Completed the functionalities of SrtpTransport.

webrtc/pc/channel.h

Index: webrtc/pc/channel.h
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index c6dc29dd08c9a1829f5b1b63f98d20bad28038bf..f001c71a8406be7200f114e01477d6a386054e1a 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -33,7 +33,6 @@
 #include "webrtc/pc/mediamonitor.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/rtcpmuxfilter.h"
-#include "webrtc/pc/rtptransportinternal.h"
 #include "webrtc/pc/srtpfilter.h"
 #include "webrtc/rtc_base/asyncinvoker.h"
 #include "webrtc/rtc_base/asyncudpsocket.h"
@@ -44,6 +43,8 @@
 
 namespace webrtc {
 class AudioSinkInterface;
+class RtpTransportInternal;
+class SrtpTransport;
 }  // namespace webrtc
 
 namespace cricket {
@@ -378,6 +379,8 @@ class BaseChannel
   void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
   int GetTransportOverheadPerPacket() const;
   void UpdateTransportOverhead();
+  // Wraps the existing RtpTransport in an SrtpTransport.
+  void EnableSrtpTransport_n();
 
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;
@@ -398,6 +401,7 @@ class BaseChannel
   DtlsTransportInternal* rtp_dtls_transport_ = nullptr;
   DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
   std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
+  webrtc::SrtpTransport* srtp_transport_ = nullptr;
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
   SrtpFilter srtp_filter_;