We're now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup.

Change-Id: Idc3df74b42000f7a6aa3eae090718dc9d9c4186f

5 files changed, 47 insertions, 39 deletions

diff --git a/media/libstagefright/rtsp/APacketSource.cpp b/media/libstagefright/rtsp/APacketSource.cpp
index a5777045..395cd28c 100644
--- a/media/libstagefright/rtsp/APacketSource.cpp
+++ b/media/libstagefright/rtsp/APacketSource.cpp
@@ -356,24 +356,10 @@ status_t APacketSource::read(
     if (!mBuffers.empty()) {
         const sp<ABuffer> buffer = *mBuffers.begin();
 
-        uint64_t ntpTime;
-        CHECK(buffer->meta()->findInt64(
-                    "ntp-time", (int64_t *)&ntpTime));
-
         MediaBuffer *mediaBuffer = new MediaBuffer(buffer->size());
-        mediaBuffer->meta_data()->setInt64(kKeyNTPTime, ntpTime);
-
-        if (mFirstAccessUnit) {
-            mFirstAccessUnit = false;
-            mFirstAccessUnitNTP = ntpTime;
-        }
-        if (ntpTime > mFirstAccessUnitNTP) {
-            ntpTime -= mFirstAccessUnitNTP;
-        } else {
-            ntpTime = 0;
-        }
 
-        int64_t timeUs = (int64_t)(ntpTime * 1E6 / (1ll << 32));
+        int64_t timeUs;
+        CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
 
         mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
 
@@ -390,10 +376,29 @@ status_t APacketSource::read(
 void APacketSource::queueAccessUnit(const sp<ABuffer> &buffer) {
     int32_t damaged;
     if (buffer->meta()->findInt32("damaged", &damaged) && damaged) {
-        // LOG(VERBOSE) << "discarding damaged AU";
+        LOG(INFO) << "discarding damaged AU";
         return;
     }
 
+    uint64_t ntpTime;
+    CHECK(buffer->meta()->findInt64(
+                "ntp-time", (int64_t *)&ntpTime));
+
+    if (mFirstAccessUnit) {
+        mFirstAccessUnit = false;
+        mFirstAccessUnitNTP = ntpTime;
+    }
+
+    if (ntpTime > mFirstAccessUnitNTP) {
+        ntpTime -= mFirstAccessUnitNTP;
+    } else {
+        ntpTime = 0;
+    }
+
+    int64_t timeUs = (int64_t)(ntpTime * 1E6 / (1ll << 32));
+
+    buffer->meta()->setInt64("timeUs", timeUs);
+
     Mutex::Autolock autoLock(mLock);
     mBuffers.push_back(buffer);
     mCondition.signal();
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 5bd306b4..469af3e2 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -28,8 +28,6 @@
 #include <arpa/inet.h>
 #include <sys/socket.h>
 
-#define IGNORE_RTCP_TIME        0
-
 namespace android {
 
 static const size_t kMaxUDPSize = 1500;
@@ -61,8 +59,9 @@ struct ARTPConnection::StreamInfo {
     struct sockaddr_in mRemoteRTCPAddr;
 };
 
-ARTPConnection::ARTPConnection()
-    : mPollEventPending(false),
+ARTPConnection::ARTPConnection(uint32_t flags)
+    : mFlags(flags),
+      mPollEventPending(false),
       mLastReceiverReportTimeUs(-1) {
 }
 
@@ -280,7 +279,10 @@ void ARTPConnection::onPollStreams() {
                 sp<ARTPSource> source = s->mSources.valueAt(i);
 
                 source->addReceiverReport(buffer);
-                source->addFIR(buffer);
+
+                if (mFlags & kRegularlyRequestFIR) {
+                    source->addFIR(buffer);
+                }
             }
 
             if (buffer->size() > 0) {
@@ -405,13 +407,11 @@ status_t ARTPConnection::parseRTP(StreamInfo *s, const sp<ABuffer> &buffer) {
     buffer->setInt32Data(u16at(&data[2]));
     buffer->setRange(payloadOffset, size - payloadOffset);
 
-#if IGNORE_RTCP_TIME
-    if (!source->timeEstablished()) {
+    if ((mFlags & kFakeTimestamps) && !source->timeEstablished()) {
         source->timeUpdate(rtpTime, 0);
-        source->timeUpdate(rtpTime + 20, 0x100000000ll);
+        source->timeUpdate(rtpTime + 90000, 0x100000000ll);
         CHECK(source->timeEstablished());
     }
-#endif
 
     source->processRTPPacket(buffer);
 
@@ -533,9 +533,9 @@ status_t ARTPConnection::parseSR(
 
     sp<ARTPSource> source = findSource(s, id);
 
-#if !IGNORE_RTCP_TIME
-    source->timeUpdate(rtpTime, ntpTime);
-#endif
+    if ((mFlags & kFakeTimestamps) == 0) {
+        source->timeUpdate(rtpTime, ntpTime);
+    }
 
     return 0;
 }
diff --git a/media/libstagefright/rtsp/ARTPConnection.h b/media/libstagefright/rtsp/ARTPConnection.h
index 49839ad6..c5351997 100644
--- a/media/libstagefright/rtsp/ARTPConnection.h
+++ b/media/libstagefright/rtsp/ARTPConnection.h
@@ -28,7 +28,12 @@ struct ARTPSource;
 struct ASessionDescription;
 
 struct ARTPConnection : public AHandler {
-    ARTPConnection();
+    enum Flags {
+        kFakeTimestamps      = 1,
+        kRegularlyRequestFIR = 2,
+    };
+
+    ARTPConnection(uint32_t flags = 0);
 
     void addStream(
             int rtpSocket, int rtcpSocket,
@@ -56,6 +61,8 @@ private:
 
     static const int64_t kSelectTimeoutUs;
 
+    uint32_t mFlags;
+
     struct StreamInfo;
     List<StreamInfo> mStreams;
 
diff --git a/media/libstagefright/rtsp/ARTPSession.cpp b/media/libstagefright/rtsp/ARTPSession.cpp
index 0e0f45ad..e0820782 100644
--- a/media/libstagefright/rtsp/ARTPSession.cpp
+++ b/media/libstagefright/rtsp/ARTPSession.cpp
@@ -40,7 +40,10 @@ status_t ARTPSession::setup(const sp<ASessionDescription> &desc) {
 
     mDesc = desc;
 
-    mRTPConn = new ARTPConnection;
+    mRTPConn = new ARTPConnection(
+            ARTPConnection::kFakeTimestamps
+                | ARTPConnection::kRegularlyRequestFIR);
+
     looper()->registerHandler(mRTPConn);
 
     for (size_t i = 1; i < mDesc->countTracks(); ++i) {
diff --git a/media/libstagefright/rtsp/ARTPSource.cpp b/media/libstagefright/rtsp/ARTPSource.cpp
index e08183e5..225f6e8d 100644
--- a/media/libstagefright/rtsp/ARTPSource.cpp
+++ b/media/libstagefright/rtsp/ARTPSource.cpp
@@ -98,7 +98,7 @@ void ARTPSource::timeUpdate(uint32_t rtpTime, uint64_t ntpTime) {
     mNTPTime[mNumTimes] = ntpTime;
     mRTPTime[mNumTimes++] = rtpTime;
 
-    if (mNumTimes == 2) {
+    if (timeEstablished()) {
         for (List<sp<ABuffer> >::iterator it = mQueue.begin();
              it != mQueue.end(); ++it) {
             sp<AMessage> meta = (*it)->meta();
@@ -112,13 +112,6 @@ void ARTPSource::timeUpdate(uint32_t rtpTime, uint64_t ntpTime) {
 }
 
 bool ARTPSource::queuePacket(const sp<ABuffer> &buffer) {
-#if 1
-    if (mNumTimes != 2) {
-        // Drop incoming packets until we've established a time base.
-        return false;
-    }
-#endif
-
     uint32_t seqNum = (uint32_t)buffer->int32Data();
 
     if (mNumTimes == 2) {