Merge "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." into gingerbread

1 files changed, 12 insertions, 12 deletions

diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 5bd306b..469af3e 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;
 }