParameter(key, value) != OK) {
+            return BAD_VALUE;
+        }
+        if (semicolon_pos == NULL) {
+            break;  // Reaches the end
+        }
+        key_start = semicolon_pos + 1;
+    }
+    return OK;
+}
+
+// Trim both leading and trailing whitespace from the given string.
+//static
+void NuPlayer::RTPSource::TrimString(String8 *s) {
+    size_t num_bytes = s->bytes();
+    const char *data = s->string();
+
+    size_t leading_space = 0;
+    while (leading_space < num_bytes && isspace(data[leading_space])) {
+        ++leading_space;
+    }
+
+    size_t i = num_bytes;
+    while (i > leading_space && isspace(data[i - 1])) {
+        --i;
+    }
+
+    s->setTo(String8(&data[leading_space], i - leading_space));
+}
+
+}  // namespace android
diff --git a/media/libmediaplayerservice/nuplayer/RTPSource.h b/media/libmediaplayerservice/nuplayer/RTPSource.h
new file mode 100644
index 0000000..8ee6598
--- /dev/null
+++ b/media/libmediaplayerservice/nuplayer/RTPSource.h
@@ -0,0 +1,200 @@
+/*
+ * Copyright (C) 2010 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 RTP_SOURCE_H_
+
+#define RTP_SOURCE_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/Utils.h>
+#include <media/BufferingSettings.h>
+#include <media/MediaSource.h>
+
+#include <utils/KeyedVector.h>
+#include <utils/Vector.h>
+#include <utils/RefBase.h>
+
+#include "AnotherPacketSource.h"
+#include "APacketSource.h"
+#include "ARTPConnection.h"
+#include "ASessionDescription.h"
+#include "NuPlayerSource.h"
+
+
+
+
+
+
+namespace android {
+
+struct ALooper;
+struct AnotherPacketSource;
+
+struct NuPlayer::RTPSource : public NuPlayer::Source {
+    RTPSource(
+            const sp<AMessage> &notify,
+            const String8& rtpParams);
+
+    virtual status_t getBufferingSettings(
+            BufferingSettings* buffering /* nonnull */) override;
+    virtual status_t setBufferingSettings(const BufferingSettings& buffering) override;
+
+    virtual void prepareAsync();
+    virtual void start();
+    virtual void stop();
+    virtual void pause();
+    virtual void resume();
+
+    virtual status_t feedMoreTSData();
+
+    virtual status_t dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit);
+
+    virtual status_t getDuration(int64_t *durationUs);
+    virtual status_t seekTo(
+            int64_t seekTimeUs,
+            MediaPlayerSeekMode mode = MediaPlayerSeekMode::SEEK_PREVIOUS_SYNC) override;
+
+    void onMessageReceived(const sp<AMessage> &msg);
+
+protected:
+    virtual ~RTPSource();
+
+    virtual sp<MetaData> getFormatMeta(bool audio);
+
+private:
+    enum {
+        kWhatAccessUnit = 'accU',
+        kWhatAccessUnitComplete = 'accu',
+        kWhatDisconnect = 'disc',
+        kWhatEOS = 'eos!',
+        kWhatPollBuffering = 'poll',
+        kWhatSetBufferingSettings = 'sBuS',
+    };
+
+    enum State {
+        DISCONNECTED,
+        CONNECTING,
+        CONNECTED,
+        PAUSED,
+    };
+
+    struct TrackInfo {
+
+        /* SDP of track */
+        bool mIsAudio;
+        int32_t mPayloadType;
+        String8 mMimeType;
+        String8 mCodecName;
+        int32_t mCodecProfile;
+        int32_t mCodecLevel;
+        int32_t mWidth;
+        int32_t mHeight;
+        String8 mLocalIp;
+        String8 mRemoteIp;
+        int32_t mLocalPort;
+        int32_t mRemotePort;
+        int32_t mTimeScale;
+        int32_t mAS;
+
+        /* a copy of TrackInfo in RTSPSource */
+        sp<AnotherPacketSource> mSource;
+        uint32_t mRTPTime;
+        int64_t mNormalPlaytimeUs;
+        bool mNPTMappingValid;
+
+        /* a copy of TrackInfo in MyHandler.h */
+        int mRTPSocket;
+        int mRTCPSocket;
+        uint32_t mFirstSeqNumInSegment;
+        bool mNewSegment;
+        int32_t mAllowedStaleAccessUnits;
+        uint32_t mRTPAnchor;
+        int64_t mNTPAnchorUs;
+        bool mEOSReceived;
+        uint32_t mNormalPlayTimeRTP;
+        int64_t mNormalPlayTimeUs;
+        sp<APacketSource> mPacketSource;
+        List<sp<ABuffer>> mPackets;
+    };
+
+    const String8 mRTPParams;
+    uint32_t mFlags;
+    State mState;
+    status_t mFinalResult;
+
+    // below 3 parameters need to be checked whether it needed or not.
+    Mutex mBufferingLock;
+    bool mBuffering;
+    bool mInPreparationPhase;
+    Mutex mBufferingSettingsLock;
+    BufferingSettings mBufferingSettings;
+
+    sp<ALooper> mLooper;
+
+    sp<ARTPConnection> mRTPConn;
+
+    Vector<TrackInfo> mTracks;
+    sp<AnotherPacketSource> mAudioTrack;
+    sp<AnotherPacketSource> mVideoTrack;
+
+    int64_t mEOSTimeoutAudio;
+    int64_t mEOSTimeoutVideo;
+
+    /* MyHandler.h */
+    bool mFirstAccessUnit;
+    bool mAllTracksHaveTime;
+    int64_t mNTPAnchorUs;
+    int64_t mMediaAnchorUs;
+    int64_t mLastMediaTimeUs;
+    int64_t mNumAccessUnitsReceived;
+    bool mReceivedFirstRTCPPacket;
+    bool mReceivedFirstRTPPacket;
+    bool mPausing;
+    int32_t mPauseGeneration;
+
+    sp<AnotherPacketSource> getSource(bool audio);
+
+    /* MyHandler.h */
+    void onTimeUpdate(int32_t trackIndex, uint32_t rtpTime, uint64_t ntpTime);
+    bool addMediaTimestamp(int32_t trackIndex, const TrackInfo *track,
+            const sp<ABuffer> &accessUnit);
+    bool dataReceivedOnAllChannels();
+    void postQueueAccessUnit(size_t trackIndex, const sp<ABuffer> &accessUnit);
+    void postQueueEOS(size_t trackIndex, status_t finalResult);
+    sp<MetaData> getTrackFormat(size_t index, int32_t *timeScale);
+    void onConnected();
+    void onDisconnected(const sp<AMessage> &msg);
+
+    void schedulePollBuffering();
+    void onPollBuffering();
+
+    bool haveSufficientDataOnAllTracks();
+
+    void setEOSTimeout(bool audio, int64_t timeout);
+
+    status_t setParameters(const String8 &params);
+    status_t setParameter(const String8 &key, const String8 &value);
+    static void TrimString(String8 *s);
+
+    DISALLOW_EVIL_CONSTRUCTORS(RTPSource);
+};
+
+}  // namespace android
+
+#endif  // RTP_SOURCE_H_