Refactor RtpSender and SSRCDatabase a bit.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 04e09c12178c6c267c998d1fe73815192b7453af..bd718e335bc1a9a4927d2b88144bc58249a43f66 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -22,6 +22,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
+#include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
 #include "webrtc/test/testsupport/gtest_prod_util.h"
 
 namespace webrtc {
@@ -337,6 +338,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // Get remote SequenceNumber.
   uint16_t RemoteSequenceNumber() const;
 
+  SSRCDatabase ssrc_database_;

[mflodman, 2016/01/25 07:26:19]

Drive-by and Stefan might already have commented on this, but:
- The intention with the database is to ensure no ssrc is reused across all
sending and receiving streams.
- The new video API already checks this when creating send and receive streams,
so there is no real use case for the database in the video engine.
- VoiceEngine might still depend on this being checked across streams though, so
this change will then break currently use VoiceEngine functionality.

+
   RTPSender rtp_sender_;
 
   RTCPSender rtcp_sender_;