Completed the functionalities of SrtpTransport.

webrtc/pc/channel.h

 /*
  *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 15 matching lines @@
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
 #include "webrtc/p2p/base/dtlstransportinternal.h"
 #include "webrtc/p2p/base/packettransportinternal.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/p2p/client/socketmonitor.h"
 #include "webrtc/pc/audiomonitor.h"
 #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"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/network.h"
 #include "webrtc/rtc_base/sigslot.h"
 #include "webrtc/rtc_base/window.h"
 
 namespace webrtc {
 class AudioSinkInterface;
+class RtpTransportInternal;
+class SrtpTransport;
 }  // namespace webrtc
 
 namespace cricket {
 
 struct CryptoParams;
 class MediaContentDescription;
 
 // BaseChannel contains logic common to voice and video, including enable,
 // marshaling calls to a worker and network threads, and connection and media
 // monitors.
@@ skipping 314 matching lines @@
                      rtc::PacketTransportInternal* rtp_packet_transport,
                      rtc::PacketTransportInternal* rtcp_packet_transport);
   void DisconnectTransportChannels_n();
   void SignalSentPacket_n(rtc::PacketTransportInternal* transport,
                           const rtc::SentPacket& sent_packet);
   void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
   bool IsReadyToSendMedia_n() const;
   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_;
   rtc::Thread* const signaling_thread_;
   rtc::AsyncInvoker invoker_;
 
   const std::string content_name_;
   std::unique_ptr<ConnectionMonitor> connection_monitor_;
 
   // Won't be set when using raw packet transports. SDP-specific thing.
   std::string transport_name_;
 
   const bool rtcp_mux_required_;
 
   // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS.
   // Temporary measure until more refactoring is done.
   // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_".
   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_;
   RtcpMuxFilter rtcp_mux_filter_;
   bool writable_ = false;
   bool was_ever_writable_ = false;
   bool has_received_packet_ = false;
   bool dtls_keyed_ = false;
   const bool srtp_required_ = true;
   int rtp_abs_sendtime_extn_id_ = -1;
@@ skipping 317 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_