Validates sending RTCP before RTP.

webrtc/video/end_to_end_tests.cc

Index: webrtc/video/end_to_end_tests.cc
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index fc793eccf60b13b3ac9d56a9320d99307fede504..a12b63ea68892bce56063042e3e58e1a2a74da0d 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -80,7 +80,7 @@ class EndToEndTest : public test::CallTest {
   void RespectsRtcpMode(RtcpMode rtcp_mode);
   void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
   void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
-  void TestRtpStatePreservation(bool use_rtx);
+  void TestRtpStatePreservation(bool use_rtx, bool wait_rtcp);
   void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare);
 };
 
@@ -2936,7 +2936,7 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
   RunBaseTest(&test);
 }
 
-void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
+void EndToEndTest::TestRtpStatePreservation(bool use_rtx, bool wait_rtcp) {
   static const uint32_t kMaxSequenceNumberGap = 100;
   class RtpSequenceObserver : public test::RtpRtcpObserver {
    public:
@@ -3105,6 +3105,16 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
     video_send_stream_ =
         sender_call_->CreateVideoSendStream(video_send_config_, one_stream);
     video_send_stream_->Start();
+    if (wait_rtcp) {
+      // Wait for SR rtcp packet before generating rtp packets.

[stefan-webrtc, 2016/02/01 17:32:07]

I'm not sure I understand this test change. How can we even expect RTCP SR to be
sent before we send RTP? 

[danilchap, 2016/02/01 18:24:18]

RTP packets will not be send until FrameGeneratorCapturer will start to work.
I know 2 ways to wait for RTCP before starting FrameGeneratorCapturer:
One, longer way - just wait ~5 sec. Process thread will try to generate an SR
report. 
Another, faster way, the one used in this test - send rapid synchronization
request (transport feedback message with feedback_message_type=5). In response
to this message RtpRtcp module tries to send back SenderReport as fast as it
can. Current implementation doesn't validate Sender/Receiver SSRCs, so I put
zeroes in both these field to make code shorter.

Pathset#1 demonstrates test fails before the change.

+      // There should be no rtcp packet or its time should be in sync with rtp.
+
+      // Rapid Resync Request forces sending RTCP Sender Report back.
+      // Alternative approach is to wait several seconds for SR to be generated.
+      // TODO(danilchap): Create rtcp packet with dedicated class when it lands.
+      const uint8_t kPacket[] = {0x85, 205, 0x00, 0x02, 0, 0, 0, 0, 0, 0, 0, 0};
+      video_send_stream_->DeliverRtcp(kPacket, sizeof(kPacket));
+    }
     CreateFrameGeneratorCapturer();
     frame_generator_capturer_->Start();
 
@@ -3137,11 +3147,15 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
 }
 
 TEST_F(EndToEndTest, DISABLED_RestartingSendStreamPreservesRtpState) {
-  TestRtpStatePreservation(false);
+  TestRtpStatePreservation(false, false);
 }
 
 TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
-  TestRtpStatePreservation(true);
+  TestRtpStatePreservation(true, false);
+}
+
+TEST_F(EndToEndTest, RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) {
+  TestRtpStatePreservation(true, true);
 }
 
 TEST_F(EndToEndTest, RespectsNetworkState) {