Refactor RtpSender and SSRCDatabase a bit.

webrtc/call/rtc_event_log_unittest.cc

Index: webrtc/call/rtc_event_log_unittest.cc
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index 70032039303129242ea52cfb4509adc361993c80..2324407b9963a0c176ddb4d7fb4220f65add6191 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -304,7 +304,7 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
                          Random* prng) {
   RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
   Clock* clock = Clock::GetRealTimeClock();
-
+  SSRCDatabase ssrc_database;
   RTPSender rtp_sender(false,     // bool audio
                        clock,     // Clock* clock
                        nullptr,   // Transport*
@@ -315,7 +315,8 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
                        nullptr,   // BitrateStatisticsObserver*
                        nullptr,   // FrameCountObserver*
                        nullptr,   // SendSideDelayObserver*
-                       nullptr);  // RtcEventLog*
+                       nullptr,   // RtcEventLog*
+                       &ssrc_database);
 
   std::vector<uint32_t> csrcs;
   for (unsigned i = 0; i < csrcs_count; i++) {