diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 11f3016..d7ca635 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -771,9 +771,14 @@
         }
     }
 
-    // Use PV_PLAYER for rtsp for now
     if (!strncasecmp(url, "rtsp://", 7)) {
-        return PV_PLAYER;
+        char value[PROPERTY_VALUE_MAX];
+        if (!property_get("media.stagefright.enable-rtsp", value, NULL)
+            || (strcmp(value, "1") && strcasecmp(value, "true"))) {
+            // For now, we're going to use PV for rtsp-based playback
+            // by default until we can clear up a few more issues.
+            return PV_PLAYER;
+        }
     }
 
     return getDefaultPlayerType();
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index f67826e..7608ec8 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -82,6 +82,7 @@
         libvpx \
         libstagefright_mpeg2ts \
         libstagefright_httplive \
+        libstagefright_rtsp \
 
 LOCAL_SHARED_LIBRARIES += \
         libstagefright_amrnb_common \
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index 88c8ee4..4c9856d 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -20,7 +20,9 @@
 
 #include <dlfcn.h>
 
+#include "include/ARTSPController.h"
 #include "include/AwesomePlayer.h"
+#include "include/LiveSource.h"
 #include "include/Prefetcher.h"
 #include "include/SoftwareRenderer.h"
 
@@ -39,7 +41,7 @@
 
 #include <surfaceflinger/ISurface.h>
 
-#include "include/LiveSource.h"
+#include <media/stagefright/foundation/ALooper.h>
 
 namespace android {
 
@@ -393,6 +395,8 @@
         mVideoBuffer = NULL;
     }
 
+    mRTSPController.clear();
+
     if (mVideoSource != NULL) {
         mVideoSource->stop();
 
@@ -1148,7 +1152,22 @@
 
         sp<MediaExtractor> extractor =
             MediaExtractor::Create(dataSource, MEDIA_MIMETYPE_CONTAINER_MPEG2TS);
+    } else if (!strncasecmp("rtsp://", mUri.string(), 7)) {
+        if (mLooper == NULL) {
+            mLooper = new ALooper;
+            mLooper->start();
+        }
+        mRTSPController = new ARTSPController(mLooper);
+        status_t err = mRTSPController->connect(mUri.string());
 
+        LOGI("ARTSPController::connect returned %d", err);
+
+        if (err != OK) {
+            mRTSPController.clear();
+            return err;
+        }
+
+        sp<MediaExtractor> extractor = mRTSPController.get();
         return setDataSource_l(extractor);
     } else {
         dataSource = DataSource::CreateFromURI(mUri.string(), &mUriHeaders);
diff --git a/media/libstagefright/include/ARTSPController.h b/media/libstagefright/include/ARTSPController.h
new file mode 100644
index 0000000..55efd41
--- /dev/null
+++ b/media/libstagefright/include/ARTSPController.h
@@ -0,0 +1,53 @@
+/*
+ * 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 A_RTSP_CONTROLLER_H_
+
+#define A_RTSP_CONTROLLER_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/MediaExtractor.h>
+
+namespace android {
+
+struct ALooper;
+struct MyHandler;
+
+struct ARTSPController : public MediaExtractor {
+    ARTSPController(const sp<ALooper> &looper);
+
+    status_t connect(const char *url);
+    void disconnect();
+
+    virtual size_t countTracks();
+    virtual sp<MediaSource> getTrack(size_t index);
+
+    virtual sp<MetaData> getTrackMetaData(
+            size_t index, uint32_t flags);
+
+protected:
+    virtual ~ARTSPController();
+
+private:
+    sp<ALooper> mLooper;
+    sp<MyHandler> mHandler;
+
+    DISALLOW_EVIL_CONSTRUCTORS(ARTSPController);
+};
+
+}  // namespace android
+
+#endif  // A_RTSP_CONTROLLER_H_
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 9455743..182aa06 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -36,6 +36,9 @@
 struct Prefetcher;
 struct TimeSource;
 
+struct ALooper;
+struct ARTSPController;
+
 struct AwesomeRenderer : public RefBase {
     AwesomeRenderer() {}
 
@@ -169,6 +172,9 @@
     sp<Prefetcher> mPrefetcher;
     sp<HTTPDataSource> mConnectingDataSource;
 
+    sp<ALooper> mLooper;
+    sp<ARTSPController> mRTSPController;
+
     struct SuspensionState {
         String8 mUri;
         KeyedVector<String8, String8> mUriHeaders;
diff --git a/media/libstagefright/rtsp/AAVCAssembler.cpp b/media/libstagefright/rtsp/AAVCAssembler.cpp
new file mode 100644
index 0000000..3dfb200
--- /dev/null
+++ b/media/libstagefright/rtsp/AAVCAssembler.cpp
@@ -0,0 +1,385 @@
+/*
+ * 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.
+ */
+
+#include "AAVCAssembler.h"
+
+#include "ARTPSource.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+#include <stdint.h>
+
+#define BE_VERBOSE      0
+
+namespace android {
+
+// static
+AAVCAssembler::AAVCAssembler(const sp<AMessage> &notify)
+    : mNotifyMsg(notify),
+      mAccessUnitRTPTime(0),
+      mNextExpectedSeqNoValid(false),
+      mNextExpectedSeqNo(0),
+      mAccessUnitDamaged(false) {
+}
+
+AAVCAssembler::~AAVCAssembler() {
+}
+
+ARTPAssembler::AssemblyStatus AAVCAssembler::addNALUnit(
+        const sp<ARTPSource> &source) {
+    List<sp<ABuffer> > *queue = source->queue();
+
+    if (queue->empty()) {
+        return NOT_ENOUGH_DATA;
+    }
+
+    if (mNextExpectedSeqNoValid) {
+        List<sp<ABuffer> >::iterator it = queue->begin();
+        while (it != queue->end()) {
+            if ((uint32_t)(*it)->int32Data() >= mNextExpectedSeqNo) {
+                break;
+            }
+
+            it = queue->erase(it);
+        }
+
+        if (queue->empty()) {
+            return NOT_ENOUGH_DATA;
+        }
+    }
+
+    sp<ABuffer> buffer = *queue->begin();
+
+    if (!mNextExpectedSeqNoValid) {
+        mNextExpectedSeqNoValid = true;
+        mNextExpectedSeqNo = (uint32_t)buffer->int32Data();
+    } else if ((uint32_t)buffer->int32Data() != mNextExpectedSeqNo) {
+#if BE_VERBOSE
+        LOG(VERBOSE) << "Not the sequence number I expected";
+#endif
+
+        return WRONG_SEQUENCE_NUMBER;
+    }
+
+    const uint8_t *data = buffer->data();
+    size_t size = buffer->size();
+
+    if (size < 1 || (data[0] & 0x80)) {
+        // Corrupt.
+
+        LOG(ERROR) << "Ignoring corrupt buffer.";
+        queue->erase(queue->begin());
+
+        ++mNextExpectedSeqNo;
+        return MALFORMED_PACKET;
+    }
+
+    unsigned nalType = data[0] & 0x1f;
+    if (nalType >= 1 && nalType <= 23) {
+        addSingleNALUnit(buffer);
+        queue->erase(queue->begin());
+        ++mNextExpectedSeqNo;
+        return OK;
+    } else if (nalType == 28) {
+        // FU-A
+        return addFragmentedNALUnit(queue);
+    } else if (nalType == 24) {
+        // STAP-A
+        bool success = addSingleTimeAggregationPacket(buffer);
+        queue->erase(queue->begin());
+        ++mNextExpectedSeqNo;
+
+        return success ? OK : MALFORMED_PACKET;
+    } else {
+        LOG(ERROR) << "Ignoring unsupported buffer (nalType=" << nalType << ")";
+
+        queue->erase(queue->begin());
+        ++mNextExpectedSeqNo;
+
+        return MALFORMED_PACKET;
+    }
+}
+
+void AAVCAssembler::addSingleNALUnit(const sp<ABuffer> &buffer) {
+#if BE_VERBOSE
+    LOG(VERBOSE) << "addSingleNALUnit of size " << buffer->size();
+    hexdump(buffer->data(), buffer->size());
+#endif
+
+    uint32_t rtpTime;
+    CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+
+    if (!mNALUnits.empty() && rtpTime != mAccessUnitRTPTime) {
+        submitAccessUnit();
+    }
+    mAccessUnitRTPTime = rtpTime;
+
+    mNALUnits.push_back(buffer);
+}
+
+bool AAVCAssembler::addSingleTimeAggregationPacket(const sp<ABuffer> &buffer) {
+    const uint8_t *data = buffer->data();
+    size_t size = buffer->size();
+
+    if (size < 3) {
+        LOG(ERROR) << "Discarding too small STAP-A packet.";
+        return false;
+    }
+
+    ++data;
+    --size;
+    while (size >= 2) {
+        size_t nalSize = (data[0] << 8) | data[1];
+
+        if (size < nalSize + 2) {
+            LOG(ERROR) << "Discarding malformed STAP-A packet.";
+            return false;
+        }
+
+        sp<ABuffer> unit = new ABuffer(nalSize);
+        memcpy(unit->data(), &data[2], nalSize);
+
+        PropagateTimes(buffer, unit);
+
+        addSingleNALUnit(unit);
+
+        data += 2 + nalSize;
+        size -= 2 + nalSize;
+    }
+
+    if (size != 0) {
+        LOG(WARNING) << "Unexpected padding at end of STAP-A packet.";
+    }
+
+    return true;
+}
+
+ARTPAssembler::AssemblyStatus AAVCAssembler::addFragmentedNALUnit(
+        List<sp<ABuffer> > *queue) {
+    CHECK(!queue->empty());
+
+    sp<ABuffer> buffer = *queue->begin();
+    const uint8_t *data = buffer->data();
+    size_t size = buffer->size();
+
+    CHECK(size > 0);
+    unsigned indicator = data[0];
+
+    CHECK((indicator & 0x1f) == 28);
+
+    if (size < 2) {
+        LOG(ERROR) << "Ignoring malformed FU buffer (size = " << size << ")";
+
+        queue->erase(queue->begin());
+        ++mNextExpectedSeqNo;
+        return MALFORMED_PACKET;
+    }
+
+    if (!(data[1] & 0x80)) {
+        // Start bit not set on the first buffer.
+
+#if BE_VERBOSE
+        LOG(ERROR) << "Start bit not set on first buffer";
+#endif
+
+        queue->erase(queue->begin());
+        ++mNextExpectedSeqNo;
+        return MALFORMED_PACKET;
+    }
+
+    uint32_t nalType = data[1] & 0x1f;
+    uint32_t nri = (data[0] >> 5) & 3;
+
+    uint32_t expectedSeqNo = (uint32_t)buffer->int32Data() + 1;
+    size_t totalSize = size - 2;
+    size_t totalCount = 1;
+    bool complete = false;
+
+    if (data[1] & 0x40) {
+        // Huh? End bit also set on the first buffer.
+
+#if BE_VERBOSE
+        LOG(WARNING) << "Grrr. This isn't fragmented at all.";
+#endif
+
+        complete = true;
+    } else {
+        List<sp<ABuffer> >::iterator it = ++queue->begin();
+        while (it != queue->end()) {
+#if BE_VERBOSE
+            LOG(VERBOSE) << "sequence length " << totalCount;
+#endif
+
+            const sp<ABuffer> &buffer = *it;
+
+            const uint8_t *data = buffer->data();
+            size_t size = buffer->size();
+
+            if ((uint32_t)buffer->int32Data() != expectedSeqNo) {
+#if BE_VERBOSE
+                LOG(VERBOSE) << "sequence not complete, expected seqNo "
+                     << expectedSeqNo << ", got "
+                     << (uint32_t)buffer->int32Data();
+#endif
+
+                return WRONG_SEQUENCE_NUMBER;
+            }
+
+            if (size < 2
+                    || data[0] != indicator
+                    || (data[1] & 0x1f) != nalType
+                    || (data[1] & 0x80)) {
+                LOG(ERROR) << "Ignoring malformed FU buffer.\n";
+
+                // Delete the whole start of the FU.
+
+                it = queue->begin();
+                for (size_t i = 0; i <= totalCount; ++i) {
+                    it = queue->erase(it);
+                }
+
+                mNextExpectedSeqNo = expectedSeqNo + 1;
+
+                return MALFORMED_PACKET;
+            }
+
+            totalSize += size - 2;
+            ++totalCount;
+
+            expectedSeqNo = expectedSeqNo + 1;
+
+            if (data[1] & 0x40) {
+                // This is the last fragment.
+                complete = true;
+                break;
+            }
+
+            ++it;
+        }
+    }
+
+    if (!complete) {
+        return NOT_ENOUGH_DATA;
+    }
+
+    mNextExpectedSeqNo = expectedSeqNo;
+
+    // We found all the fragments that make up the complete NAL unit.
+
+    // Leave room for the header. So far totalSize did not include the
+    // header byte.
+    ++totalSize;
+
+    sp<ABuffer> unit = new ABuffer(totalSize);
+    PropagateTimes(buffer, unit);
+
+    unit->data()[0] = (nri << 5) | nalType;
+
+    size_t offset = 1;
+    List<sp<ABuffer> >::iterator it = queue->begin();
+    for (size_t i = 0; i < totalCount; ++i) {
+        const sp<ABuffer> &buffer = *it;
+
+#if BE_VERBOSE
+        LOG(VERBOSE) << "piece #" << (i + 1) << "/" << totalCount;
+        hexdump(buffer->data(), buffer->size());
+#endif
+
+        memcpy(unit->data() + offset, buffer->data() + 2, buffer->size() - 2);
+        offset += buffer->size() - 2;
+
+        it = queue->erase(it);
+    }
+
+    unit->setRange(0, totalSize);
+
+    addSingleNALUnit(unit);
+
+#if BE_VERBOSE
+    LOG(VERBOSE) << "successfully assembled a NAL unit from fragments.";
+#endif
+
+    return OK;
+}
+
+void AAVCAssembler::submitAccessUnit() {
+    CHECK(!mNALUnits.empty());
+
+#if BE_VERBOSE
+    LOG(VERBOSE) << "Access unit complete (" << mNALUnits.size() << " nal units)";
+#endif
+
+    uint64_t ntpTime;
+    CHECK((*mNALUnits.begin())->meta()->findInt64(
+                "ntp-time", (int64_t *)&ntpTime));
+
+    size_t totalSize = 0;
+    for (List<sp<ABuffer> >::iterator it = mNALUnits.begin();
+         it != mNALUnits.end(); ++it) {
+        totalSize += 4 + (*it)->size();
+    }
+
+    sp<ABuffer> accessUnit = new ABuffer(totalSize);
+    size_t offset = 0;
+    for (List<sp<ABuffer> >::iterator it = mNALUnits.begin();
+         it != mNALUnits.end(); ++it) {
+        memcpy(accessUnit->data() + offset, "\x00\x00\x00\x01", 4);
+        offset += 4;
+
+        sp<ABuffer> nal = *it;
+        memcpy(accessUnit->data() + offset, nal->data(), nal->size());
+        offset += nal->size();
+    }
+
+    accessUnit->meta()->setInt64("ntp-time", ntpTime);
+
+#if 0
+    printf(mAccessUnitDamaged ? "X" : ".");
+    fflush(stdout);
+#endif
+
+    if (mAccessUnitDamaged) {
+        accessUnit->meta()->setInt32("damaged", true);
+    }
+
+    mNALUnits.clear();
+    mAccessUnitDamaged = false;
+
+    sp<AMessage> msg = mNotifyMsg->dup();
+    msg->setObject("access-unit", accessUnit);
+    msg->post();
+}
+
+ARTPAssembler::AssemblyStatus AAVCAssembler::assembleMore(
+        const sp<ARTPSource> &source) {
+    AssemblyStatus status = addNALUnit(source);
+    if (status == MALFORMED_PACKET) {
+        mAccessUnitDamaged = true;
+    }
+    return status;
+}
+
+void AAVCAssembler::packetLost() {
+    CHECK(mNextExpectedSeqNoValid);
+    ++mNextExpectedSeqNo;
+
+    mAccessUnitDamaged = true;
+}
+
+}  // namespace android
diff --git a/media/libstagefright/rtsp/AAVCAssembler.h b/media/libstagefright/rtsp/AAVCAssembler.h
new file mode 100644
index 0000000..1e97520
--- /dev/null
+++ b/media/libstagefright/rtsp/AAVCAssembler.h
@@ -0,0 +1,61 @@
+/*
+ * 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 A_AVC_ASSEMBLER_H_
+
+#define A_AVC_ASSEMBLER_H_
+
+#include "ARTPAssembler.h"
+
+#include <utils/List.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct ABuffer;
+struct AMessage;
+
+struct AAVCAssembler : public ARTPAssembler {
+    AAVCAssembler(const sp<AMessage> &notify);
+
+protected:
+    virtual ~AAVCAssembler();
+
+    virtual AssemblyStatus assembleMore(const sp<ARTPSource> &source);
+    virtual void packetLost();
+
+private:
+    sp<AMessage> mNotifyMsg;
+
+    uint32_t mAccessUnitRTPTime;
+    bool mNextExpectedSeqNoValid;
+    uint32_t mNextExpectedSeqNo;
+    bool mAccessUnitDamaged;
+    List<sp<ABuffer> > mNALUnits;
+
+    AssemblyStatus addNALUnit(const sp<ARTPSource> &source);
+    void addSingleNALUnit(const sp<ABuffer> &buffer);
+    AssemblyStatus addFragmentedNALUnit(List<sp<ABuffer> > *queue);
+    bool addSingleTimeAggregationPacket(const sp<ABuffer> &buffer);
+
+    void submitAccessUnit();
+
+    DISALLOW_EVIL_CONSTRUCTORS(AAVCAssembler);
+};
+
+}  // namespace android
+
+#endif  // A_AVC_ASSEMBLER_H_
diff --git a/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp b/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
new file mode 100644
index 0000000..0549d84
--- /dev/null
+++ b/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
@@ -0,0 +1,166 @@
+/*
+ * 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.
+ */
+
+#include "AMPEG4AudioAssembler.h"
+
+#include "ARTPSource.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+namespace android {
+
+AMPEG4AudioAssembler::AMPEG4AudioAssembler(const sp<AMessage> &notify)
+    : mNotifyMsg(notify),
+      mAccessUnitRTPTime(0),
+      mNextExpectedSeqNoValid(false),
+      mNextExpectedSeqNo(0),
+      mAccessUnitDamaged(false) {
+}
+
+AMPEG4AudioAssembler::~AMPEG4AudioAssembler() {
+}
+
+ARTPAssembler::AssemblyStatus AMPEG4AudioAssembler::assembleMore(
+        const sp<ARTPSource> &source) {
+    AssemblyStatus status = addPacket(source);
+    if (status == MALFORMED_PACKET) {
+        mAccessUnitDamaged = true;
+    }
+    return status;
+}
+
+ARTPAssembler::AssemblyStatus AMPEG4AudioAssembler::addPacket(
+        const sp<ARTPSource> &source) {
+    List<sp<ABuffer> > *queue = source->queue();
+
+    if (queue->empty()) {
+        return NOT_ENOUGH_DATA;
+    }
+
+    if (mNextExpectedSeqNoValid) {
+        List<sp<ABuffer> >::iterator it = queue->begin();
+        while (it != queue->end()) {
+            if ((uint32_t)(*it)->int32Data() >= mNextExpectedSeqNo) {
+                break;
+            }
+
+            it = queue->erase(it);
+        }
+
+        if (queue->empty()) {
+            return NOT_ENOUGH_DATA;
+        }
+    }
+
+    sp<ABuffer> buffer = *queue->begin();
+
+    if (!mNextExpectedSeqNoValid) {
+        mNextExpectedSeqNoValid = true;
+        mNextExpectedSeqNo = (uint32_t)buffer->int32Data();
+    } else if ((uint32_t)buffer->int32Data() != mNextExpectedSeqNo) {
+#if VERBOSE
+        LOG(VERBOSE) << "Not the sequence number I expected";
+#endif
+
+        return WRONG_SEQUENCE_NUMBER;
+    }
+
+    uint32_t rtpTime;
+    CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+
+    if (mPackets.size() > 0 && rtpTime != mAccessUnitRTPTime) {
+        submitAccessUnit();
+    }
+    mAccessUnitRTPTime = rtpTime;
+
+    mPackets.push_back(buffer);
+
+    queue->erase(queue->begin());
+    ++mNextExpectedSeqNo;
+
+    return OK;
+}
+
+void AMPEG4AudioAssembler::submitAccessUnit() {
+    CHECK(!mPackets.empty());
+
+#if VERBOSE
+    LOG(VERBOSE) << "Access unit complete (" << mPackets.size() << " packets)";
+#endif
+
+    uint64_t ntpTime;
+    CHECK((*mPackets.begin())->meta()->findInt64(
+                "ntp-time", (int64_t *)&ntpTime));
+
+    size_t totalSize = 0;
+    List<sp<ABuffer> >::iterator it = mPackets.begin();
+    while (it != mPackets.end()) {
+        const sp<ABuffer> &unit = *it;
+
+        size_t n = 0;
+        while (unit->data()[n] == 0xff) {
+            ++n;
+        }
+        ++n;
+
+        totalSize += unit->size() - n;
+        ++it;
+    }
+
+    sp<ABuffer> accessUnit = new ABuffer(totalSize);
+    size_t offset = 0;
+    it = mPackets.begin();
+    while (it != mPackets.end()) {
+        const sp<ABuffer> &unit = *it;
+
+        size_t n = 0;
+        while (unit->data()[n] == 0xff) {
+            ++n;
+        }
+        ++n;
+
+        memcpy((uint8_t *)accessUnit->data() + offset,
+               unit->data() + n, unit->size() - n);
+
+        offset += unit->size() - n;
+
+        ++it;
+    }
+
+    accessUnit->meta()->setInt64("ntp-time", ntpTime);
+
+    if (mAccessUnitDamaged) {
+        accessUnit->meta()->setInt32("damaged", true);
+    }
+
+    mPackets.clear();
+    mAccessUnitDamaged = false;
+
+    sp<AMessage> msg = mNotifyMsg->dup();
+    msg->setObject("access-unit", accessUnit);
+    msg->post();
+}
+
+void AMPEG4AudioAssembler::packetLost() {
+    CHECK(mNextExpectedSeqNoValid);
+    ++mNextExpectedSeqNo;
+
+    mAccessUnitDamaged = true;
+}
+
+}  // namespace android
diff --git a/media/libstagefright/rtsp/AMPEG4AudioAssembler.h b/media/libstagefright/rtsp/AMPEG4AudioAssembler.h
new file mode 100644
index 0000000..5c2a2dd
--- /dev/null
+++ b/media/libstagefright/rtsp/AMPEG4AudioAssembler.h
@@ -0,0 +1,56 @@
+/*
+ * 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 A_MPEG4_AUDIO_ASSEMBLER_H_
+
+#define A_MPEG4_AUDIO_ASSEMBLER_H_
+
+#include "ARTPAssembler.h"
+
+#include <utils/List.h>
+
+#include <stdint.h>
+
+namespace android {
+
+struct AMessage;
+
+struct AMPEG4AudioAssembler : public ARTPAssembler {
+    AMPEG4AudioAssembler(const sp<AMessage> &notify);
+
+protected:
+    virtual ~AMPEG4AudioAssembler();
+
+    virtual AssemblyStatus assembleMore(const sp<ARTPSource> &source);
+    virtual void packetLost();
+
+private:
+    sp<AMessage> mNotifyMsg;
+    uint32_t mAccessUnitRTPTime;
+    bool mNextExpectedSeqNoValid;
+    uint32_t mNextExpectedSeqNo;
+    bool mAccessUnitDamaged;
+    List<sp<ABuffer> > mPackets;
+
+    AssemblyStatus addPacket(const sp<ARTPSource> &source);
+    void submitAccessUnit();
+
+    DISALLOW_EVIL_CONSTRUCTORS(AMPEG4AudioAssembler);
+};
+
+}  // namespace android
+
+#endif  // A_MPEG4_AUDIO_ASSEMBLER_H_
diff --git a/media/libstagefright/rtsp/APacketSource.cpp b/media/libstagefright/rtsp/APacketSource.cpp
new file mode 100644
index 0000000..2869d54
--- /dev/null
+++ b/media/libstagefright/rtsp/APacketSource.cpp
@@ -0,0 +1,345 @@
+/*
+ * 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.
+ */
+
+#include "APacketSource.h"
+
+#include "ASessionDescription.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/base64.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+static bool GetAttribute(const char *s, const char *key, AString *value) {
+    value->clear();
+
+    size_t keyLen = strlen(key);
+
+    for (;;) {
+        const char *colonPos = strchr(s, ';');
+
+        size_t len =
+            (colonPos == NULL) ? strlen(s) : colonPos - s;
+
+        if (len >= keyLen + 1 && s[keyLen] == '=' && !strncmp(s, key, keyLen)) {
+            value->setTo(&s[keyLen + 1], len - keyLen - 1);
+            return true;
+        }
+
+        if (colonPos == NULL) {
+            return false;
+        }
+
+        s = colonPos + 1;
+    }
+}
+
+static sp<ABuffer> decodeHex(const AString &s) {
+    if ((s.size() % 2) != 0) {
+        return NULL;
+    }
+
+    size_t outLen = s.size() / 2;
+    sp<ABuffer> buffer = new ABuffer(outLen);
+    uint8_t *out = buffer->data();
+
+    uint8_t accum = 0;
+    for (size_t i = 0; i < s.size(); ++i) {
+        char c = s.c_str()[i];
+        unsigned value;
+        if (c >= '0' && c <= '9') {
+            value = c - '0';
+        } else if (c >= 'a' && c <= 'f') {
+            value = c - 'a' + 10;
+        } else if (c >= 'A' && c <= 'F') {
+            value = c - 'A' + 10;
+        } else {
+            return NULL;
+        }
+
+        accum = (accum << 4) | value;
+
+        if (i & 1) {
+            *out++ = accum;
+
+            accum = 0;
+        }
+    }
+
+    return buffer;
+}
+
+static sp<ABuffer> MakeAVCCodecSpecificData(const char *params) {
+    AString val;
+    CHECK(GetAttribute(params, "profile-level-id", &val));
+
+    sp<ABuffer> profileLevelID = decodeHex(val);
+    CHECK(profileLevelID != NULL);
+    CHECK_EQ(profileLevelID->size(), 3u);
+
+    Vector<sp<ABuffer> > paramSets;
+
+    size_t numSeqParameterSets = 0;
+    size_t totalSeqParameterSetSize = 0;
+    size_t numPicParameterSets = 0;
+    size_t totalPicParameterSetSize = 0;
+
+    CHECK(GetAttribute(params, "sprop-parameter-sets", &val));
+    size_t start = 0;
+    for (;;) {
+        ssize_t commaPos = val.find(",", start);
+        size_t end = (commaPos < 0) ? val.size() : commaPos;
+
+        AString nalString(val, start, end - start);
+        sp<ABuffer> nal = decodeBase64(nalString);
+        CHECK(nal != NULL);
+        CHECK_GT(nal->size(), 0u);
+        CHECK_LE(nal->size(), 65535u);
+
+        uint8_t nalType = nal->data()[0] & 0x1f;
+        if (numSeqParameterSets == 0) {
+            CHECK_EQ((unsigned)nalType, 7u);
+        } else if (numPicParameterSets > 0) {
+            CHECK_EQ((unsigned)nalType, 8u);
+        }
+        if (nalType == 7) {
+            ++numSeqParameterSets;
+            totalSeqParameterSetSize += nal->size();
+        } else  {
+            CHECK_EQ((unsigned)nalType, 8u);
+            ++numPicParameterSets;
+            totalPicParameterSetSize += nal->size();
+        }
+
+        paramSets.push(nal);
+
+        if (commaPos < 0) {
+            break;
+        }
+
+        start = commaPos + 1;
+    }
+
+    CHECK_LT(numSeqParameterSets, 32u);
+    CHECK_LE(numPicParameterSets, 255u);
+
+    size_t csdSize =
+        1 + 3 + 1 + 1
+        + 2 * numSeqParameterSets + totalSeqParameterSetSize
+        + 1 + 2 * numPicParameterSets + totalPicParameterSetSize;
+
+    sp<ABuffer> csd = new ABuffer(csdSize);
+    uint8_t *out = csd->data();
+
+    *out++ = 0x01;  // configurationVersion
+    memcpy(out, profileLevelID->data(), 3);
+    out += 3;
+    *out++ = (0x3f << 2) | 1;  // lengthSize == 2 bytes
+    *out++ = 0xe0 | numSeqParameterSets;
+
+    for (size_t i = 0; i < numSeqParameterSets; ++i) {
+        sp<ABuffer> nal = paramSets.editItemAt(i);
+
+        *out++ = nal->size() >> 8;
+        *out++ = nal->size() & 0xff;
+
+        memcpy(out, nal->data(), nal->size());
+
+        out += nal->size();
+    }
+
+    *out++ = numPicParameterSets;
+
+    for (size_t i = 0; i < numPicParameterSets; ++i) {
+        sp<ABuffer> nal = paramSets.editItemAt(i + numSeqParameterSets);
+
+        *out++ = nal->size() >> 8;
+        *out++ = nal->size() & 0xff;
+
+        memcpy(out, nal->data(), nal->size());
+
+        out += nal->size();
+    }
+
+    hexdump(csd->data(), csd->size());
+
+    return csd;
+}
+
+sp<ABuffer> MakeAACCodecSpecificData(const char *params) {
+    AString val;
+    CHECK(GetAttribute(params, "config", &val));
+
+    sp<ABuffer> config = decodeHex(val);
+    CHECK(config != NULL);
+    CHECK_GE(config->size(), 4u);
+
+    const uint8_t *data = config->data();
+    uint32_t x = data[0] << 24 | data[1] << 16 | data[2] << 8 | data[3];
+    x = (x >> 1) & 0xffff;
+
+    static const uint8_t kStaticESDS[] = {
+        0x03, 22,
+        0x00, 0x00,     // ES_ID
+        0x00,           // streamDependenceFlag, URL_Flag, OCRstreamFlag
+
+        0x04, 17,
+        0x40,                       // Audio ISO/IEC 14496-3
+        0x00, 0x00, 0x00, 0x00,
+        0x00, 0x00, 0x00, 0x00,
+        0x00, 0x00, 0x00, 0x00,
+
+        0x05, 2,
+        // AudioSpecificInfo follows
+    };
+
+    sp<ABuffer> csd = new ABuffer(sizeof(kStaticESDS) + 2);
+    memcpy(csd->data(), kStaticESDS, sizeof(kStaticESDS));
+    csd->data()[sizeof(kStaticESDS)] = (x >> 8) & 0xff;
+    csd->data()[sizeof(kStaticESDS) + 1] = x & 0xff;
+
+    hexdump(csd->data(), csd->size());
+
+    return csd;
+}
+
+APacketSource::APacketSource(
+        const sp<ASessionDescription> &sessionDesc, size_t index)
+    : mFormat(new MetaData),
+      mEOSResult(OK) {
+    unsigned long PT;
+    AString desc;
+    AString params;
+    sessionDesc->getFormatType(index, &PT, &desc, &params);
+
+    int64_t durationUs;
+    if (sessionDesc->getDurationUs(&durationUs)) {
+        mFormat->setInt64(kKeyDuration, durationUs);
+    } else {
+        mFormat->setInt64(kKeyDuration, 60 * 60 * 1000000ll);
+    }
+
+    if (!strncmp(desc.c_str(), "H264/", 5)) {
+        mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
+
+        int32_t width, height;
+        sessionDesc->getDimensions(index, PT, &width, &height);
+
+        mFormat->setInt32(kKeyWidth, width);
+        mFormat->setInt32(kKeyHeight, height);
+
+        sp<ABuffer> codecSpecificData =
+            MakeAVCCodecSpecificData(params.c_str());
+
+        mFormat->setData(
+                kKeyAVCC, 0,
+                codecSpecificData->data(), codecSpecificData->size());
+
+    } else if (!strncmp(desc.c_str(), "MP4A-LATM", 9)) {
+        mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AAC);
+
+        int32_t sampleRate, numChannels;
+        ASessionDescription::ParseFormatDesc(
+                desc.c_str(), &sampleRate, &numChannels);
+
+        mFormat->setInt32(kKeySampleRate, sampleRate);
+        mFormat->setInt32(kKeyChannelCount, numChannels);
+
+        sp<ABuffer> codecSpecificData =
+            MakeAACCodecSpecificData(params.c_str());
+
+        mFormat->setData(
+                kKeyESDS, 0,
+                codecSpecificData->data(), codecSpecificData->size());
+    } else {
+        TRESPASS();
+    }
+}
+
+APacketSource::~APacketSource() {
+}
+
+status_t APacketSource::start(MetaData *params) {
+    return OK;
+}
+
+status_t APacketSource::stop() {
+    return OK;
+}
+
+sp<MetaData> APacketSource::getFormat() {
+    return mFormat;
+}
+
+status_t APacketSource::read(
+        MediaBuffer **out, const ReadOptions *) {
+    *out = NULL;
+
+    Mutex::Autolock autoLock(mLock);
+    while (mEOSResult == OK && mBuffers.empty()) {
+        mCondition.wait(mLock);
+    }
+
+    if (!mBuffers.empty()) {
+        const sp<ABuffer> buffer = *mBuffers.begin();
+
+        uint64_t ntpTime;
+        CHECK(buffer->meta()->findInt64(
+                    "ntp-time", (int64_t *)&ntpTime));
+
+        int64_t timeUs = (int64_t)(ntpTime * 1E6 / (1ll << 32));
+
+        MediaBuffer *mediaBuffer = new MediaBuffer(buffer->size());
+        mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
+        memcpy(mediaBuffer->data(), buffer->data(), buffer->size());
+        *out = mediaBuffer;
+
+        mBuffers.erase(mBuffers.begin());
+        return OK;
+    }
+
+    return mEOSResult;
+}
+
+void APacketSource::queueAccessUnit(const sp<ABuffer> &buffer) {
+    int32_t damaged;
+    if (buffer->meta()->findInt32("damaged", &damaged) && damaged) {
+        // LOG(VERBOSE) << "discarding damaged AU";
+        return;
+    }
+
+    Mutex::Autolock autoLock(mLock);
+    mBuffers.push_back(buffer);
+    mCondition.signal();
+}
+
+void APacketSource::signalEOS(status_t result) {
+    CHECK(result != OK);
+
+    Mutex::Autolock autoLock(mLock);
+    mEOSResult = result;
+    mCondition.signal();
+}
+
+}  // namespace android
diff --git a/media/libstagefright/rtsp/APacketSource.h b/media/libstagefright/rtsp/APacketSource.h
new file mode 100644
index 0000000..4040eee
--- /dev/null
+++ b/media/libstagefright/rtsp/APacketSource.h
@@ -0,0 +1,61 @@
+/*
+ * 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 A_PACKET_SOURCE_H_
+
+#define A_PACKET_SOURCE_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/threads.h>
+#include <utils/List.h>
+
+namespace android {
+
+struct ABuffer;
+struct ASessionDescription;
+
+struct APacketSource : public MediaSource {
+    APacketSource(const sp<ASessionDescription> &sessionDesc, size_t index);
+
+    virtual status_t start(MetaData *params = NULL);
+    virtual status_t stop();
+    virtual sp<MetaData> getFormat();
+
+    virtual status_t read(
+            MediaBuffer **buffer, const ReadOptions *options = NULL);
+
+    void queueAccessUnit(const sp<ABuffer> &buffer);
+    void signalEOS(status_t result);
+
+protected:
+    virtual ~APacketSource();
+
+private:
+    Mutex mLock;
+    Condition mCondition;
+
+    sp<MetaData> mFormat;
+    List<sp<ABuffer> > mBuffers;
+    status_t mEOSResult;
+
+    DISALLOW_EVIL_CONSTRUCTORS(APacketSource);
+};
+
+
+}  // namespace android
+
+#endif  // A_PACKET_SOURCE_H_
diff --git a/media/libstagefright/rtsp/ARTPAssembler.cpp b/media/libstagefright/rtsp/ARTPAssembler.cpp
new file mode 100644
index 0000000..24225b8
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTPAssembler.cpp
@@ -0,0 +1,78 @@
+/*
+ * 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.
+ */
+
+#include "ARTPAssembler.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include <stdint.h>
+
+namespace android {
+
+static int64_t getNowUs() {
+    struct timeval tv;
+    gettimeofday(&tv, NULL);
+
+    return (int64_t)tv.tv_usec + tv.tv_sec * 1000000ll;
+}
+
+ARTPAssembler::ARTPAssembler()
+    : mFirstFailureTimeUs(-1) {
+}
+
+void ARTPAssembler::PropagateTimes(
+        const sp<ABuffer> &from, const sp<ABuffer> &to) {
+    uint32_t rtpTime;
+    CHECK(from->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+
+    uint64_t ntpTime = 0;
+    CHECK(from->meta()->findInt64("ntp-time", (int64_t *)&ntpTime));
+
+    to->meta()->setInt32("rtp-time", rtpTime);
+    to->meta()->setInt64("ntp-time", ntpTime);
+}
+
+void ARTPAssembler::onPacketReceived(const sp<ARTPSource> &source) {
+    AssemblyStatus status;
+    for (;;) {
+        status = assembleMore(source);
+
+        if (status == WRONG_SEQUENCE_NUMBER) {
+            if (mFirstFailureTimeUs >= 0) {
+                if (getNowUs() - mFirstFailureTimeUs > 10000ll) {
+                    mFirstFailureTimeUs = -1;
+
+                    // LOG(VERBOSE) << "waited too long for packet.";
+                    packetLost();
+                    continue;
+                }
+            } else {
+                mFirstFailureTimeUs = getNowUs();
+            }
+            break;
+        } else {
+            mFirstFailureTimeUs = -1;
+
+            if (status == NOT_ENOUGH_DATA) {
+                break;
+            }
+        }
+    }
+}
+
+}  // namespace android
diff --git a/media/libstagefright/rtsp/ARTPAssembler.h b/media/libstagefright/rtsp/ARTPAssembler.h
new file mode 100644
index 0000000..892bd65
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTPAssembler.h
@@ -0,0 +1,56 @@
+/*
+ * 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 A_RTP_ASSEMBLER_H_
+
+#define A_RTP_ASSEMBLER_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct ABuffer;
+struct ARTPSource;
+
+struct ARTPAssembler : public RefBase {
+    enum AssemblyStatus {
+        MALFORMED_PACKET,
+        WRONG_SEQUENCE_NUMBER,
+        NOT_ENOUGH_DATA,
+        OK
+    };
+
+    ARTPAssembler();
+
+    void onPacketReceived(const sp<ARTPSource> &source);
+
+protected:
+    static void PropagateTimes(
+        const sp<ABuffer> &from, const sp<ABuffer> &to);
+
+    virtual AssemblyStatus assembleMore(const sp<ARTPSource> &source) = 0;
+    virtual void packetLost() = 0;
+
+private:
+    int64_t mFirstFailureTimeUs;
+
+    DISALLOW_EVIL_CONSTRUCTORS(ARTPAssembler);
+};
+
+}  // namespace android
+
+#endif  // A_RTP_ASSEMBLER_H_
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
new file mode 100644
index 0000000..a4413f0
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -0,0 +1,499 @@
+/*
+ * 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.
+ */
+
+#include "ARTPConnection.h"
+
+#include "ARTPSource.h"
+#include "ASessionDescription.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/AString.h>
+
+#include <arpa/inet.h>
+#include <sys/socket.h>
+
+#define VERBOSE         0
+
+#if VERBOSE
+#include "hexdump.h"
+#endif
+
+namespace android {
+
+static uint16_t u16at(const uint8_t *data) {
+    return data[0] << 8 | data[1];
+}
+
+static uint32_t u32at(const uint8_t *data) {
+    return u16at(data) << 16 | u16at(&data[2]);
+}
+
+static uint64_t u64at(const uint8_t *data) {
+    return (uint64_t)(u32at(data)) << 32 | u32at(&data[4]);
+}
+
+// static
+const int64_t ARTPConnection::kSelectTimeoutUs = 1000ll;
+
+struct ARTPConnection::StreamInfo {
+    int mRTPSocket;
+    int mRTCPSocket;
+    sp<ASessionDescription> mSessionDesc;
+    size_t mIndex;
+    sp<AMessage> mNotifyMsg;
+};
+
+ARTPConnection::ARTPConnection()
+    : mPollEventPending(false) {
+}
+
+ARTPConnection::~ARTPConnection() {
+}
+
+void ARTPConnection::addStream(
+        int rtpSocket, int rtcpSocket,
+        const sp<ASessionDescription> &sessionDesc,
+        size_t index,
+        const sp<AMessage> &notify) {
+    sp<AMessage> msg = new AMessage(kWhatAddStream, id());
+    msg->setInt32("rtp-socket", rtpSocket);
+    msg->setInt32("rtcp-socket", rtcpSocket);
+    msg->setObject("session-desc", sessionDesc);
+    msg->setSize("index", index);
+    msg->setMessage("notify", notify);
+    msg->post();
+}
+
+void ARTPConnection::removeStream(int rtpSocket, int rtcpSocket) {
+    sp<AMessage> msg = new AMessage(kWhatRemoveStream, id());
+    msg->setInt32("rtp-socket", rtpSocket);
+    msg->setInt32("rtcp-socket", rtcpSocket);
+    msg->post();
+}
+
+static void bumpSocketBufferSize(int s) {
+    int size = 256 * 1024;
+    CHECK_EQ(setsockopt(s, SOL_SOCKET, SO_RCVBUF, &size, sizeof(size)), 0);
+}
+
+// static
+void ARTPConnection::MakePortPair(
+        int *rtpSocket, int *rtcpSocket, unsigned *rtpPort) {
+    *rtpSocket = socket(AF_INET, SOCK_DGRAM, 0);
+    CHECK_GE(*rtpSocket, 0);
+
+    bumpSocketBufferSize(*rtpSocket);
+
+    *rtcpSocket = socket(AF_INET, SOCK_DGRAM, 0);
+    CHECK_GE(*rtcpSocket, 0);
+
+    bumpSocketBufferSize(*rtcpSocket);
+
+    unsigned start = (rand() * 1000)/ RAND_MAX + 15550;
+    start &= ~1;
+
+    for (unsigned port = start; port < 65536; port += 2) {
+        struct sockaddr_in addr;
+        memset(addr.sin_zero, 0, sizeof(addr.sin_zero));
+        addr.sin_family = AF_INET;
+        addr.sin_addr.s_addr = INADDR_ANY;
+        addr.sin_port = htons(port);
+
+        if (bind(*rtpSocket,
+                 (const struct sockaddr *)&addr, sizeof(addr)) < 0) {
+            continue;
+        }
+
+        addr.sin_port = htons(port + 1);
+
+        if (bind(*rtcpSocket,
+                 (const struct sockaddr *)&addr, sizeof(addr)) == 0) {
+            *rtpPort = port;
+            return;
+        }
+    }
+
+    TRESPASS();
+}
+
+void ARTPConnection::onMessageReceived(const sp<AMessage> &msg) {
+    switch (msg->what()) {
+        case kWhatAddStream:
+        {
+            onAddStream(msg);
+            break;
+        }
+
+        case kWhatRemoveStream:
+        {
+            onRemoveStream(msg);
+            break;
+        }
+
+        case kWhatPollStreams:
+        {
+            onPollStreams();
+            break;
+        }
+
+        default:
+        {
+            TRESPASS();
+            break;
+        }
+    }
+}
+
+void ARTPConnection::onAddStream(const sp<AMessage> &msg) {
+    mStreams.push_back(StreamInfo());
+    StreamInfo *info = &*--mStreams.end();
+
+    int32_t s;
+    CHECK(msg->findInt32("rtp-socket", &s));
+    info->mRTPSocket = s;
+    CHECK(msg->findInt32("rtcp-socket", &s));
+    info->mRTCPSocket = s;
+
+    sp<RefBase> obj;
+    CHECK(msg->findObject("session-desc", &obj));
+    info->mSessionDesc = static_cast<ASessionDescription *>(obj.get());
+
+    CHECK(msg->findSize("index", &info->mIndex));
+    CHECK(msg->findMessage("notify", &info->mNotifyMsg));
+
+    postPollEvent();
+}
+
+void ARTPConnection::onRemoveStream(const sp<AMessage> &msg) {
+    int32_t rtpSocket, rtcpSocket;
+    CHECK(msg->findInt32("rtp-socket", &rtpSocket));
+    CHECK(msg->findInt32("rtcp-socket", &rtcpSocket));
+
+    List<StreamInfo>::iterator it = mStreams.begin();
+    while (it != mStreams.end()
+           && (it->mRTPSocket != rtpSocket || it->mRTCPSocket != rtcpSocket)) {
+        ++it;
+    }
+
+    if (it == mStreams.end()) {
+        TRESPASS();
+    }
+
+    mStreams.erase(it);
+}
+
+void ARTPConnection::postPollEvent() {
+    if (mPollEventPending) {
+        return;
+    }
+
+    sp<AMessage> msg = new AMessage(kWhatPollStreams, id());
+    msg->post();
+
+    mPollEventPending = true;
+}
+
+void ARTPConnection::onPollStreams() {
+    mPollEventPending = false;
+
+    if (mStreams.empty()) {
+        return;
+    }
+
+    struct timeval tv;
+    tv.tv_sec = 0;
+    tv.tv_usec = kSelectTimeoutUs;
+
+    fd_set rs;
+    FD_ZERO(&rs);
+
+    int maxSocket = -1;
+    for (List<StreamInfo>::iterator it = mStreams.begin();
+         it != mStreams.end(); ++it) {
+        FD_SET(it->mRTPSocket, &rs);
+        FD_SET(it->mRTCPSocket, &rs);
+
+        if (it->mRTPSocket > maxSocket) {
+            maxSocket = it->mRTPSocket;
+        }
+        if (it->mRTCPSocket > maxSocket) {
+            maxSocket = it->mRTCPSocket;
+        }
+    }
+
+    int res = select(maxSocket + 1, &rs, NULL, NULL, &tv);
+    CHECK_GE(res, 0);
+
+    if (res > 0) {
+        for (List<StreamInfo>::iterator it = mStreams.begin();
+             it != mStreams.end(); ++it) {
+            if (FD_ISSET(it->mRTPSocket, &rs)) {
+                receive(&*it, true);
+            }
+            if (FD_ISSET(it->mRTCPSocket, &rs)) {
+                receive(&*it, false);
+            }
+        }
+    }
+
+    postPollEvent();
+}
+
+status_t ARTPConnection::receive(StreamInfo *s, bool receiveRTP) {
+    sp<ABuffer> buffer = new ABuffer(65536);
+
+    struct sockaddr_in from;
+    socklen_t fromSize = sizeof(from);
+
+    ssize_t nbytes = recvfrom(
+            receiveRTP ? s->mRTPSocket : s->mRTCPSocket,
+            buffer->data(),
+            buffer->capacity(),
+            0,
+            (struct sockaddr *)&from,
+            &fromSize);
+
+    if (nbytes < 0) {
+        return -1;
+    }
+
+    buffer->setRange(0, nbytes);
+
+    status_t err;
+    if (receiveRTP) {
+        err = parseRTP(s, buffer);
+    } else {
+        err = parseRTCP(s, buffer);
+    }
+
+    return err;
+}
+
+status_t ARTPConnection::parseRTP(StreamInfo *s, const sp<ABuffer> &buffer) {
+    size_t size = buffer->size();
+
+    if (size < 12) {
+        // Too short to be a valid RTP header.
+        return -1;
+    }
+
+    const uint8_t *data = buffer->data();
+
+    if ((data[0] >> 6) != 2) {
+        // Unsupported version.
+        return -1;
+    }
+
+    if (data[0] & 0x20) {
+        // Padding present.
+
+        size_t paddingLength = data[size - 1];
+
+        if (paddingLength + 12 > size) {
+            // If we removed this much padding we'd end up with something
+            // that's too short to be a valid RTP header.
+            return -1;
+        }
+
+        size -= paddingLength;
+    }
+
+    int numCSRCs = data[0] & 0x0f;
+
+    size_t payloadOffset = 12 + 4 * numCSRCs;
+
+    if (size < payloadOffset) {
+        // Not enough data to fit the basic header and all the CSRC entries.
+        return -1;
+    }
+
+    if (data[0] & 0x10) {
+        // Header eXtension present.
+
+        if (size < payloadOffset + 4) {
+            // Not enough data to fit the basic header, all CSRC entries
+            // and the first 4 bytes of the extension header.
+
+            return -1;
+        }
+
+        const uint8_t *extensionData = &data[payloadOffset];
+
+        size_t extensionLength =
+            4 * (extensionData[2] << 8 | extensionData[3]);
+
+        if (size < payloadOffset + 4 + extensionLength) {
+            return -1;
+        }
+
+        payloadOffset += 4 + extensionLength;
+    }
+
+    uint32_t srcId = u32at(&data[8]);
+
+    sp<ARTPSource> source;
+    ssize_t index = mSources.indexOfKey(srcId);
+    if (index < 0) {
+        index = mSources.size();
+
+        source = new ARTPSource(
+                srcId, s->mSessionDesc, s->mIndex, s->mNotifyMsg);
+
+        mSources.add(srcId, source);
+    } else {
+        source = mSources.valueAt(index);
+    }
+
+    uint32_t rtpTime = u32at(&data[4]);
+
+    sp<AMessage> meta = buffer->meta();
+    meta->setInt32("ssrc", srcId);
+    meta->setInt32("rtp-time", rtpTime);
+    meta->setInt32("PT", data[1] & 0x7f);
+    meta->setInt32("M", data[1] >> 7);
+
+    buffer->setInt32Data(u16at(&data[2]));
+
+#if VERBOSE
+    printf("RTP = {\n"
+           "  PT: %d\n"
+           "  sequence number: %d\n"
+           "  RTP-time: 0x%08x\n"
+           "  M: %d\n"
+           "  SSRC: 0x%08x\n"
+           "}\n",
+           data[1] & 0x7f,
+           u16at(&data[2]),
+           rtpTime,
+           data[1] >> 7,
+           srcId);
+
+    // hexdump(&data[payloadOffset], size - payloadOffset);
+#endif
+
+    buffer->setRange(payloadOffset, size - payloadOffset);
+
+    source->processRTPPacket(buffer);
+
+    return OK;
+}
+
+status_t ARTPConnection::parseRTCP(StreamInfo *s, const sp<ABuffer> &buffer) {
+    const uint8_t *data = buffer->data();
+    size_t size = buffer->size();
+
+    while (size > 0) {
+        if (size < 8) {
+            // Too short to be a valid RTCP header
+            return -1;
+        }
+
+        if ((data[0] >> 6) != 2) {
+            // Unsupported version.
+            return -1;
+        }
+
+        if (data[0] & 0x20) {
+            // Padding present.
+
+            size_t paddingLength = data[size - 1];
+
+            if (paddingLength + 12 > size) {
+                // If we removed this much padding we'd end up with something
+                // that's too short to be a valid RTP header.
+                return -1;
+            }
+
+            size -= paddingLength;
+        }
+
+        size_t headerLength = 4 * (data[2] << 8 | data[3]) + 4;
+
+        if (size < headerLength) {
+            // Only received a partial packet?
+            return -1;
+        }
+
+        switch (data[1]) {
+            case 200:
+            {
+                parseSR(s, data, headerLength);
+                break;
+            }
+
+            default:
+            {
+#if VERBOSE
+                printf("Unknown RTCP packet type %d of size %ld\n",
+                       data[1], headerLength);
+
+                hexdump(data, headerLength);
+#endif
+                break;
+            }
+        }
+
+        data += headerLength;
+        size -= headerLength;
+    }
+
+    return OK;
+}
+
+status_t ARTPConnection::parseSR(
+        StreamInfo *s, const uint8_t *data, size_t size) {
+    size_t RC = data[0] & 0x1f;
+
+    if (size < (7 + RC * 6) * 4) {
+        // Packet too short for the minimal SR header.
+        return -1;
+    }
+
+    uint32_t id = u32at(&data[4]);
+    uint64_t ntpTime = u64at(&data[8]);
+    uint32_t rtpTime = u32at(&data[16]);
+
+#if VERBOSE
+    printf("SR = {\n"
+           "  SSRC:      0x%08x\n"
+           "  NTP-time:  0x%016llx\n"
+           "  RTP-time:  0x%08x\n"
+           "}\n",
+           id, ntpTime, rtpTime);
+#endif
+
+    sp<ARTPSource> source;
+    ssize_t index = mSources.indexOfKey(id);
+    if (index < 0) {
+        index = mSources.size();
+
+        source = new ARTPSource(
+                id, s->mSessionDesc, s->mIndex, s->mNotifyMsg);
+
+        mSources.add(id, source);
+    } else {
+        source = mSources.valueAt(index);
+    }
+
+    source->timeUpdate(rtpTime, ntpTime);
+
+    return 0;
+}
+
+}  // namespace android
+
diff --git a/media/libstagefright/rtsp/ARTPConnection.h b/media/libstagefright/rtsp/ARTPConnection.h
new file mode 100644
index 0000000..c77e3a4
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTPConnection.h
@@ -0,0 +1,83 @@
+/*
+ * 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 A_RTP_CONNECTION_H_
+
+#define A_RTP_CONNECTION_H_
+
+#include <media/stagefright/foundation/AHandler.h>
+#include <utils/List.h>
+
+namespace android {
+
+struct ABuffer;
+struct ARTPSource;
+struct ASessionDescription;
+
+struct ARTPConnection : public AHandler {
+    ARTPConnection();
+
+    void addStream(
+            int rtpSocket, int rtcpSocket,
+            const sp<ASessionDescription> &sessionDesc, size_t index,
+            const sp<AMessage> &notify);
+
+    void removeStream(int rtpSocket, int rtcpSocket);
+
+    // Creates a pair of UDP datagram sockets bound to adjacent ports
+    // (the rtpSocket is bound to an even port, the rtcpSocket to the
+    // next higher port).
+    static void MakePortPair(
+            int *rtpSocket, int *rtcpSocket, unsigned *rtpPort);
+
+protected:
+    virtual ~ARTPConnection();
+    virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+    enum {
+        kWhatAddStream,
+        kWhatRemoveStream,
+        kWhatPollStreams,
+    };
+
+    static const int64_t kSelectTimeoutUs;
+
+    struct StreamInfo;
+    List<StreamInfo> mStreams;
+
+    KeyedVector<uint32_t, sp<ARTPSource> > mSources;
+
+    bool mPollEventPending;
+
+    void onAddStream(const sp<AMessage> &msg);
+    void onRemoveStream(const sp<AMessage> &msg);
+    void onPollStreams();
+
+    status_t receive(StreamInfo *info, bool receiveRTP);
+
+    status_t parseRTP(StreamInfo *info, const sp<ABuffer> &buffer);
+    status_t parseRTCP(StreamInfo *info, const sp<ABuffer> &buffer);
+    status_t parseSR(StreamInfo *info, const uint8_t *data, size_t size);
+
+    void postPollEvent();
+
+    DISALLOW_EVIL_CONSTRUCTORS(ARTPConnection);
+};
+
+}  // namespace android
+
+#endif  // A_RTP_CONNECTION_H_
diff --git a/media/libstagefright/rtsp/ARTPSource.cpp b/media/libstagefright/rtsp/ARTPSource.cpp
new file mode 100644
index 0000000..f05daa8
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTPSource.cpp
@@ -0,0 +1,250 @@
+/*
+ * 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.
+ */
+
+#include "ARTPSource.h"
+
+#include "AAVCAssembler.h"
+#include "AMPEG4AudioAssembler.h"
+#include "ASessionDescription.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#define VERBOSE         0
+
+namespace android {
+
+ARTPSource::ARTPSource(
+        uint32_t id,
+        const sp<ASessionDescription> &sessionDesc, size_t index,
+        const sp<AMessage> &notify)
+    : mID(id),
+      mHighestSeqNumber(0),
+      mNumBuffersReceived(0),
+      mNumTimes(0) {
+    unsigned long PT;
+    AString desc;
+    AString params;
+    sessionDesc->getFormatType(index, &PT, &desc, &params);
+
+    if (!strncmp(desc.c_str(), "H264/", 5)) {
+        mAssembler = new AAVCAssembler(notify);
+    } else if (!strncmp(desc.c_str(), "MP4A-LATM", 9)) {
+        mAssembler = new AMPEG4AudioAssembler(notify);
+    } else {
+        TRESPASS();
+    }
+}
+
+static uint32_t AbsDiff(uint32_t seq1, uint32_t seq2) {
+    return seq1 > seq2 ? seq1 - seq2 : seq2 - seq1;
+}
+
+void ARTPSource::processRTPPacket(const sp<ABuffer> &buffer) {
+    if (queuePacket(buffer) && mNumTimes == 2 && mAssembler != NULL) {
+        mAssembler->onPacketReceived(this);
+    }
+
+    dump();
+}
+
+void ARTPSource::timeUpdate(uint32_t rtpTime, uint64_t ntpTime) {
+#if VERBOSE
+    LOG(VERBOSE) << "timeUpdate";
+#endif
+
+    if (mNumTimes == 2) {
+        mNTPTime[0] = mNTPTime[1];
+        mRTPTime[0] = mRTPTime[1];
+        mNumTimes = 1;
+    }
+    mNTPTime[mNumTimes] = ntpTime;
+    mRTPTime[mNumTimes++] = rtpTime;
+
+    if (mNumTimes == 2) {
+        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));
+        }
+    }
+}
+
+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);
+        return true;
+    }
+
+    // Only the lower 16-bit of the sequence numbers are transmitted,
+    // derive the high-order bits by choosing the candidate closest
+    // to the highest sequence number (extended to 32 bits) received so far.
+
+    uint32_t seq1 = seqNum | (mHighestSeqNumber & 0xffff0000);
+    uint32_t seq2 = seqNum | ((mHighestSeqNumber & 0xffff0000) + 0x10000);
+    uint32_t seq3 = seqNum | ((mHighestSeqNumber & 0xffff0000) - 0x10000);
+    uint32_t diff1 = AbsDiff(seq1, mHighestSeqNumber);
+    uint32_t diff2 = AbsDiff(seq2, mHighestSeqNumber);
+    uint32_t diff3 = AbsDiff(seq3, mHighestSeqNumber);
+
+    if (diff1 < diff2) {
+        if (diff1 < diff3) {
+            // diff1 < diff2 ^ diff1 < diff3
+            seqNum = seq1;
+        } else {
+            // diff3 <= diff1 < diff2
+            seqNum = seq3;
+        }
+    } else if (diff2 < diff3) {
+        // diff2 <= diff1 ^ diff2 < diff3
+        seqNum = seq2;
+    } else {
+        // diff3 <= diff2 <= diff1
+        seqNum = seq3;
+    }
+
+    if (seqNum > mHighestSeqNumber) {
+        mHighestSeqNumber = seqNum;
+    }
+
+    buffer->setInt32Data(seqNum);
+
+    List<sp<ABuffer> >::iterator it = mQueue.begin();
+    while (it != mQueue.end() && (uint32_t)(*it)->int32Data() < seqNum) {
+        ++it;
+    }
+
+    if (it != mQueue.end() && (uint32_t)(*it)->int32Data() == seqNum) {
+        LOG(WARNING) << "Discarding duplicate buffer";
+        return false;
+    }
+
+    mQueue.insert(it, buffer);
+
+    return true;
+}
+
+void ARTPSource::dump() const {
+    if ((mNumBuffersReceived % 128) != 0) {
+        return;
+    }
+
+#if 0
+    if (mAssembler == NULL) {
+        char tmp[20];
+        sprintf(tmp, "0x%08x", mID);
+
+        int32_t numMissing = 0;
+
+        if (!mQueue.empty()) {
+            List<sp<ABuffer> >::const_iterator it = mQueue.begin();
+            uint32_t expectedSeqNum = (uint32_t)(*it)->int32Data();
+            ++expectedSeqNum;
+            ++it;
+
+            for (; it != mQueue.end(); ++it) {
+                uint32_t seqNum = (uint32_t)(*it)->int32Data();
+                CHECK_GE(seqNum, expectedSeqNum);
+
+                if (seqNum != expectedSeqNum) {
+                    numMissing += seqNum - expectedSeqNum;
+                    expectedSeqNum = seqNum;
+                }
+
+                ++expectedSeqNum;
+            }
+        }
+
+        LOG(VERBOSE) << "[" << tmp << "] Missing " << numMissing
+             << " / " << (mNumBuffersReceived + numMissing) << " packets. ("
+             << (100.0 * numMissing / (mNumBuffersReceived + numMissing))
+             << " %%)";
+    }
+#endif
+
+#if 0
+    AString out;
+    
+    out.append(tmp);
+    out.append(" [");
+
+    List<sp<ABuffer> >::const_iterator it = mQueue.begin();
+    while (it != mQueue.end()) {
+        uint32_t start = (uint32_t)(*it)->int32Data();
+
+        out.append(start);
+
+        ++it;
+        uint32_t expected = start + 1;
+
+        while (it != mQueue.end()) {
+            uint32_t seqNum = (uint32_t)(*it)->int32Data();
+
+            if (seqNum != expected) {
+                if (expected > start + 1) {
+                    out.append("-");
+                    out.append(expected - 1);
+                }
+                out.append(", ");
+                break;
+            }
+
+            ++it;
+            ++expected;
+        }
+
+        if (it == mQueue.end()) {
+            if (expected > start + 1) {
+                out.append("-");
+                out.append(expected - 1);
+            }
+        }
+    }
+
+    out.append("]");
+
+    LOG(VERBOSE) << out;
+#endif
+}
+
+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]);
+}
+
+}  // namespace android
+
+
diff --git a/media/libstagefright/rtsp/ARTPSource.h b/media/libstagefright/rtsp/ARTPSource.h
new file mode 100644
index 0000000..b93cd56
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTPSource.h
@@ -0,0 +1,67 @@
+/*
+ * 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 A_RTP_SOURCE_H_
+
+#define A_RTP_SOURCE_H_
+
+#include <stdint.h>
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/List.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct ABuffer;
+struct AMessage;
+struct ARTPAssembler;
+struct ASessionDescription;
+
+struct ARTPSource : public RefBase {
+    ARTPSource(
+            uint32_t id,
+            const sp<ASessionDescription> &sessionDesc, size_t index,
+            const sp<AMessage> &notify);
+
+    void processRTPPacket(const sp<ABuffer> &buffer);
+    void timeUpdate(uint32_t rtpTime, uint64_t ntpTime);
+
+    List<sp<ABuffer> > *queue() { return &mQueue; }
+
+private:
+    uint32_t mID;
+    uint32_t mHighestSeqNumber;
+    int32_t mNumBuffersReceived;
+
+    List<sp<ABuffer> > mQueue;
+    sp<ARTPAssembler> mAssembler;
+
+    size_t mNumTimes;
+    uint64_t mNTPTime[2];
+    uint32_t mRTPTime[2];
+
+    uint64_t RTP2NTP(uint32_t rtpTime) const;
+
+    bool queuePacket(const sp<ABuffer> &buffer);
+    void dump() const;
+
+    DISALLOW_EVIL_CONSTRUCTORS(ARTPSource);
+};
+
+}  // namespace android
+
+#endif  // A_RTP_SOURCE_H_
diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp
new file mode 100644
index 0000000..e9162c0
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTSPConnection.cpp
@@ -0,0 +1,549 @@
+/*
+ * 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.
+ */
+
+#include "ARTSPConnection.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include <arpa/inet.h>
+#include <fcntl.h>
+#include <netdb.h>
+#include <sys/socket.h>
+
+namespace android {
+
+// static
+const int64_t ARTSPConnection::kSelectTimeoutUs = 1000ll;
+
+ARTSPConnection::ARTSPConnection()
+    : mState(DISCONNECTED),
+      mSocket(-1),
+      mConnectionID(0),
+      mNextCSeq(0),
+      mReceiveResponseEventPending(false) {
+}
+
+ARTSPConnection::~ARTSPConnection() {
+    if (mSocket >= 0) {
+        LOG(ERROR) << "Connection is still open, closing the socket.";
+        close(mSocket);
+        mSocket = -1;
+    }
+}
+
+void ARTSPConnection::connect(const char *url, const sp<AMessage> &reply) {
+    sp<AMessage> msg = new AMessage(kWhatConnect, id());
+    msg->setString("url", url);
+    msg->setMessage("reply", reply);
+    msg->post();
+}
+
+void ARTSPConnection::disconnect(const sp<AMessage> &reply) {
+    sp<AMessage> msg = new AMessage(kWhatDisconnect, id());
+    msg->setMessage("reply", reply);
+    msg->post();
+}
+
+void ARTSPConnection::sendRequest(
+        const char *request, const sp<AMessage> &reply) {
+    sp<AMessage> msg = new AMessage(kWhatSendRequest, id());
+    msg->setString("request", request);
+    msg->setMessage("reply", reply);
+    msg->post();
+}
+
+void ARTSPConnection::onMessageReceived(const sp<AMessage> &msg) {
+    switch (msg->what()) {
+        case kWhatConnect:
+            onConnect(msg);
+            break;
+
+        case kWhatDisconnect:
+            onDisconnect(msg);
+            break;
+
+        case kWhatCompleteConnection:
+            onCompleteConnection(msg);
+            break;
+
+        case kWhatSendRequest:
+            onSendRequest(msg);
+            break;
+
+        case kWhatReceiveResponse:
+            onReceiveResponse();
+            break;
+
+        default:
+            TRESPASS();
+            break;
+    }
+}
+
+// static
+bool ARTSPConnection::ParseURL(
+        const char *url, AString *host, unsigned *port, AString *path) {
+    host->clear();
+    *port = 0;
+    path->clear();
+
+    if (strncasecmp("rtsp://", url, 7)) {
+        return false;
+    }
+
+    const char *slashPos = strchr(&url[7], '/');
+
+    if (slashPos == NULL) {
+        host->setTo(&url[7]);
+        path->setTo("/");
+    } else {
+        host->setTo(&url[7], slashPos - &url[7]);
+        path->setTo(slashPos);
+    }
+
+    char *colonPos = strchr(host->c_str(), ':');
+
+    if (colonPos != NULL) {
+        unsigned long x;
+        if (!ParseSingleUnsignedLong(colonPos + 1, &x) || x >= 65536) {
+            return false;
+        }
+
+        *port = x;
+
+        size_t colonOffset = colonPos - host->c_str();
+        size_t trailing = host->size() - colonOffset;
+        host->erase(colonOffset, trailing);
+    } else {
+        *port = 554;
+    }
+
+    return true;
+}
+
+void ARTSPConnection::onConnect(const sp<AMessage> &msg) {
+    ++mConnectionID;
+
+    if (mState != DISCONNECTED) {
+        close(mSocket);
+        mSocket = -1;
+
+        flushPendingRequests();
+    }
+
+    mState = CONNECTING;
+
+    mSocket = socket(AF_INET, SOCK_STREAM, 0);
+
+    // Make socket non-blocking.
+    int flags = fcntl(mSocket, F_GETFL, 0);
+    CHECK_NE(flags, -1);
+    CHECK_NE(fcntl(mSocket, F_SETFL, flags | O_NONBLOCK), -1);
+
+    AString url;
+    CHECK(msg->findString("url", &url));
+
+    AString host, path;
+    unsigned port;
+    CHECK(ParseURL(url.c_str(), &host, &port, &path));
+
+    struct hostent *ent = gethostbyname(host.c_str());
+    CHECK(ent != NULL);
+
+    struct sockaddr_in remote;
+    memset(remote.sin_zero, 0, sizeof(remote.sin_zero));
+    remote.sin_family = AF_INET;
+    remote.sin_addr.s_addr = *(in_addr_t *)ent->h_addr;
+    remote.sin_port = htons(port);
+
+    int err = ::connect(
+            mSocket, (const struct sockaddr *)&remote, sizeof(remote));
+
+    sp<AMessage> reply;
+    CHECK(msg->findMessage("reply", &reply));
+
+    reply->setInt32("server-ip", ntohl(remote.sin_addr.s_addr));
+
+    if (err < 0) {
+        if (errno == EINPROGRESS) {
+            sp<AMessage> msg = new AMessage(kWhatCompleteConnection, id());
+            msg->setMessage("reply", reply);
+            msg->setInt32("connection-id", mConnectionID);
+            msg->post();
+            return;
+        }
+
+        reply->setInt32("result", -errno);
+        mState = DISCONNECTED;
+
+        close(mSocket);
+        mSocket = -1;
+    } else {
+        reply->setInt32("result", OK);
+        mState = CONNECTED;
+        mNextCSeq = 1;
+
+        postReceiveReponseEvent();
+    }
+
+    reply->post();
+}
+
+void ARTSPConnection::onDisconnect(const sp<AMessage> &msg) {
+    if (mState == CONNECTED || mState == CONNECTING) {
+        close(mSocket);
+        mSocket = -1;
+
+        flushPendingRequests();
+    } 
+
+    sp<AMessage> reply;
+    CHECK(msg->findMessage("reply", &reply));
+
+    reply->setInt32("result", OK);
+    mState = DISCONNECTED;
+
+    reply->post();
+}
+
+void ARTSPConnection::onCompleteConnection(const sp<AMessage> &msg) {
+    sp<AMessage> reply;
+    CHECK(msg->findMessage("reply", &reply));
+
+    int32_t connectionID;
+    CHECK(msg->findInt32("connection-id", &connectionID));
+
+    if ((connectionID != mConnectionID) || mState != CONNECTING) {
+        // While we were attempting to connect, the attempt was
+        // cancelled.
+        reply->setInt32("result", -ECONNABORTED);
+        reply->post();
+        return;
+    }
+
+    struct timeval tv;
+    tv.tv_sec = 0;
+    tv.tv_usec = kSelectTimeoutUs;
+
+    fd_set ws;
+    FD_ZERO(&ws);
+    FD_SET(mSocket, &ws);
+
+    int res = select(mSocket + 1, NULL, &ws, NULL, &tv);
+    CHECK_GE(res, 0);
+
+    if (res == 0) {
+        // Timed out. Not yet connected.
+
+        msg->post();
+        return;
+    }
+
+    int err;
+    socklen_t optionLen = sizeof(err);
+    CHECK_EQ(getsockopt(mSocket, SOL_SOCKET, SO_ERROR, &err, &optionLen), 0);
+    CHECK_EQ(optionLen, (socklen_t)sizeof(err));
+
+    if (err != 0) {
+        LOG(ERROR) << "err = " << err << " (" << strerror(err) << ")";
+
+        reply->setInt32("result", -err);
+
+        mState = DISCONNECTED;
+        close(mSocket);
+        mSocket = -1;
+    } else {
+        reply->setInt32("result", OK);
+        mState = CONNECTED;
+        mNextCSeq = 1;
+
+        postReceiveReponseEvent();
+    }
+
+    reply->post();
+}
+
+void ARTSPConnection::onSendRequest(const sp<AMessage> &msg) {
+    sp<AMessage> reply;
+    CHECK(msg->findMessage("reply", &reply));
+
+    if (mState != CONNECTED) {
+        reply->setInt32("result", -ENOTCONN);
+        reply->post();
+        return;
+    }
+
+    AString request;
+    CHECK(msg->findString("request", &request));
+
+    // Find the boundary between headers and the body.
+    ssize_t i = request.find("\r\n\r\n");
+    CHECK_GE(i, 0);
+
+    int32_t cseq = mNextCSeq++;
+
+    AString cseqHeader = "CSeq: ";
+    cseqHeader.append(cseq);
+    cseqHeader.append("\r\n");
+
+    request.insert(cseqHeader, i + 2);
+
+    LOG(VERBOSE) << request;
+
+    size_t numBytesSent = 0;
+    while (numBytesSent < request.size()) {
+        ssize_t n =
+            send(mSocket, request.c_str() + numBytesSent,
+                 request.size() - numBytesSent, 0);
+
+        if (n == 0) {
+            // Server closed the connection.
+            TRESPASS();
+        } else if (n < 0) {
+            if (errno == EINTR) {
+                continue;
+            }
+
+            TRESPASS();
+        }
+
+        numBytesSent += (size_t)n;
+    }
+
+    mPendingRequests.add(cseq, reply);
+}
+
+void ARTSPConnection::onReceiveResponse() {
+    mReceiveResponseEventPending = false;
+
+    if (mState != CONNECTED) {
+        return;
+    }
+
+    struct timeval tv;
+    tv.tv_sec = 0;
+    tv.tv_usec = kSelectTimeoutUs;
+
+    fd_set rs;
+    FD_ZERO(&rs);
+    FD_SET(mSocket, &rs);
+
+    int res = select(mSocket + 1, &rs, NULL, NULL, &tv);
+    CHECK_GE(res, 0);
+
+    if (res == 1) {
+        if (!receiveRTSPReponse()) {
+            // Something horrible, irreparable has happened.
+            flushPendingRequests();
+            return;
+        }
+    }
+
+    postReceiveReponseEvent();
+}
+
+void ARTSPConnection::flushPendingRequests() {
+    for (size_t i = 0; i < mPendingRequests.size(); ++i) {
+        sp<AMessage> reply = mPendingRequests.valueAt(i);
+
+        reply->setInt32("result", -ECONNABORTED);
+        reply->post();
+    }
+
+    mPendingRequests.clear();
+}
+
+void ARTSPConnection::postReceiveReponseEvent() {
+    if (mReceiveResponseEventPending) {
+        return;
+    }
+
+    sp<AMessage> msg = new AMessage(kWhatReceiveResponse, id());
+    msg->post();
+
+    mReceiveResponseEventPending = true;
+}
+
+bool ARTSPConnection::receiveLine(AString *line) {
+    line->clear();
+
+    bool sawCR = false;
+    for (;;) {
+        char c;
+        ssize_t n = recv(mSocket, &c, 1, 0);
+        if (n == 0) {
+            // Server closed the connection.
+            return false;
+        } else if (n < 0) {
+            if (errno == EINTR) {
+                continue;
+            }
+
+            TRESPASS();
+        }
+
+        if (sawCR && c == '\n') {
+            line->erase(line->size() - 1, 1);
+            return true;
+        }
+
+        line->append(&c, 1);
+
+        sawCR = (c == '\r');
+    }
+}
+
+bool ARTSPConnection::receiveRTSPReponse() {
+    sp<ARTSPResponse> response = new ARTSPResponse;
+
+    if (!receiveLine(&response->mStatusLine)) {
+        return false;
+    }
+
+    LOG(INFO) << "status: " << response->mStatusLine;
+
+    ssize_t space1 = response->mStatusLine.find(" ");
+    if (space1 < 0) {
+        return false;
+    }
+    ssize_t space2 = response->mStatusLine.find(" ", space1 + 1);
+    if (space2 < 0) {
+        return false;
+    }
+
+    AString statusCodeStr(
+            response->mStatusLine, space1 + 1, space2 - space1 - 1);
+
+    if (!ParseSingleUnsignedLong(
+                statusCodeStr.c_str(), &response->mStatusCode)
+            || response->mStatusCode < 100 || response->mStatusCode > 999) {
+        return false;
+    }
+
+    AString line;
+    for (;;) {
+        if (!receiveLine(&line)) {
+            break;
+        }
+
+        if (line.empty()) {
+            break;
+        }
+
+        LOG(VERBOSE) << "line: " << line;
+
+        ssize_t colonPos = line.find(":");
+        if (colonPos < 0) {
+            // Malformed header line.
+            return false;
+        }
+
+        AString key(line, 0, colonPos);
+        key.trim();
+        key.tolower();
+
+        line.erase(0, colonPos + 1);
+        line.trim();
+
+        response->mHeaders.add(key, line);
+    }
+
+    unsigned long contentLength = 0;
+
+    ssize_t i = response->mHeaders.indexOfKey("content-length");
+
+    if (i >= 0) {
+        AString value = response->mHeaders.valueAt(i);
+        if (!ParseSingleUnsignedLong(value.c_str(), &contentLength)) {
+            return false;
+        }
+    }
+
+    if (contentLength > 0) {
+        response->mContent = new ABuffer(contentLength);
+
+        size_t numBytesRead = 0;
+        while (numBytesRead < contentLength) {
+            ssize_t n = recv(
+                    mSocket, response->mContent->data() + numBytesRead,
+                    contentLength - numBytesRead, 0);
+
+            if (n == 0) {
+                // Server closed the connection.
+                TRESPASS();
+            } else if (n < 0) {
+                if (errno == EINTR) {
+                    continue;
+                }
+
+                TRESPASS();
+            }
+
+            numBytesRead += (size_t)n;
+        }
+    }
+
+    return notifyResponseListener(response);
+}
+
+// static
+bool ARTSPConnection::ParseSingleUnsignedLong(
+        const char *from, unsigned long *x) {
+    char *end;
+    *x = strtoul(from, &end, 10);
+
+    if (end == from || *end != '\0') {
+        return false;
+    }
+
+    return true;
+}
+
+bool ARTSPConnection::notifyResponseListener(
+        const sp<ARTSPResponse> &response) {
+    ssize_t i = response->mHeaders.indexOfKey("cseq");
+
+    if (i < 0) {
+        return true;
+    }
+
+    AString value = response->mHeaders.valueAt(i);
+
+    unsigned long cseq;
+    if (!ParseSingleUnsignedLong(value.c_str(), &cseq)) {
+        return false;
+    }
+
+    i = mPendingRequests.indexOfKey(cseq);
+
+    if (i < 0) {
+        // Unsolicited response?
+        TRESPASS();
+    }
+
+    sp<AMessage> reply = mPendingRequests.valueAt(i);
+    mPendingRequests.removeItemsAt(i);
+
+    reply->setInt32("result", OK);
+    reply->setObject("response", response);
+    reply->post();
+
+    return true;
+}
+
+}  // namespace android
diff --git a/media/libstagefright/rtsp/ARTSPConnection.h b/media/libstagefright/rtsp/ARTSPConnection.h
new file mode 100644
index 0000000..3577a2f
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTSPConnection.h
@@ -0,0 +1,97 @@
+/*
+ * 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 A_RTSP_CONNECTION_H_
+
+#define A_RTSP_CONNECTION_H_
+
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/foundation/AString.h>
+
+namespace android {
+
+struct ABuffer;
+
+struct ARTSPResponse : public RefBase {
+    unsigned long mStatusCode;
+    AString mStatusLine;
+    KeyedVector<AString,AString> mHeaders;
+    sp<ABuffer> mContent;
+};
+
+struct ARTSPConnection : public AHandler {
+    ARTSPConnection();
+
+    void connect(const char *url, const sp<AMessage> &reply);
+    void disconnect(const sp<AMessage> &reply);
+
+    void sendRequest(const char *request, const sp<AMessage> &reply);
+
+protected:
+    virtual ~ARTSPConnection();
+    virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+    enum State {
+        DISCONNECTED,
+        CONNECTING,
+        CONNECTED,
+    };
+
+    enum {
+        kWhatConnect            = 'conn',
+        kWhatDisconnect         = 'disc',
+        kWhatCompleteConnection = 'comc',
+        kWhatSendRequest        = 'sreq',
+        kWhatReceiveResponse    = 'rres',
+    };
+
+    static const int64_t kSelectTimeoutUs;
+
+    State mState;
+    int mSocket;
+    int32_t mConnectionID;
+    int32_t mNextCSeq;
+    bool mReceiveResponseEventPending;
+
+    KeyedVector<int32_t, sp<AMessage> > mPendingRequests;
+
+    void onConnect(const sp<AMessage> &msg);
+    void onDisconnect(const sp<AMessage> &msg);
+    void onCompleteConnection(const sp<AMessage> &msg);
+    void onSendRequest(const sp<AMessage> &msg);
+    void onReceiveResponse();
+
+    void flushPendingRequests();
+    void postReceiveReponseEvent();
+
+    // Return false iff something went unrecoverably wrong.
+    bool receiveRTSPReponse();
+    bool receiveLine(AString *line);
+    bool notifyResponseListener(const sp<ARTSPResponse> &response);
+
+    static bool ParseURL(
+            const char *url, AString *host, unsigned *port, AString *path);
+
+    static bool ParseSingleUnsignedLong(
+            const char *from, unsigned long *x);
+
+    DISALLOW_EVIL_CONSTRUCTORS(ARTSPConnection);
+};
+
+}  // namespace android
+
+#endif  // A_RTSP_CONNECTION_H_
diff --git a/media/libstagefright/rtsp/ARTSPController.cpp b/media/libstagefright/rtsp/ARTSPController.cpp
new file mode 100644
index 0000000..7b87d42
--- /dev/null
+++ b/media/libstagefright/rtsp/ARTSPController.cpp
@@ -0,0 +1,75 @@
+/*
+ * 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.
+ */
+
+#include "ARTSPController.h"
+
+#include "MyHandler.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+
+namespace android {
+
+ARTSPController::ARTSPController(const sp<ALooper> &looper)
+    : mLooper(looper) {
+}
+
+ARTSPController::~ARTSPController() {
+}
+
+status_t ARTSPController::connect(const char *url) {
+    if (mHandler != NULL) {
+        return ERROR_ALREADY_CONNECTED;
+    }
+
+    mHandler = new MyHandler(url, mLooper);
+    sleep(10);
+
+    return OK;
+}
+
+void ARTSPController::disconnect() {
+    if (mHandler == NULL) {
+        return;
+    }
+
+    mHandler.clear();
+}
+
+size_t ARTSPController::countTracks() {
+    if (mHandler == NULL) {
+        return 0;
+    }
+
+    return mHandler->countTracks();
+}
+
+sp<MediaSource> ARTSPController::getTrack(size_t index) {
+    CHECK(mHandler != NULL);
+
+    return mHandler->getPacketSource(index);
+}
+
+sp<MetaData> ARTSPController::getTrackMetaData(
+        size_t index, uint32_t flags) {
+    CHECK(mHandler != NULL);
+
+    return mHandler->getPacketSource(index)->getFormat();
+}
+
+}  // namespace android
diff --git a/media/libstagefright/rtsp/ASessionDescription.cpp b/media/libstagefright/rtsp/ASessionDescription.cpp
new file mode 100644
index 0000000..ca4c55e
--- /dev/null
+++ b/media/libstagefright/rtsp/ASessionDescription.cpp
@@ -0,0 +1,270 @@
+/*
+ * 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.
+ */
+
+#include "ASessionDescription.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AString.h>
+
+#include <stdlib.h>
+
+namespace android {
+
+ASessionDescription::ASessionDescription()
+    : mIsValid(false) {
+}
+
+ASessionDescription::~ASessionDescription() {
+}
+
+bool ASessionDescription::setTo(const void *data, size_t size) {
+    mIsValid = parse(data, size);
+
+    if (!mIsValid) {
+        mTracks.clear();
+        mFormats.clear();
+    }
+
+    return mIsValid;
+}
+
+bool ASessionDescription::parse(const void *data, size_t size) {
+    mTracks.clear();
+    mFormats.clear();
+
+    mTracks.push(Attribs());
+    mFormats.push(AString("[root]"));
+
+    AString desc((const char *)data, size);
+    LOG(VERBOSE) << desc;
+
+    size_t i = 0;
+    for (;;) {
+        ssize_t eolPos = desc.find("\r\n", i);
+        if (eolPos < 0) {
+            break;
+        }
+
+        AString line(desc, i, eolPos - i);
+
+        if (line.size() < 2 || line.c_str()[1] != '=') {
+            return false;
+        }
+
+        switch (line.c_str()[0]) {
+            case 'v':
+            {
+                if (strcmp(line.c_str(), "v=0")) {
+                    return false;
+                }
+                break;
+            }
+
+            case 'a':
+            case 'b':
+            {
+                AString key, value;
+
+                ssize_t colonPos = line.find(":", 2);
+                if (colonPos < 0) {
+                    key = line;
+                } else {
+                    key.setTo(line, 0, colonPos);
+
+                    if (key == "a=fmtp" || key == "a=rtpmap"
+                            || key == "a=framesize") {
+                        ssize_t spacePos = line.find(" ", colonPos + 1);
+                        if (spacePos < 0) {
+                            return false;
+                        }
+
+                        key.setTo(line, 0, spacePos);
+
+                        colonPos = spacePos;
+                    }
+
+                    value.setTo(line, colonPos + 1, line.size() - colonPos - 1);
+                }
+
+                key.trim();
+                value.trim();
+
+                LOG(VERBOSE) << "adding '" << key << "' => '" << value << "'";
+
+                mTracks.editItemAt(mTracks.size() - 1).add(key, value);
+                break;
+            }
+
+            case 'm':
+            {
+                LOG(VERBOSE) << "new section '" << AString(line, 2, line.size() - 2) << "'";
+
+                mTracks.push(Attribs());
+                mFormats.push(AString(line, 2, line.size() - 2));
+                break;
+            }
+        }
+
+        i = eolPos + 2;
+    }
+
+    return true;
+}
+
+bool ASessionDescription::isValid() const {
+    return mIsValid;
+}
+
+size_t ASessionDescription::countTracks() const {
+    return mTracks.size();
+}
+
+void ASessionDescription::getFormat(size_t index, AString *value) const {
+    CHECK_GE(index, 0u);
+    CHECK_LT(index, mTracks.size());
+
+    *value = mFormats.itemAt(index);
+}
+
+bool ASessionDescription::findAttribute(
+        size_t index, const char *key, AString *value) const {
+    CHECK_GE(index, 0u);
+    CHECK_LT(index, mTracks.size());
+
+    value->clear();
+
+    const Attribs &track = mTracks.itemAt(index);
+    ssize_t i = track.indexOfKey(AString(key));
+
+    if (i < 0) {
+        return false;
+    }
+
+    *value = track.valueAt(i);
+
+    return true;
+}
+
+void ASessionDescription::getFormatType(
+        size_t index, unsigned long *PT,
+        AString *desc, AString *params) const {
+    AString format;
+    getFormat(index, &format);
+
+    char *lastSpacePos = strrchr(format.c_str(), ' ');
+    CHECK(lastSpacePos != NULL);
+
+    char *end;
+    unsigned long x = strtoul(lastSpacePos + 1, &end, 10);
+    CHECK_GT(end, lastSpacePos + 1);
+    CHECK_EQ(*end, '\0');
+
+    *PT = x;
+
+    char key[20];
+    sprintf(key, "a=rtpmap:%lu", x);
+
+    CHECK(findAttribute(index, key, desc));
+
+    sprintf(key, "a=fmtp:%lu", x);
+    if (!findAttribute(index, key, params)) {
+        params->clear();
+    }
+}
+
+void ASessionDescription::getDimensions(
+        size_t index, unsigned long PT,
+        int32_t *width, int32_t *height) const {
+    char key[20];
+    sprintf(key, "a=framesize:%lu", PT);
+    AString value;
+    CHECK(findAttribute(index, key, &value));
+
+    const char *s = value.c_str();
+    char *end;
+    *width = strtoul(s, &end, 10);
+    CHECK_GT(end, s);
+    CHECK_EQ(*end, '-');
+
+    s = end + 1;
+    *height = strtoul(s, &end, 10);
+    CHECK_GT(end, s);
+    CHECK_EQ(*end, '\0');
+}
+
+bool ASessionDescription::getDurationUs(int64_t *durationUs) const {
+    *durationUs = 0;
+
+    CHECK(mIsValid);
+
+    AString value;
+    if (!findAttribute(0, "a=range", &value)) {
+        return false;
+    }
+
+    if (value == "npt=now-") {
+        return false;
+    }
+
+    if (strncmp(value.c_str(), "npt=", 4)) {
+        return false;
+    }
+
+    const char *s = value.c_str() + 4;
+    char *end;
+    double from = strtod(s, &end);
+    CHECK_GT(end, s);
+    CHECK_EQ(*end, '-');
+
+    s = end + 1;
+    double to = strtod(s, &end);
+    CHECK_GT(end, s);
+    CHECK_EQ(*end, '\0');
+
+    CHECK_GE(to, from);
+
+    *durationUs = (int64_t)((to - from) * 1E6);
+
+    return true;
+}
+
+// static
+void ASessionDescription::ParseFormatDesc(
+        const char *desc, int32_t *timescale, int32_t *numChannels) {
+    const char *slash1 = strchr(desc, '/');
+    CHECK(slash1 != NULL);
+
+    const char *s = slash1 + 1;
+    char *end;
+    unsigned long x = strtoul(s, &end, 10);
+    CHECK_GT(end, s);
+    CHECK(*end == '\0' || *end == '/');
+
+    *timescale = x;
+    *numChannels = 1;
+
+    if (*end == '/') {
+        s = end + 1;
+        unsigned long x = strtoul(s, &end, 10);
+        CHECK_GT(end, s);
+        CHECK_EQ(*end, '\0');
+
+        *numChannels = x;
+    }
+}
+
+}  // namespace android
+
diff --git a/media/libstagefright/rtsp/ASessionDescription.h b/media/libstagefright/rtsp/ASessionDescription.h
new file mode 100644
index 0000000..b26980f
--- /dev/null
+++ b/media/libstagefright/rtsp/ASessionDescription.h
@@ -0,0 +1,75 @@
+/*
+ * 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 A_SESSION_DESCRIPTION_H_
+
+#define A_SESSION_DESCRIPTION_H_
+
+#include <sys/types.h>
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+struct AString;
+
+struct ASessionDescription : public RefBase {
+    ASessionDescription();
+
+    bool setTo(const void *data, size_t size);
+    bool isValid() const;
+
+    // Actually, 1 + number of tracks, as index 0 is reserved for the
+    // session description root-level attributes.
+    size_t countTracks() const;
+    void getFormat(size_t index, AString *value) const;
+
+    void getFormatType(
+            size_t index, unsigned long *PT,
+            AString *desc, AString *params) const;
+
+    void getDimensions(
+            size_t index, unsigned long PT,
+            int32_t *width, int32_t *height) const;
+
+    bool getDurationUs(int64_t *durationUs) const;
+
+    static void ParseFormatDesc(
+            const char *desc, int32_t *timescale, int32_t *numChannels);
+
+    bool findAttribute(size_t index, const char *key, AString *value) const;
+
+protected:
+    virtual ~ASessionDescription();
+
+private:
+    typedef KeyedVector<AString,AString> Attribs;
+
+    bool mIsValid;
+    Vector<Attribs> mTracks;
+    Vector<AString> mFormats;
+
+    bool parse(const void *data, size_t size);
+
+    DISALLOW_EVIL_CONSTRUCTORS(ASessionDescription);
+};
+
+}  // namespace android
+
+#endif  // A_SESSION_DESCRIPTION_H_
diff --git a/media/libstagefright/rtsp/Android.mk b/media/libstagefright/rtsp/Android.mk
new file mode 100644
index 0000000..4608fa0
--- /dev/null
+++ b/media/libstagefright/rtsp/Android.mk
@@ -0,0 +1,28 @@
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:=       \
+        ARTSPController.cpp         \
+        AAVCAssembler.cpp           \
+        AMPEG4AudioAssembler.cpp    \
+        APacketSource.cpp           \
+        ARTPAssembler.cpp           \
+        ARTPConnection.cpp          \
+        ARTPSource.cpp              \
+        ARTSPConnection.cpp         \
+        ASessionDescription.cpp     \
+
+LOCAL_C_INCLUDES:= \
+	$(JNI_H_INCLUDE) \
+	$(TOP)/external/opencore/extern_libs_v2/khronos/openmax/include \
+        $(TOP)/frameworks/base/media/libstagefright/include \
+
+LOCAL_MODULE:= libstagefright_rtsp
+
+ifeq ($(TARGET_ARCH),arm)
+    LOCAL_CFLAGS += -Wno-psabi
+endif
+
+include $(BUILD_STATIC_LIBRARY)
+
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
new file mode 100644
index 0000000..74bb798
--- /dev/null
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -0,0 +1,442 @@
+/*
+ * 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 MY_HANDLER_H_
+
+#define MY_HANDLER_H_
+
+#include "APacketSource.h"
+#include "ARTPConnection.h"
+#include "ARTSPConnection.h"
+#include "ASessionDescription.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaErrors.h>
+
+namespace android {
+
+struct MyHandler : public AHandler {
+    MyHandler(const char *url, const sp<ALooper> &looper)
+        : mLooper(looper),
+          mConn(new ARTSPConnection),
+          mRTPConn(new ARTPConnection),
+          mSessionURL(url),
+          mSetupTracksSuccessful(false),
+          mFirstAccessUnit(true),
+          mFirstAccessUnitNTP(-1) {
+        mLooper->registerHandler(this);
+        mLooper->registerHandler(mConn);
+        mLooper->registerHandler(mRTPConn);
+        sp<AMessage> reply = new AMessage('conn', id());
+        mConn->connect(mSessionURL.c_str(), reply);
+    }
+
+    virtual void onMessageReceived(const sp<AMessage> &msg) {
+        switch (msg->what()) {
+            case 'conn':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "connection request completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                if (result == OK) {
+                    AString request;
+                    request = "DESCRIBE ";
+                    request.append(mSessionURL);
+                    request.append(" RTSP/1.0\r\n");
+                    request.append("Accept: application/sdp\r\n");
+                    request.append("\r\n");
+
+                    sp<AMessage> reply = new AMessage('desc', id());
+                    mConn->sendRequest(request.c_str(), reply);
+                }
+                break;
+            }
+
+            case 'disc':
+            {
+                LOG(INFO) << "disconnect completed";
+
+                (new AMessage('quit', id()))->post();
+                break;
+            }
+
+            case 'desc':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "DESCRIBE completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                if (result == OK) {
+                    sp<RefBase> obj;
+                    CHECK(msg->findObject("response", &obj));
+                    sp<ARTSPResponse> response =
+                        static_cast<ARTSPResponse *>(obj.get());
+
+                    if (response->mStatusCode == 302) {
+                        ssize_t i = response->mHeaders.indexOfKey("location");
+                        CHECK_GE(i, 0);
+
+                        mSessionURL = response->mHeaders.valueAt(i);
+
+                        AString request;
+                        request = "DESCRIBE ";
+                        request.append(mSessionURL);
+                        request.append(" RTSP/1.0\r\n");
+                        request.append("Accept: application/sdp\r\n");
+                        request.append("\r\n");
+
+                        sp<AMessage> reply = new AMessage('desc', id());
+                        mConn->sendRequest(request.c_str(), reply);
+                        break;
+                    }
+
+                    CHECK_EQ(response->mStatusCode, 200u);
+
+                    mSessionDesc = new ASessionDescription;
+
+                    mSessionDesc->setTo(
+                            response->mContent->data(),
+                            response->mContent->size());
+
+                    CHECK(mSessionDesc->isValid());
+
+                    ssize_t i = response->mHeaders.indexOfKey("content-base");
+                    if (i >= 0) {
+                        mBaseURL = response->mHeaders.valueAt(i);
+                    } else {
+                        i = response->mHeaders.indexOfKey("content-location");
+                        if (i >= 0) {
+                            mBaseURL = response->mHeaders.valueAt(i);
+                        } else {
+                            mBaseURL = mSessionURL;
+                        }
+                    }
+
+                    CHECK_GT(mSessionDesc->countTracks(), 1u);
+                    setupTrack(1);
+                } else {
+                    sp<AMessage> reply = new AMessage('disc', id());
+                    mConn->disconnect(reply);
+                }
+                break;
+            }
+
+            case 'setu':
+            {
+                size_t index;
+                CHECK(msg->findSize("index", &index));
+
+                size_t trackIndex;
+                CHECK(msg->findSize("track-index", &trackIndex));
+
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "SETUP(" << index << ") completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                TrackInfo *track = &mTracks.editItemAt(trackIndex);
+
+                if (result == OK) {
+                    sp<RefBase> obj;
+                    CHECK(msg->findObject("response", &obj));
+                    sp<ARTSPResponse> response =
+                        static_cast<ARTSPResponse *>(obj.get());
+
+                    CHECK_EQ(response->mStatusCode, 200u);
+
+                    ssize_t i = response->mHeaders.indexOfKey("session");
+                    CHECK_GE(i, 0);
+
+                    if (index == 1) {
+                        mSessionID = response->mHeaders.valueAt(i);
+                        i = mSessionID.find(";");
+                        if (i >= 0) {
+                            // Remove options, i.e. ";timeout=90"
+                            mSessionID.erase(i, mSessionID.size() - i);
+                        }
+                    }
+
+                    sp<AMessage> notify = new AMessage('accu', id());
+                    notify->setSize("track-index", trackIndex);
+
+                    mRTPConn->addStream(
+                            track->mRTPSocket, track->mRTCPSocket,
+                            mSessionDesc, index,
+                            notify);
+
+                    track->mPacketSource =
+                        new APacketSource(mSessionDesc, index);
+
+                    mSetupTracksSuccessful = true;
+
+                    ++index;
+                    if (index < mSessionDesc->countTracks()) {
+                        setupTrack(index);
+                        break;
+                    }
+                } else {
+                    close(track->mRTPSocket);
+                    close(track->mRTCPSocket);
+
+                    mTracks.removeItemsAt(mTracks.size() - 1);
+                }
+
+                if (mSetupTracksSuccessful) {
+                    AString request = "PLAY ";
+                    request.append(mSessionURL);
+                    request.append(" RTSP/1.0\r\n");
+
+                    request.append("Session: ");
+                    request.append(mSessionID);
+                    request.append("\r\n");
+
+                    request.append("\r\n");
+
+                    sp<AMessage> reply = new AMessage('play', id());
+                    mConn->sendRequest(request.c_str(), reply);
+                } else {
+                    sp<AMessage> reply = new AMessage('disc', id());
+                    mConn->disconnect(reply);
+                }
+                break;
+            }
+
+            case 'play':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "PLAY completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                if (result == OK) {
+                    sp<RefBase> obj;
+                    CHECK(msg->findObject("response", &obj));
+                    sp<ARTSPResponse> response =
+                        static_cast<ARTSPResponse *>(obj.get());
+
+                    CHECK_EQ(response->mStatusCode, 200u);
+
+                    sp<AMessage> msg = new AMessage('abor', id());
+                    msg->post(60000000ll);
+                } else {
+                    sp<AMessage> reply = new AMessage('disc', id());
+                    mConn->disconnect(reply);
+                }
+
+                break;
+            }
+
+            case 'abor':
+            {
+                for (size_t i = 0; i < mTracks.size(); ++i) {
+                    mTracks.editItemAt(i).mPacketSource->signalEOS(
+                            ERROR_END_OF_STREAM);
+                }
+
+                sp<AMessage> reply = new AMessage('tear', id());
+
+                AString request;
+                request = "TEARDOWN ";
+
+                // XXX should use aggregate url from SDP here...
+                request.append(mSessionURL);
+                request.append(" RTSP/1.0\r\n");
+
+                request.append("Session: ");
+                request.append(mSessionID);
+                request.append("\r\n");
+
+                request.append("\r\n");
+
+                mConn->sendRequest(request.c_str(), reply);
+                break;
+            }
+
+            case 'tear':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "TEARDOWN completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                sp<AMessage> reply = new AMessage('disc', id());
+                mConn->disconnect(reply);
+                break;
+            }
+
+            case 'quit':
+            {
+                mLooper->stop();
+                break;
+            }
+
+            case 'accu':
+            {
+                size_t trackIndex;
+                CHECK(msg->findSize("track-index", &trackIndex));
+
+                sp<RefBase> obj;
+                CHECK(msg->findObject("access-unit", &obj));
+
+                sp<ABuffer> accessUnit = static_cast<ABuffer *>(obj.get());
+
+                uint64_t ntpTime;
+                CHECK(accessUnit->meta()->findInt64(
+                            "ntp-time", (int64_t *)&ntpTime));
+
+                if (mFirstAccessUnit) {
+                    mFirstAccessUnit = false;
+                    mFirstAccessUnitNTP = ntpTime;
+                }
+                if (ntpTime > mFirstAccessUnitNTP) {
+                    ntpTime -= mFirstAccessUnitNTP;
+                } else {
+                    ntpTime = 0;
+                }
+
+                accessUnit->meta()->setInt64("ntp-time", ntpTime);
+
+                TrackInfo *track = &mTracks.editItemAt(trackIndex);
+                track->mPacketSource->queueAccessUnit(accessUnit);
+                break;
+            }
+
+            default:
+                TRESPASS();
+                break;
+        }
+    }
+
+    sp<APacketSource> getPacketSource(size_t index) {
+        CHECK_GE(index, 0u);
+        CHECK_LT(index, mTracks.size());
+
+        return mTracks.editItemAt(index).mPacketSource;
+    }
+
+    size_t countTracks() const {
+        return mTracks.size();
+    }
+
+private:
+    sp<ALooper> mLooper;
+    sp<ARTSPConnection> mConn;
+    sp<ARTPConnection> mRTPConn;
+    sp<ASessionDescription> mSessionDesc;
+    AString mSessionURL;
+    AString mBaseURL;
+    AString mSessionID;
+    bool mSetupTracksSuccessful;
+    bool mFirstAccessUnit;
+    uint64_t mFirstAccessUnitNTP;
+
+    struct TrackInfo {
+        int mRTPSocket;
+        int mRTCPSocket;
+
+        sp<APacketSource> mPacketSource;
+    };
+    Vector<TrackInfo> mTracks;
+
+    void setupTrack(size_t index) {
+        AString url;
+        CHECK(mSessionDesc->findAttribute(index, "a=control", &url));
+
+        AString trackURL;
+        CHECK(MakeURL(mBaseURL.c_str(), url.c_str(), &trackURL));
+
+        mTracks.push(TrackInfo());
+        TrackInfo *info = &mTracks.editItemAt(mTracks.size() - 1);
+
+        unsigned rtpPort;
+        ARTPConnection::MakePortPair(
+                &info->mRTPSocket, &info->mRTCPSocket, &rtpPort);
+
+        AString request = "SETUP ";
+        request.append(trackURL);
+        request.append(" RTSP/1.0\r\n");
+
+        request.append("Transport: RTP/AVP/UDP;unicast;client_port=");
+        request.append(rtpPort);
+        request.append("-");
+        request.append(rtpPort + 1);
+        request.append("\r\n");
+
+        if (index > 1) {
+            request.append("Session: ");
+            request.append(mSessionID);
+            request.append("\r\n");
+        }
+
+        request.append("\r\n");
+
+        sp<AMessage> reply = new AMessage('setu', id());
+        reply->setSize("index", index);
+        reply->setSize("track-index", mTracks.size() - 1);
+        mConn->sendRequest(request.c_str(), reply);
+    }
+
+    static bool MakeURL(const char *baseURL, const char *url, AString *out) {
+        out->clear();
+
+        if (strncasecmp("rtsp://", baseURL, 7)) {
+            // Base URL must be absolute
+            return false;
+        }
+
+        if (!strncasecmp("rtsp://", url, 7)) {
+            // "url" is already an absolute URL, ignore base URL.
+            out->setTo(url);
+            return true;
+        }
+
+        size_t n = strlen(baseURL);
+        if (baseURL[n - 1] == '/') {
+            out->setTo(baseURL);
+            out->append(url);
+        } else {
+            char *slashPos = strrchr(baseURL, '/');
+
+            if (slashPos > &baseURL[6]) {
+                out->setTo(baseURL, slashPos - baseURL);
+            } else {
+                out->setTo(baseURL);
+            }
+
+            out->append("/");
+            out->append(url);
+        }
+
+        return true;
+    }
+
+    DISALLOW_EVIL_CONSTRUCTORS(MyHandler);
+};
+
+}  // namespace android
+
+#endif  // MY_HANDLER_H_
diff --git a/media/libstagefright/rtsp/MyTransmitter.h b/media/libstagefright/rtsp/MyTransmitter.h
new file mode 100644
index 0000000..009a3b1
--- /dev/null
+++ b/media/libstagefright/rtsp/MyTransmitter.h
@@ -0,0 +1,981 @@
+/*
+ * 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 MY_TRANSMITTER_H_
+
+#define MY_TRANSMITTER_H_
+
+#include "ARTPConnection.h"
+
+#include <arpa/inet.h>
+#include <sys/socket.h>
+
+#include <openssl/md5.h>
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/base64.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+#ifdef ANDROID
+#include "VideoSource.h"
+
+#include <media/stagefright/OMXClient.h>
+#include <media/stagefright/OMXCodec.h>
+#endif
+
+namespace android {
+
+#define TRACK_SUFFIX    "trackid=1"
+#define PT              96
+#define PT_STR          "96"
+
+#define USERNAME        "bcast"
+#define PASSWORD        "test"
+
+static int uniformRand(int limit) {
+    return ((double)rand() * limit) / RAND_MAX;
+}
+
+static bool GetAttribute(const char *s, const char *key, AString *value) {
+    value->clear();
+
+    size_t keyLen = strlen(key);
+
+    for (;;) {
+        const char *colonPos = strchr(s, ';');
+
+        size_t len =
+            (colonPos == NULL) ? strlen(s) : colonPos - s;
+
+        if (len >= keyLen + 1 && s[keyLen] == '=' && !strncmp(s, key, keyLen)) {
+            value->setTo(&s[keyLen + 1], len - keyLen - 1);
+            return true;
+        }
+
+        if (colonPos == NULL) {
+            return false;
+        }
+
+        s = colonPos + 1;
+    }
+}
+
+struct MyTransmitter : public AHandler {
+    MyTransmitter(const char *url, const sp<ALooper> &looper)
+        : mServerURL(url),
+          mLooper(looper),
+          mConn(new ARTSPConnection),
+          mConnected(false),
+          mAuthType(NONE),
+          mRTPSocket(-1),
+          mRTCPSocket(-1),
+          mSourceID(rand()),
+          mSeqNo(uniformRand(65536)),
+          mRTPTimeBase(rand()),
+          mNumSamplesSent(0),
+          mNumRTPSent(0),
+          mNumRTPOctetsSent(0),
+          mLastRTPTime(0),
+          mLastNTPTime(0) {
+        mStreamURL = mServerURL;
+        mStreamURL.append("/bazong.sdp");
+
+        mTrackURL = mStreamURL;
+        mTrackURL.append("/");
+        mTrackURL.append(TRACK_SUFFIX);
+
+        mLooper->registerHandler(this);
+        mLooper->registerHandler(mConn);
+
+        sp<AMessage> reply = new AMessage('conn', id());
+        mConn->connect(mServerURL.c_str(), reply);
+
+#ifdef ANDROID
+        int width = 640;
+        int height = 480;
+
+        sp<MediaSource> source = new VideoSource(width, height);
+
+        sp<MetaData> encMeta = new MetaData;
+        encMeta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
+        encMeta->setInt32(kKeyWidth, width);
+        encMeta->setInt32(kKeyHeight, height);
+
+        OMXClient client;
+        client.connect();
+
+        mEncoder = OMXCodec::Create(
+                client.interface(), encMeta,
+                true /* createEncoder */, source);
+
+        mEncoder->start();
+
+        MediaBuffer *buffer;
+        CHECK_EQ(mEncoder->read(&buffer), (status_t)OK);
+        CHECK(buffer != NULL);
+
+        makeH264SPropParamSets(buffer);
+
+        buffer->release();
+        buffer = NULL;
+#endif
+    }
+
+    uint64_t ntpTime() {
+        struct timeval tv;
+        gettimeofday(&tv, NULL);
+
+        uint64_t nowUs = tv.tv_sec * 1000000ll + tv.tv_usec;
+
+        nowUs += ((70ll * 365 + 17) * 24) * 60 * 60 * 1000000ll;
+
+        uint64_t hi = nowUs / 1000000ll;
+        uint64_t lo = ((1ll << 32) * (nowUs % 1000000ll)) / 1000000ll;
+
+        return (hi << 32) | lo;
+    }
+
+    void issueAnnounce() {
+        AString sdp;
+        sdp = "v=0\r\n";
+
+        sdp.append("o=- ");
+
+        uint64_t ntp = ntpTime();
+        sdp.append(ntp);
+        sdp.append(" ");
+        sdp.append(ntp);
+        sdp.append(" IN IP4 127.0.0.0\r\n");
+
+        sdp.append(
+              "s=Sample\r\n"
+              "i=Playing around with ANNOUNCE\r\n"
+              "c=IN IP4 ");
+
+        struct in_addr addr;
+        addr.s_addr = htonl(mServerIP);
+
+        sdp.append(inet_ntoa(addr));
+
+        sdp.append(
+              "\r\n"
+              "t=0 0\r\n"
+              "a=range:npt=now-\r\n");
+
+#ifdef ANDROID
+        sp<MetaData> meta = mEncoder->getFormat();
+        int32_t width, height;
+        CHECK(meta->findInt32(kKeyWidth, &width));
+        CHECK(meta->findInt32(kKeyHeight, &height));
+
+        sdp.append(
+              "m=video 0 RTP/AVP " PT_STR "\r\n"
+              "b=AS 320000\r\n"
+              "a=rtpmap:" PT_STR " H264/90000\r\n");
+
+        sdp.append("a=cliprect 0,0,");
+        sdp.append(height);
+        sdp.append(",");
+        sdp.append(width);
+        sdp.append("\r\n");
+
+        sdp.append(
+              "a=framesize:" PT_STR " ");
+        sdp.append(width);
+        sdp.append("-");
+        sdp.append(height);
+        sdp.append("\r\n");
+
+        sdp.append(
+              "a=fmtp:" PT_STR " profile-level-id=42C015;sprop-parameter-sets=");
+
+        sdp.append(mSeqParamSet);
+        sdp.append(",");
+        sdp.append(mPicParamSet);
+        sdp.append(";packetization-mode=1\r\n");
+#else
+        sdp.append(
+                "m=audio 0 RTP/AVP " PT_STR "\r\n"
+                "a=rtpmap:" PT_STR " L8/8000/1\r\n");
+#endif
+
+        sdp.append("a=control:" TRACK_SUFFIX "\r\n");
+
+        AString request;
+        request.append("ANNOUNCE ");
+        request.append(mStreamURL);
+        request.append(" RTSP/1.0\r\n");
+
+        addAuthentication(&request, "ANNOUNCE", mStreamURL.c_str());
+
+        request.append("Content-Type: application/sdp\r\n");
+        request.append("Content-Length: ");
+        request.append(sdp.size());
+        request.append("\r\n");
+
+        request.append("\r\n");
+        request.append(sdp);
+
+        sp<AMessage> reply = new AMessage('anno', id());
+        mConn->sendRequest(request.c_str(), reply);
+    }
+
+    void H(const AString &s, AString *out) {
+        out->clear();
+
+        MD5_CTX m;
+        MD5_Init(&m);
+        MD5_Update(&m, s.c_str(), s.size());
+
+        uint8_t key[16];
+        MD5_Final(key, &m);
+
+        for (size_t i = 0; i < 16; ++i) {
+            char nibble = key[i] >> 4;
+            if (nibble <= 9) {
+                nibble += '0';
+            } else {
+                nibble += 'a' - 10;
+            }
+            out->append(&nibble, 1);
+
+            nibble = key[i] & 0x0f;
+            if (nibble <= 9) {
+                nibble += '0';
+            } else {
+                nibble += 'a' - 10;
+            }
+            out->append(&nibble, 1);
+        }
+    }
+
+    void authenticate(const sp<ARTSPResponse> &response) {
+        ssize_t i = response->mHeaders.indexOfKey("www-authenticate");
+        CHECK_GE(i, 0);
+
+        AString value = response->mHeaders.valueAt(i);
+
+        if (!strncmp(value.c_str(), "Basic", 5)) {
+            mAuthType = BASIC;
+        } else {
+            CHECK(!strncmp(value.c_str(), "Digest", 6));
+            mAuthType = DIGEST;
+
+            i = value.find("nonce=");
+            CHECK_GE(i, 0);
+            CHECK_EQ(value.c_str()[i + 6], '\"');
+            ssize_t j = value.find("\"", i + 7);
+            CHECK_GE(j, 0);
+
+            mNonce.setTo(value, i + 7, j - i - 7);
+        }
+
+        issueAnnounce();
+    }
+
+    void addAuthentication(
+            AString *request, const char *method, const char *url) {
+        if (mAuthType == NONE) {
+            return;
+        }
+
+        if (mAuthType == BASIC) {
+            request->append("Authorization: Basic YmNhc3Q6dGVzdAo=\r\n");
+            return;
+        }
+
+        CHECK_EQ((int)mAuthType, (int)DIGEST);
+
+        AString A1;
+        A1.append(USERNAME);
+        A1.append(":");
+        A1.append("Streaming Server");
+        A1.append(":");
+        A1.append(PASSWORD);
+
+        AString A2;
+        A2.append(method);
+        A2.append(":");
+        A2.append(url);
+
+        AString HA1, HA2;
+        H(A1, &HA1);
+        H(A2, &HA2);
+
+        AString tmp;
+        tmp.append(HA1);
+        tmp.append(":");
+        tmp.append(mNonce);
+        tmp.append(":");
+        tmp.append(HA2);
+
+        AString digest;
+        H(tmp, &digest);
+
+        request->append("Authorization: Digest ");
+        request->append("nonce=\"");
+        request->append(mNonce);
+        request->append("\", ");
+        request->append("username=\"" USERNAME "\", ");
+        request->append("uri=\"");
+        request->append(url);
+        request->append("\", ");
+        request->append("response=\"");
+        request->append(digest);
+        request->append("\"");
+        request->append("\r\n");
+    }
+
+    virtual void onMessageReceived(const sp<AMessage> &msg) {
+        switch (msg->what()) {
+            case 'conn':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "connection request completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                if (result != OK) {
+                    (new AMessage('quit', id()))->post();
+                    break;
+                }
+
+                mConnected = true;
+
+                CHECK(msg->findInt32("server-ip", (int32_t *)&mServerIP));
+
+                issueAnnounce();
+                break;
+            }
+
+            case 'anno':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "ANNOUNCE completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                sp<RefBase> obj;
+                CHECK(msg->findObject("response", &obj));
+                sp<ARTSPResponse> response;
+
+                if (result == OK) {
+                    response = static_cast<ARTSPResponse *>(obj.get());
+                    CHECK(response != NULL);
+
+                    if (response->mStatusCode == 401) {
+                        if (mAuthType != NONE) {
+                            LOG(INFO) << "FAILED to authenticate";
+                            (new AMessage('quit', id()))->post();
+                            break;
+                        }
+
+                        authenticate(response);
+                        break;
+                    }
+                }
+
+                if (result != OK || response->mStatusCode != 200) {
+                    (new AMessage('quit', id()))->post();
+                    break;
+                }
+
+                unsigned rtpPort;
+                ARTPConnection::MakePortPair(&mRTPSocket, &mRTCPSocket, &rtpPort);
+
+                // (new AMessage('poll', id()))->post();
+
+                AString request;
+                request.append("SETUP ");
+                request.append(mTrackURL);
+                request.append(" RTSP/1.0\r\n");
+
+                addAuthentication(&request, "SETUP", mTrackURL.c_str());
+
+                request.append("Transport: RTP/AVP;unicast;client_port=");
+                request.append(rtpPort);
+                request.append("-");
+                request.append(rtpPort + 1);
+                request.append(";mode=record\r\n");
+                request.append("\r\n");
+
+                sp<AMessage> reply = new AMessage('setu', id());
+                mConn->sendRequest(request.c_str(), reply);
+                break;
+            }
+
+#if 0
+            case 'poll':
+            {
+                fd_set rs;
+                FD_ZERO(&rs);
+                FD_SET(mRTCPSocket, &rs);
+
+                struct timeval tv;
+                tv.tv_sec = 0;
+                tv.tv_usec = 0;
+
+                int res = select(mRTCPSocket + 1, &rs, NULL, NULL, &tv);
+
+                if (res == 1) {
+                    sp<ABuffer> buffer = new ABuffer(65536);
+                    ssize_t n = recv(mRTCPSocket, buffer->data(), buffer->size(), 0);
+
+                    if (n <= 0) {
+                        LOG(ERROR) << "recv returned " << n;
+                    } else {
+                        LOG(INFO) << "recv returned " << n << " bytes of data.";
+
+                        hexdump(buffer->data(), n);
+                    }
+                }
+
+                msg->post(50000);
+                break;
+            }
+#endif
+
+            case 'setu':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "SETUP completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                sp<RefBase> obj;
+                CHECK(msg->findObject("response", &obj));
+                sp<ARTSPResponse> response;
+
+                if (result == OK) {
+                    response = static_cast<ARTSPResponse *>(obj.get());
+                    CHECK(response != NULL);
+                }
+
+                if (result != OK || response->mStatusCode != 200) {
+                    (new AMessage('quit', id()))->post();
+                    break;
+                }
+
+                ssize_t i = response->mHeaders.indexOfKey("session");
+                CHECK_GE(i, 0);
+                mSessionID = response->mHeaders.valueAt(i);
+                i = mSessionID.find(";");
+                if (i >= 0) {
+                    // Remove options, i.e. ";timeout=90"
+                    mSessionID.erase(i, mSessionID.size() - i);
+                }
+
+                i = response->mHeaders.indexOfKey("transport");
+                CHECK_GE(i, 0);
+                AString transport = response->mHeaders.valueAt(i);
+
+                LOG(INFO) << "transport = '" << transport << "'";
+
+                AString value;
+                CHECK(GetAttribute(transport.c_str(), "server_port", &value));
+
+                unsigned rtpPort, rtcpPort;
+                CHECK_EQ(sscanf(value.c_str(), "%u-%u", &rtpPort, &rtcpPort), 2);
+
+                CHECK(GetAttribute(transport.c_str(), "source", &value));
+
+                memset(mRemoteAddr.sin_zero, 0, sizeof(mRemoteAddr.sin_zero));
+                mRemoteAddr.sin_family = AF_INET;
+                mRemoteAddr.sin_addr.s_addr = inet_addr(value.c_str());
+                mRemoteAddr.sin_port = htons(rtpPort);
+
+                mRemoteRTCPAddr = mRemoteAddr;
+                mRemoteRTCPAddr.sin_port = htons(rtpPort + 1);
+
+                CHECK_EQ(0, connect(mRTPSocket,
+                                    (const struct sockaddr *)&mRemoteAddr,
+                                    sizeof(mRemoteAddr)));
+
+                CHECK_EQ(0, connect(mRTCPSocket,
+                                    (const struct sockaddr *)&mRemoteRTCPAddr,
+                                    sizeof(mRemoteRTCPAddr)));
+
+                uint32_t x = ntohl(mRemoteAddr.sin_addr.s_addr);
+                LOG(INFO) << "sending data to "
+                     << (x >> 24)
+                     << "."
+                     << ((x >> 16) & 0xff)
+                     << "."
+                     << ((x >> 8) & 0xff)
+                     << "."
+                     << (x & 0xff)
+                     << ":"
+                     << rtpPort;
+
+                AString request;
+                request.append("RECORD ");
+                request.append(mStreamURL);
+                request.append(" RTSP/1.0\r\n");
+
+                addAuthentication(&request, "RECORD", mStreamURL.c_str());
+
+                request.append("Session: ");
+                request.append(mSessionID);
+                request.append("\r\n");
+                request.append("\r\n");
+
+                sp<AMessage> reply = new AMessage('reco', id());
+                mConn->sendRequest(request.c_str(), reply);
+                break;
+            }
+
+            case 'reco':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "RECORD completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                sp<RefBase> obj;
+                CHECK(msg->findObject("response", &obj));
+                sp<ARTSPResponse> response;
+
+                if (result == OK) {
+                    response = static_cast<ARTSPResponse *>(obj.get());
+                    CHECK(response != NULL);
+                }
+
+                if (result != OK) {
+                    (new AMessage('quit', id()))->post();
+                    break;
+                }
+
+                (new AMessage('more', id()))->post();
+                (new AMessage('sr  ', id()))->post();
+                (new AMessage('aliv', id()))->post(30000000ll);
+                break;
+            }
+
+            case 'aliv':
+            {
+                if (!mConnected) {
+                    break;
+                }
+
+                AString request;
+                request.append("OPTIONS ");
+                request.append(mStreamURL);
+                request.append(" RTSP/1.0\r\n");
+
+                addAuthentication(&request, "RECORD", mStreamURL.c_str());
+
+                request.append("Session: ");
+                request.append(mSessionID);
+                request.append("\r\n");
+                request.append("\r\n");
+
+                sp<AMessage> reply = new AMessage('opts', id());
+                mConn->sendRequest(request.c_str(), reply);
+                break;
+            }
+
+            case 'opts':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "OPTIONS completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                if (!mConnected) {
+                    break;
+                }
+
+                (new AMessage('aliv', id()))->post(30000000ll);
+                break;
+            }
+
+            case 'more':
+            {
+                if (!mConnected) {
+                    break;
+                }
+
+                sp<ABuffer> buffer = new ABuffer(65536);
+                uint8_t *data = buffer->data();
+                data[0] = 0x80;
+                data[1] = (1 << 7) | PT;  // M-bit
+                data[2] = (mSeqNo >> 8) & 0xff;
+                data[3] = mSeqNo & 0xff;
+                data[8] = mSourceID >> 24;
+                data[9] = (mSourceID >> 16) & 0xff;
+                data[10] = (mSourceID >> 8) & 0xff;
+                data[11] = mSourceID & 0xff;
+
+#ifdef ANDROID
+                MediaBuffer *mediaBuf = NULL;
+                for (;;) {
+                    CHECK_EQ(mEncoder->read(&mediaBuf), (status_t)OK);
+                    if (mediaBuf->range_length() > 0) {
+                        break;
+                    }
+                    mediaBuf->release();
+                    mediaBuf = NULL;
+                }
+
+                int64_t timeUs;
+                CHECK(mediaBuf->meta_data()->findInt64(kKeyTime, &timeUs));
+
+                uint32_t rtpTime = mRTPTimeBase + (timeUs * 9 / 100ll);
+
+                const uint8_t *mediaData =
+                    (const uint8_t *)mediaBuf->data() + mediaBuf->range_offset();
+
+                CHECK(!memcmp("\x00\x00\x00\x01", mediaData, 4));
+
+                CHECK_LE(mediaBuf->range_length() - 4 + 12, buffer->size());
+
+                memcpy(&data[12],
+                       mediaData + 4, mediaBuf->range_length() - 4);
+
+                buffer->setRange(0, mediaBuf->range_length() - 4 + 12);
+
+                mediaBuf->release();
+                mediaBuf = NULL;
+#else
+                uint32_t rtpTime = mRTPTimeBase + mNumRTPSent * 128;
+                memset(&data[12], 0, 128);
+                buffer->setRange(0, 12 + 128);
+#endif
+
+                data[4] = rtpTime >> 24;
+                data[5] = (rtpTime >> 16) & 0xff;
+                data[6] = (rtpTime >> 8) & 0xff;
+                data[7] = rtpTime & 0xff;
+
+                ssize_t n = send(
+                        mRTPSocket, data, buffer->size(), 0);
+                if (n < 0) {
+                    LOG(ERROR) << "send failed (" << strerror(errno) << ")";
+                }
+                CHECK_EQ(n, (ssize_t)buffer->size());
+
+                ++mSeqNo;
+
+                ++mNumRTPSent;
+                mNumRTPOctetsSent += buffer->size() - 12;
+
+                mLastRTPTime = rtpTime;
+                mLastNTPTime = ntpTime();
+
+#ifdef ANDROID
+                if (mNumRTPSent < 60 * 25) {  // 60 secs worth
+                    msg->post(40000);
+#else
+                if (mNumRTPOctetsSent < 8000 * 60) {
+                    msg->post(1000000ll * 128 / 8000);
+#endif
+                } else {
+                    LOG(INFO) << "That's enough, pausing.";
+
+                    AString request;
+                    request.append("PAUSE ");
+                    request.append(mStreamURL);
+                    request.append(" RTSP/1.0\r\n");
+
+                    addAuthentication(&request, "PAUSE", mStreamURL.c_str());
+
+                    request.append("Session: ");
+                    request.append(mSessionID);
+                    request.append("\r\n");
+                    request.append("\r\n");
+
+                    sp<AMessage> reply = new AMessage('paus', id());
+                    mConn->sendRequest(request.c_str(), reply);
+                }
+                break;
+            }
+
+            case 'sr  ':
+            {
+                if (!mConnected) {
+                    break;
+                }
+
+                sp<ABuffer> buffer = new ABuffer(65536);
+                buffer->setRange(0, 0);
+
+                addSR(buffer);
+                addSDES(buffer);
+
+                uint8_t *data = buffer->data();
+                ssize_t n = send(
+                        mRTCPSocket, data, buffer->size(), 0);
+                CHECK_EQ(n, (ssize_t)buffer->size());
+
+                msg->post(3000000);
+                break;
+            }
+
+            case 'paus':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "PAUSE completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                sp<RefBase> obj;
+                CHECK(msg->findObject("response", &obj));
+                sp<ARTSPResponse> response;
+
+                AString request;
+                request.append("TEARDOWN ");
+                request.append(mStreamURL);
+                request.append(" RTSP/1.0\r\n");
+
+                addAuthentication(&request, "TEARDOWN", mStreamURL.c_str());
+
+                request.append("Session: ");
+                request.append(mSessionID);
+                request.append("\r\n");
+                request.append("\r\n");
+
+                sp<AMessage> reply = new AMessage('tear', id());
+                mConn->sendRequest(request.c_str(), reply);
+                break;
+            }
+
+            case 'tear':
+            {
+                int32_t result;
+                CHECK(msg->findInt32("result", &result));
+
+                LOG(INFO) << "TEARDOWN completed with result "
+                     << result << " (" << strerror(-result) << ")";
+
+                sp<RefBase> obj;
+                CHECK(msg->findObject("response", &obj));
+                sp<ARTSPResponse> response;
+
+                if (result == OK) {
+                    response = static_cast<ARTSPResponse *>(obj.get());
+                    CHECK(response != NULL);
+                }
+
+                (new AMessage('quit', id()))->post();
+                break;
+            }
+
+            case 'disc':
+            {
+                LOG(INFO) << "disconnect completed";
+
+                mConnected = false;
+                (new AMessage('quit', id()))->post();
+                break;
+            }
+
+            case 'quit':
+            {
+                if (mConnected) {
+                    mConn->disconnect(new AMessage('disc', id()));
+                    break;
+                }
+
+                if (mRTPSocket >= 0) {
+                    close(mRTPSocket);
+                    mRTPSocket = -1;
+                }
+
+                if (mRTCPSocket >= 0) {
+                    close(mRTCPSocket);
+                    mRTCPSocket = -1;
+                }
+
+#ifdef ANDROID
+                mEncoder->stop();
+                mEncoder.clear();
+#endif
+
+                mLooper->stop();
+                break;
+            }
+
+            default:
+                TRESPASS();
+        }
+    }
+
+protected:
+    virtual ~MyTransmitter() {
+    }
+
+private:
+    enum AuthType {
+        NONE,
+        BASIC,
+        DIGEST
+    };
+
+    AString mServerURL;
+    AString mTrackURL;
+    AString mStreamURL;
+
+    sp<ALooper> mLooper;
+    sp<ARTSPConnection> mConn;
+    bool mConnected;
+    uint32_t mServerIP;
+    AuthType mAuthType;
+    AString mNonce;
+    AString mSessionID;
+    int mRTPSocket, mRTCPSocket;
+    uint32_t mSourceID;
+    uint32_t mSeqNo;
+    uint32_t mRTPTimeBase;
+    struct sockaddr_in mRemoteAddr;
+    struct sockaddr_in mRemoteRTCPAddr;
+    size_t mNumSamplesSent;
+    uint32_t mNumRTPSent;
+    uint32_t mNumRTPOctetsSent;
+    uint32_t mLastRTPTime;
+    uint64_t mLastNTPTime;
+
+#ifdef ANDROID
+    sp<MediaSource> mEncoder;
+    AString mSeqParamSet;
+    AString mPicParamSet;
+
+    void makeH264SPropParamSets(MediaBuffer *buffer) {
+        static const char kStartCode[] = "\x00\x00\x00\x01";
+
+        const uint8_t *data =
+            (const uint8_t *)buffer->data() + buffer->range_offset();
+        size_t size = buffer->range_length();
+
+        CHECK_GE(size, 0u);
+        CHECK(!memcmp(kStartCode, data, 4));
+
+        data += 4;
+        size -= 4;
+
+        size_t startCodePos = 0;
+        while (startCodePos + 3 < size
+                && memcmp(kStartCode, &data[startCodePos], 4)) {
+            ++startCodePos;
+        }
+
+        CHECK_LT(startCodePos + 3, size);
+
+        encodeBase64(data, startCodePos, &mSeqParamSet);
+
+        encodeBase64(&data[startCodePos + 4], size - startCodePos - 4,
+                     &mPicParamSet);
+    }
+#endif
+
+    void addSR(const sp<ABuffer> &buffer) {
+        uint8_t *data = buffer->data() + buffer->size();
+
+        data[0] = 0x80 | 0;
+        data[1] = 200;  // SR
+        data[2] = 0;
+        data[3] = 6;
+        data[4] = mSourceID >> 24;
+        data[5] = (mSourceID >> 16) & 0xff;
+        data[6] = (mSourceID >> 8) & 0xff;
+        data[7] = mSourceID & 0xff;
+
+        data[8] = mLastNTPTime >> (64 - 8);
+        data[9] = (mLastNTPTime >> (64 - 16)) & 0xff;
+        data[10] = (mLastNTPTime >> (64 - 24)) & 0xff;
+        data[11] = (mLastNTPTime >> 32) & 0xff;
+        data[12] = (mLastNTPTime >> 24) & 0xff;
+        data[13] = (mLastNTPTime >> 16) & 0xff;
+        data[14] = (mLastNTPTime >> 8) & 0xff;
+        data[15] = mLastNTPTime & 0xff;
+
+        data[16] = (mLastRTPTime >> 24) & 0xff;
+        data[17] = (mLastRTPTime >> 16) & 0xff;
+        data[18] = (mLastRTPTime >> 8) & 0xff;
+        data[19] = mLastRTPTime & 0xff;
+
+        data[20] = mNumRTPSent >> 24;
+        data[21] = (mNumRTPSent >> 16) & 0xff;
+        data[22] = (mNumRTPSent >> 8) & 0xff;
+        data[23] = mNumRTPSent & 0xff;
+
+        data[24] = mNumRTPOctetsSent >> 24;
+        data[25] = (mNumRTPOctetsSent >> 16) & 0xff;
+        data[26] = (mNumRTPOctetsSent >> 8) & 0xff;
+        data[27] = mNumRTPOctetsSent & 0xff;
+
+        buffer->setRange(buffer->offset(), buffer->size() + 28);
+    }
+
+    void addSDES(const sp<ABuffer> &buffer) {
+        uint8_t *data = buffer->data() + buffer->size();
+        data[0] = 0x80 | 1;
+        data[1] = 202;  // SDES
+        data[4] = mSourceID >> 24;
+        data[5] = (mSourceID >> 16) & 0xff;
+        data[6] = (mSourceID >> 8) & 0xff;
+        data[7] = mSourceID & 0xff;
+
+        size_t offset = 8;
+
+        data[offset++] = 1;  // CNAME
+
+        static const char *kCNAME = "andih@laptop";
+        data[offset++] = strlen(kCNAME);
+
+        memcpy(&data[offset], kCNAME, strlen(kCNAME));
+        offset += strlen(kCNAME);
+
+        data[offset++] = 7;  // NOTE
+
+        static const char *kNOTE = "Hell's frozen over.";
+        data[offset++] = strlen(kNOTE);
+
+        memcpy(&data[offset], kNOTE, strlen(kNOTE));
+        offset += strlen(kNOTE);
+
+        data[offset++] = 0;
+
+        if ((offset % 4) > 0) {
+            size_t count = 4 - (offset % 4);
+            switch (count) {
+                case 3:
+                    data[offset++] = 0;
+                case 2:
+                    data[offset++] = 0;
+                case 1:
+                    data[offset++] = 0;
+            }
+        }
+
+        size_t numWords = (offset / 4) - 1;
+        data[2] = numWords >> 8;
+        data[3] = numWords & 0xff;
+
+        buffer->setRange(buffer->offset(), buffer->size() + offset);
+    }
+
+    DISALLOW_EVIL_CONSTRUCTORS(MyTransmitter);
+};
+
+}  // namespace android
+
+#endif  // MY_TRANSMITTER_H_
diff --git a/media/libstagefright/rtsp/VideoSource.h b/media/libstagefright/rtsp/VideoSource.h
new file mode 100644
index 0000000..ae0c85b
--- /dev/null
+++ b/media/libstagefright/rtsp/VideoSource.h
@@ -0,0 +1,98 @@
+/*
+ * 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 VIDEO_SOURCE_H_
+
+#define VIDEO_SOURCE_H_
+
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+
+namespace android {
+
+class VideoSource : public MediaSource {
+    static const int32_t kFramerate = 24;  // fps
+
+public:
+    VideoSource(int width, int height)
+        : mWidth(width),
+          mHeight(height),
+          mSize((width * height * 3) / 2) {
+        mGroup.add_buffer(new MediaBuffer(mSize));
+    }
+
+    virtual sp<MetaData> getFormat() {
+        sp<MetaData> meta = new MetaData;
+        meta->setInt32(kKeyWidth, mWidth);
+        meta->setInt32(kKeyHeight, mHeight);
+        meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_RAW);
+
+        return meta;
+    }
+
+    virtual status_t start(MetaData *params) {
+        mNumFramesOutput = 0;
+        return OK;
+    }
+
+    virtual status_t stop() {
+        return OK;
+    }
+
+    virtual status_t read(
+            MediaBuffer **buffer, const MediaSource::ReadOptions *options) {
+        if (mNumFramesOutput == kFramerate * 100) {
+            // Stop returning data after 10 secs.
+            return ERROR_END_OF_STREAM;
+        }
+
+        // printf("VideoSource::read\n");
+        status_t err = mGroup.acquire_buffer(buffer);
+        if (err != OK) {
+            return err;
+        }
+
+        char x = (char)((double)rand() / RAND_MAX * 255);
+        memset((*buffer)->data(), x, mSize);
+        (*buffer)->set_range(0, mSize);
+        (*buffer)->meta_data()->clear();
+        (*buffer)->meta_data()->setInt64(
+                kKeyTime, (mNumFramesOutput * 1000000) / kFramerate);
+        ++mNumFramesOutput;
+
+        // printf("VideoSource::read - returning buffer\n");
+        // LOG(INFO)("VideoSource::read - returning buffer");
+        return OK;
+    }
+
+protected:
+    virtual ~VideoSource() {}
+
+private:
+    MediaBufferGroup mGroup;
+    int mWidth, mHeight;
+    size_t mSize;
+    int64_t mNumFramesOutput;;
+
+    VideoSource(const VideoSource &);
+    VideoSource &operator=(const VideoSource &);
+};
+
+}  // namespace android
+
+#endif  // VIDEO_SOURCE_H_