Add histogram stats for send delay for a sent video stream.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 332a0f81f9e098f30b16168a630a8336fc1e9296..4d68faf69fbb406a33de473aa7b1772208b25fa6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -134,7 +134,7 @@ class RtpSenderTest : public ::testing::Test {
   void SetUpRtpSender(bool pacer) {
     rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                     pacer ? &mock_paced_sender_ : nullptr,
-                                    nullptr, nullptr, nullptr, nullptr,
+                                    nullptr, nullptr, nullptr, nullptr, nullptr,
                                     nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
@@ -847,7 +847,7 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
                                   &mock_paced_sender_, nullptr, nullptr,
-                                  nullptr, nullptr, nullptr));
+                                  nullptr, nullptr, nullptr, nullptr));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
   // Make all packets go through the pacer.
@@ -980,7 +980,7 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
 
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                   &mock_paced_sender_, nullptr, nullptr,
-                                  nullptr, &callback, nullptr));
+                                  nullptr, &callback, nullptr, nullptr));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
@@ -1033,7 +1033,7 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
   } callback;
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                   nullptr, nullptr, nullptr, &callback, nullptr,
-                                  nullptr));
+                                  nullptr, nullptr));
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals.
   const uint32_t kNumPackets = 15;
@@ -1092,7 +1092,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
     payload_ = kAudioPayload;
     rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
                                     nullptr, nullptr, nullptr, nullptr, nullptr,
-                                    nullptr));
+                                    nullptr, nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };