Completed the functionalities of SrtpTransport.

webrtc/pc/rtptransport.h

 /*
  *  Copyright 2017 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 38 matching lines @@
 
   PacketTransportInterface* GetRtpPacketTransport() const override;
   PacketTransportInterface* GetRtcpPacketTransport() const override;
 
   // TODO(zstein): Use these RtcpParameters for configuration elsewhere.
   RTCError SetParameters(const RtpTransportParameters& parameters) override;
   RtpTransportParameters GetParameters() const override;
 
   bool IsWritable(bool rtcp) const override;
 
-  bool SendPacket(bool rtcp,
-                  rtc::CopyOnWriteBuffer* packet,
-                  const rtc::PacketOptions& options,
-                  int flags) override;
+  bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
+                     const rtc::PacketOptions& options,
+                     int flags) override;
+
+  bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
+                      const rtc::PacketOptions& options,
+                      int flags) override;
 
   bool HandlesPayloadType(int payload_type) const override;
 
   void AddHandledPayloadType(int payload_type) override;
 
  protected:
   // TODO(zstein): Remove this when we remove RtpTransportAdapter.
   RtpTransportAdapter* GetInternal() override;
 
  private:
   bool HandlesPacket(const uint8_t* data, size_t len);
 
   void OnReadyToSend(rtc::PacketTransportInternal* transport);
 
   // Updates "ready to send" for an individual channel and fires
   // SignalReadyToSend.
   void SetReadyToSend(bool rtcp, bool ready);
 
   void MaybeSignalReadyToSend();
 
+  bool SendPacket(bool rtcp,
+                  rtc::CopyOnWriteBuffer* packet,
+                  const rtc::PacketOptions& options,
+                  int flags);
+
   void OnReadPacket(rtc::PacketTransportInternal* transport,
                     const char* data,
                     size_t len,
                     const rtc::PacketTime& packet_time,
                     int flags);
 
   bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
 
   bool rtcp_mux_enabled_;
 
   rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
   rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr;
 
   bool ready_to_send_ = false;
   bool rtp_ready_to_send_ = false;
   bool rtcp_ready_to_send_ = false;
 
   RtpTransportParameters parameters_;
 
   cricket::BundleFilter bundle_filter_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_RTPTRANSPORT_H_