Introduce RtpTransportInternal and SrtpTransport.

webrtc/pc/channel.cc

 /*
  *  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 13 matching lines @@
 #include "webrtc/rtc_base/dscp.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/networkroute.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/trace_event.h"
 // Adding 'nogncheck' to disable the gn include headers check to support modular
 // WebRTC build targets.
 #include "webrtc/media/engine/webrtcvoiceengine.h"  // nogncheck
 #include "webrtc/p2p/base/packettransportinternal.h"
 #include "webrtc/pc/channelmanager.h"
+#include "webrtc/pc/rtptransport.h"
+#include "webrtc/pc/srtptransport.h"
 
 namespace cricket {
 using rtc::Bind;
 
 namespace {
 // See comment below for why we need to use a pointer to a unique_ptr.
 bool SetRawAudioSink_w(VoiceMediaChannel* channel,
                        uint32_t ssrc,
                        std::unique_ptr<webrtc::AudioSinkInterface>* sink) {
   channel->SetRawAudioSink(ssrc, std::move(*sink));
@@ skipping 105 matching lines @@
                          rtc::Thread* signaling_thread,
                          MediaChannel* media_channel,
                          const std::string& content_name,
                          bool rtcp_mux_required,
                          bool srtp_required)
     : worker_thread_(worker_thread),
       network_thread_(network_thread),
       signaling_thread_(signaling_thread),
       content_name_(content_name),
       rtcp_mux_required_(rtcp_mux_required),
-      rtp_transport_(rtc::MakeUnique<webrtc::RtpTransport>(rtcp_mux_required)),
+      rtp_transport_(
+          srtp_required
+              ? rtc::WrapUnique<webrtc::RtpTransportInternal>(
+                    new webrtc::SrtpTransport(rtcp_mux_required, content_name))
+              : rtc::MakeUnique<webrtc::RtpTransport>(rtcp_mux_required)),
       srtp_required_(srtp_required),
       media_channel_(media_channel),
       selected_candidate_pair_(nullptr) {
   RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
 #if defined(ENABLE_EXTERNAL_AUTH)
   srtp_filter_.EnableExternalAuth();
 #endif
   rtp_transport_->SignalReadyToSend.connect(
       this, &BaseChannel::OnTransportReadyToSend);
   // TODO(zstein):  RtpTransport::SignalPacketReceived will probably be replaced
   // with a callback interface later so that the demuxer can select which
   // channel to signal.
   rtp_transport_->SignalPacketReceived.connect(this,
-                                              &BaseChannel::OnPacketReceived);
+                                               &BaseChannel::OnPacketReceived);
   LOG(LS_INFO) << "Created channel for " << content_name;
 }
 
 BaseChannel::~BaseChannel() {
   TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
   RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   Deinit();
   StopConnectionMonitor();
   // Eats any outstanding messages or packets.
   worker_thread_->Clear(&invoker_);
@@ skipping 2264 matching lines @@
 
 void RtpDataChannel::OnDataChannelReadyToSend(bool writable) {
   // This is usded for congestion control to indicate that the stream is ready
   // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
   // that the transport channel is ready.
   signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
                            new DataChannelReadyToSendMessageData(writable));
 }
 
 }  // namespace cricket