Revert of 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 b95bd529b005a71e5b749dc1333425467ece376d..c6dc29dd08c9a1829f5b1b63f98d20bad28038bf 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -33,6 +33,7 @@
 #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"
@@ -43,8 +44,6 @@
 
 namespace webrtc {
 class AudioSinkInterface;
-class RtpTransportInternal;
-class SrtpTransport;
 }  // namespace webrtc
 
 namespace cricket {
@@ -100,12 +99,12 @@
   const std::string& transport_name() const { return transport_name_; }
   bool enabled() const { return enabled_; }
 
-  // This function returns true if we are using SDES.
-  bool sdes_active() const { return sdes_negotiator_.IsActive(); }
-  // The following function returns true if we are using DTLS-based keying.
-  bool dtls_active() const { return dtls_active_; }
-  // This function returns true if using SRTP (DTLS-based keying or SDES).
-  bool srtp_active() const { return sdes_active() || dtls_active(); }
+  // This function returns true if we are using SRTP.
+  bool secure() const { return srtp_filter_.IsActive(); }
+  // The following function returns true if we are using
+  // DTLS-based keying. If you turned off SRTP later, however
+  // you could have secure() == false and dtls_secure() == true.
+  bool secure_dtls() const { return dtls_keyed_; }
 
   bool writable() const { return writable_; }
 
@@ -189,6 +188,8 @@
       override;
   int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
 
+  SrtpFilter* srtp_filter() { return &srtp_filter_; }
+
   virtual cricket::MediaType media_type() = 0;
 
   // This function returns true if we require SRTP for call setup.
@@ -377,8 +378,6 @@
   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_;
@@ -399,16 +398,16 @@
   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 sdes_negotiator_;
+  SrtpFilter srtp_filter_;
   RtcpMuxFilter rtcp_mux_filter_;
   bool writable_ = false;
   bool was_ever_writable_ = false;
   bool has_received_packet_ = false;
-  bool dtls_active_ = false;
+  bool dtls_keyed_ = false;
   const bool srtp_required_ = true;
+  int rtp_abs_sendtime_extn_id_ = -1;
 
   // MediaChannel related members that should be accessed from the worker
   // thread.