Add configuration and test for extended RTCP reference time reports to new video api.

R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d

5 files changed, 112 insertions, 8 deletions

diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 8cac9969a5..a376df79c9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -1778,10 +1778,9 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
       rtcpPacketTypeFlags |= kRtcpTmmbn;
       _sendTMMBN = false;
   }
-  if (xrSendReceiverReferenceTimeEnabled_ &&
-      (rtcpPacketTypeFlags & kRtcpReport))
+  if (rtcpPacketTypeFlags & kRtcpReport)
   {
-      if (!_sending)
+      if (xrSendReceiverReferenceTimeEnabled_ && !_sending)
       {
           rtcpPacketTypeFlags |= kRtcpXrReceiverReferenceTime;
       }
diff --git a/webrtc/video/call_tests.cc b/webrtc/video/call_tests.cc
index 120fa16df6..3ec66fdd66 100644
--- a/webrtc/video/call_tests.cc
+++ b/webrtc/video/call_tests.cc
@@ -127,6 +127,7 @@ class CallTest : public ::testing::Test {
 
   void ReceivesPliAndRecovers(int rtp_history_ms);
   void RespectsRtcpMode(newapi::RtcpMode rtcp_mode);
+  void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
 
   scoped_ptr<Call> sender_call_;
   scoped_ptr<Call> receiver_call_;
@@ -1051,4 +1052,100 @@ TEST_F(CallTest, ReceiveStreamSendsRemb) {
   observer.StopSending();
   DestroyStreams();
 }
+
+void CallTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
+  static const int kNumRtcpReportPacketsToObserve = 5;
+  class RtcpXrObserver : public test::RtpRtcpObserver {
+   public:
+    explicit RtcpXrObserver(bool enable_rrtr)
+        : test::RtpRtcpObserver(kDefaultTimeoutMs),
+          enable_rrtr_(enable_rrtr),
+          sent_rtcp_sr_(0),
+          sent_rtcp_rr_(0),
+          sent_rtcp_rrtr_(0),
+          sent_rtcp_dlrr_(0) {}
+   private:
+    // Receive stream should send RR packets (and RRTR packets if enabled).
+    virtual Action OnReceiveRtcp(const uint8_t* packet,
+                                 size_t length) OVERRIDE {
+      RTCPUtility::RTCPParserV2 parser(packet, length, true);
+      EXPECT_TRUE(parser.IsValid());
+
+      RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
+      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
+        if (packet_type == RTCPUtility::kRtcpRrCode) {
+          ++sent_rtcp_rr_;
+        } else if (
+            packet_type == RTCPUtility::kRtcpXrReceiverReferenceTimeCode) {
+          ++sent_rtcp_rrtr_;
+        }
+        EXPECT_NE(packet_type, RTCPUtility::kRtcpSrCode);
+        EXPECT_NE(packet_type, RTCPUtility::kRtcpXrDlrrReportBlockItemCode);
+        packet_type = parser.Iterate();
+      }
+      return SEND_PACKET;
+    }
+    // Send stream should send SR packets (and DLRR packets if enabled).
+    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
+      RTCPUtility::RTCPParserV2 parser(packet, length, true);
+      EXPECT_TRUE(parser.IsValid());
+
+      RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
+      while (packet_type != RTCPUtility::kRtcpNotValidCode) {
+        if (packet_type == RTCPUtility::kRtcpSrCode) {
+          ++sent_rtcp_sr_;
+        } else if (packet_type == RTCPUtility::kRtcpXrDlrrReportBlockItemCode) {
+          ++sent_rtcp_dlrr_;
+        }
+        EXPECT_NE(packet_type, RTCPUtility::kRtcpXrReceiverReferenceTimeCode);
+        packet_type = parser.Iterate();
+      }
+      if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
+          sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve) {
+        if (enable_rrtr_) {
+          EXPECT_GT(sent_rtcp_rrtr_, 0);
+          EXPECT_GT(sent_rtcp_dlrr_, 0);
+        } else {
+          EXPECT_EQ(0, sent_rtcp_rrtr_);
+          EXPECT_EQ(0, sent_rtcp_dlrr_);
+        }
+        observation_complete_->Set();
+      }
+      return SEND_PACKET;
+    }
+    bool enable_rrtr_;
+    int sent_rtcp_sr_;
+    int sent_rtcp_rr_;
+    int sent_rtcp_rrtr_;
+    int sent_rtcp_dlrr_;
+  } observer(enable_rrtr);
+
+  CreateCalls(Call::Config(observer.SendTransport()),
+              Call::Config(observer.ReceiveTransport()));
+  observer.SetReceivers(receiver_call_->Receiver(),
+                        sender_call_->Receiver());
+
+  CreateTestConfigs();
+  receive_config_.rtp.rtcp_mode = newapi::kRtcpReducedSize;
+  receive_config_.rtp.rtcp_xr.receiver_reference_time_report = enable_rrtr;
+
+  CreateStreams();
+  CreateFrameGenerator();
+  StartSending();
+
+  EXPECT_EQ(kEventSignaled, observer.Wait())
+      << "Timed out while waiting for RTCP SR/RR packets to be sent.";
+
+  StopSending();
+  observer.StopSending();
+  DestroyStreams();
+}
+
+TEST_F(CallTest, ReceiverReferenceTimeReportEnabled) {
+  TestXrReceiverReferenceTimeReport(true);
+}
+
+TEST_F(CallTest, ReceiverReferenceTimeReportDisabled) {
+  TestXrReceiverReferenceTimeReport(false);
+}
 }  // namespace webrtc
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 747ab8f219..71fc1521e4 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -127,6 +127,10 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
   image_process_->RegisterPreRenderCallback(channel_,
                                             config_.pre_render_callback);
 
+  if (config.rtp.rtcp_xr.receiver_reference_time_report) {
+    rtp_rtcp_->SetRtcpXrRrtrStatus(channel_, true);
+  }
+
   clock_ = Clock::GetRealTimeClock();
 }
 
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 6e58313b3a..c0439e7242 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -412,7 +412,6 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
         rtp_rtcp->DeregisterSendRtpHeaderExtension(
             kRtpExtensionAbsoluteSendTime);
       }
-      rtp_rtcp->SetRtcpXrRrtrStatus(rtp_rtcp_->RtcpXrRrtrStatus());
       rtp_rtcp->RegisterSendFrameCountObserver(
           rtp_rtcp_->GetSendFrameCountObserver());
       rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(
@@ -939,10 +938,6 @@ int ViEChannel::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) {
 void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   rtp_rtcp_->SetRtcpXrRrtrStatus(enable);
-  for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
-       it != simulcast_rtp_rtcp_.end(); it++) {
-    (*it)->SetRtcpXrRrtrStatus(enable);
-  }
 }
 
 void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
diff --git a/webrtc/video_receive_stream.h b/webrtc/video_receive_stream.h
index c6669b9d33..55e5a626ae 100644
--- a/webrtc/video_receive_stream.h
+++ b/webrtc/video_receive_stream.h
@@ -119,6 +119,15 @@ class VideoReceiveStream {
       // See RtcpMode for description.
       newapi::RtcpMode rtcp_mode;
 
+      // Extended RTCP settings.
+      struct RtcpXr {
+        RtcpXr() : receiver_reference_time_report(false) {}
+
+        // True if RTCP Receiver Reference Time Report Block extension
+        // (RFC 3611) should be enabled.
+        bool receiver_reference_time_report;
+      } rtcp_xr;
+
       // See draft-alvestrand-rmcat-remb for information.
       bool remb;