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..d2df25287ea210ce23bca075ba1d936a850ab44b 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -304,18 +304,19 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
                          Random* prng) {
   RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
   Clock* clock = Clock::GetRealTimeClock();
-
-  RTPSender rtp_sender(false,     // bool audio
-                       clock,     // Clock* clock
-                       nullptr,   // Transport*
-                       nullptr,   // RtpAudioFeedback*
-                       nullptr,   // PacedSender*
-                       nullptr,   // PacketRouter*
-                       nullptr,   // SendTimeObserver*
-                       nullptr,   // BitrateStatisticsObserver*
-                       nullptr,   // FrameCountObserver*
-                       nullptr,   // SendSideDelayObserver*
-                       nullptr);  // RtcEventLog*
+  SSRCDatabase ssrc_database;
+  RTPSender rtp_sender(false,    // bool audio
+                       clock,    // Clock* clock
+                       nullptr,  // Transport*
+                       nullptr,  // RtpAudioFeedback*
+                       nullptr,  // PacedSender*
+                       nullptr,  // PacketRouter*
+                       nullptr,  // SendTimeObserver*
+                       nullptr,  // BitrateStatisticsObserver*
+                       nullptr,  // FrameCountObserver*
+                       nullptr,  // SendSideDelayObserver*
+                       nullptr,  // RtcEventLog*
+                       &ssrc_database);
 
   std::vector<uint32_t> csrcs;
   for (unsigned i = 0; i < csrcs_count; i++) {