Prevent RTCP SR to be sent with bogus timestamp.

This CL makes sure no RTCP SR is sent before there is a valid timestamp
to set in the SR, based on the first sent media packet.

BUG=webrtc:1600
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1506103006 .

Cr-Commit-Position: refs/heads/master@{#10964}

2 files changed, 29 insertions, 3 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 451f8bfa93..3d20bab31f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -183,8 +183,13 @@ int32_t ModuleRtpRtcpImpl::Process() {
       set_rtt_ms(rtt_stats_->LastProcessedRtt());
   }
 
-  if (rtcp_sender_.TimeToSendRTCPReport())
-    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+  // For sending streams, make sure to not send a SR before media has been sent.
+  if (rtcp_sender_.TimeToSendRTCPReport()) {
+    RTCPSender::FeedbackState state = GetFeedbackState();
+    // Prevent sending streams to send SR before any media has been sent.
+    if (!rtcp_sender_.Sending() || state.packets_sent > 0)
+      rtcp_sender_.SendRTCP(state, kRtcpReport);
+  }
 
   if (UpdateRTCPReceiveInformationTimers()) {
     // A receiver has timed out
@@ -402,6 +407,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     const RTPFragmentationHeader* fragmentation,
     const RTPVideoHeader* rtp_video_hdr) {
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
+  // Make sure an RTCP report isn't queued behind a key frame.
   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
       rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
   }
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 a5e2076a21..2576e321e8 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 +346,27 @@ TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
   EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
 }
 
+TEST_F(RtpRtcpImplTest, NoSrBeforeMedia) {
+  // Ignore fake transport delays in this test.
+  sender_.transport_.SimulateNetworkDelay(0, &clock_);
+  receiver_.transport_.SimulateNetworkDelay(0, &clock_);
+
+  sender_.impl_->Process();
+  EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
+
+  // Verify no SR is sent before media has been sent, RR should still be sent
+  // from the receiving module though.
+  clock_.AdvanceTimeMilliseconds(2000);
+  int64_t current_time = clock_.TimeInMilliseconds();
+  sender_.impl_->Process();
+  receiver_.impl_->Process();
+  EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms);
+  EXPECT_EQ(receiver_.RtcpSent().first_packet_time_ms, current_time);
+
+  SendFrame(&sender_, kBaseLayerTid);
+  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 +543,4 @@ TEST_F(RtpRtcpImplTest, UniqueNackRequests) {
   EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
   EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
 }
-
 }  // namespace webrtc