Revert of Separating SCTP code from BaseChannel/MediaChannel.

webrtc/api/peerconnection.cc

 /*
  *  Copyright 2012 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 20 matching lines @@
 #include "webrtc/api/videocapturertracksource.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/media/sctp/sctptransport.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace {
 
 using webrtc::DataChannel;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::RtpSenderInternal;
@@ skipping 593 matching lines @@
   media_controller_.reset(factory_->CreateMediaController(
       configuration.media_config, event_log_.get()));
 
   session_.reset(new WebRtcSession(
       media_controller_.get(), factory_->network_thread(),
       factory_->worker_thread(), factory_->signaling_thread(),
       port_allocator_.get(),
       std::unique_ptr<cricket::TransportController>(
           factory_->CreateTransportController(
               port_allocator_.get(),
-              configuration.redetermine_role_on_ice_restart)),
-#ifdef HAVE_SCTP
-      std::unique_ptr<cricket::SctpTransportInternalFactory>(
-          new cricket::SctpTransportFactory(factory_->network_thread()))
-#else
-      nullptr
-#endif
-          ));
+              configuration.redetermine_role_on_ice_restart))));
 
   stats_.reset(new StatsCollector(this));
   stats_collector_ = RTCStatsCollector::Create(this);
 
   // Initialize the WebRtcSession. It creates transport channels etc.
   if (!session_->Initialize(factory_->options(), std::move(cert_generator),
                             configuration)) {
     return false;
   }
 
@@ skipping 446 matching lines @@
   std::string error;
   if (!session_->SetLocalDescription(desc, &error)) {
     PostSetSessionDescriptionFailure(observer, error);
     return;
   }
 
   // If setting the description decided our SSL role, allocate any necessary
   // SCTP sids.
   rtc::SSLRole role;
   if (session_->data_channel_type() == cricket::DCT_SCTP &&
-      session_->GetSctpSslRole(&role)) {
+      session_->GetSslRole(session_->data_channel(), &role)) {
     AllocateSctpSids(role);
   }
 
   // Update state and SSRC of local MediaStreams and DataChannels based on the
   // local session description.
   const cricket::ContentInfo* audio_content =
       GetFirstAudioContent(desc->description());
   if (audio_content) {
     if (audio_content->rejected) {
       RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
@@ skipping 61 matching lines @@
   std::string error;
   if (!session_->SetRemoteDescription(desc, &error)) {
     PostSetSessionDescriptionFailure(observer, error);
     return;
   }
 
   // If setting the description decided our SSL role, allocate any necessary
   // SCTP sids.
   rtc::SSLRole role;
   if (session_->data_channel_type() == cricket::DCT_SCTP &&
-      session_->GetSctpSslRole(&role)) {
+      session_->GetSslRole(session_->data_channel(), &role)) {
     AllocateSctpSids(role);
   }
 
   const cricket::SessionDescription* remote_desc = desc->description();
   const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
   const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
   const cricket::AudioContentDescription* audio_desc =
       GetFirstAudioContentDescription(remote_desc);
   const cricket::VideoContentDescription* video_desc =
       GetFirstVideoContentDescription(remote_desc);
@@ skipping 921 matching lines @@
   if (session_->data_channel_type() == cricket::DCT_NONE) {
     LOG(LS_ERROR)
         << "InternalCreateDataChannel: Data is not supported in this call.";
     return nullptr;
   }
   InternalDataChannelInit new_config =
       config ? (*config) : InternalDataChannelInit();
   if (session_->data_channel_type() == cricket::DCT_SCTP) {
     if (new_config.id < 0) {
       rtc::SSLRole role;
-      if ((session_->GetSctpSslRole(&role)) &&
+      if ((session_->GetSslRole(session_->data_channel(), &role)) &&
           !sid_allocator_.AllocateSid(role, &new_config.id)) {
         LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
         return nullptr;
       }
     } else if (!sid_allocator_.ReserveSid(new_config.id)) {
       LOG(LS_ERROR) << "Failed to create a SCTP data channel "
                     << "because the id is already in use or out of range.";
       return nullptr;
     }
   }
@@ skipping 266 matching lines @@
 
 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
                                         int64_t max_size_bytes) {
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   event_log_->StopLogging();
 }
 }  // namespace webrtc