Introduce RtpTransportInternal and SrtpTransport.

webrtc/pc/rtptransport.cc

 /*
  *  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 59 matching lines @@
 }
 
 bool RtpTransport::IsWritable(bool rtcp) const {
   rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
                                                 ? rtcp_packet_transport_
                                                 : rtp_packet_transport_;
   return transport && transport->writable();
 }
 
 bool RtpTransport::SendPacket(bool rtcp,
-                              const rtc::CopyOnWriteBuffer* packet,
+                              rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketOptions& options,
                               int flags) {
   rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
                                                 ? rtcp_packet_transport_
                                                 : rtp_packet_transport_;
   int ret = transport->SendPacket(packet->data<char>(), packet->size(), options,
                                   flags);
   if (ret != static_cast<int>(packet->size())) {
     if (transport->GetError() == ENOTCONN) {
       LOG(LS_WARNING) << "Got ENOTCONN from transport.";
@@ skipping 112 matching lines @@
   }
   if (rtcp) {
     // Permit all (seemingly valid) RTCP packets.
     return true;
   }
   // Check whether we handle this payload.
   return HandlesPacket(packet->data(), packet->size());
 }
 
 }  // namespace webrtc