Prevent RTCP SR to be sent with bogus timestamp.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index a5e2076a2137dae344dc6d01a409b15611be3fd6..b4165e5d29ebb60fb7c71e90c38c784b18ac5344 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -346,6 +346,18 @@ TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
   EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
 }
 
+TEST_F(RtpRtcpImplTest, NoSrBeforeMedia) {

[stefan-webrtc, 2015/12/10 08:05:40]

Do you think there's anything else to test, such as that RR still will be sent
even if we haven't sent rtp packets? Maybe we already have a test that verifies
this?

[mflodman, 2015/12/10 08:59:53]

I added a test verifying the receive module still is sending RR.

+  sender_.impl_->Process();
+  EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
+  clock_.AdvanceTimeMilliseconds(2000);
+  int64_t current_time = clock_.TimeInMilliseconds();
+  sender_.impl_->Process();
+  EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
+  SendFrame(&sender_, kBaseLayerTid);
+  // sender_.impl_->Process();

[mflodman, 2015/12/10 06:08:20]

Depending on the answer to the other comment, I'll remove this or add an extra
check that no SR is sent with SendFrame().

+  EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, current_time);
+}
+
 TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_Nack) {
   EXPECT_EQ(-1, receiver_.RtcpSent().first_packet_time_ms);
   EXPECT_EQ(-1, sender_.RtcpReceived().first_packet_time_ms);
@@ -522,5 +534,4 @@ TEST_F(RtpRtcpImplTest, UniqueNackRequests) {
   EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
   EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
 }
-
 }  // namespace webrtc