Rename "PacketTransportInterface" to "PacketTransportInternal".

webrtc/pc/webrtcsession_unittest.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 16 matching lines @@
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/base/fakevideorenderer.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/engine/fakewebrtccall.h"
 #include "webrtc/media/sctp/sctptransportinternal.h"
-#include "webrtc/p2p/base/packettransportinterface.h"
+#include "webrtc/p2p/base/packettransportinternal.h"
 #include "webrtc/p2p/base/stunserver.h"
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 #include "webrtc/pc/audiotrack.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/pc/fakemediacontroller.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/peerconnection.h"
 #include "webrtc/pc/sctputils.h"
@@ skipping 155 matching lines @@
   std::vector<PeerConnectionInterface::IceConnectionState>
       ice_connection_state_history_;
   size_t num_candidates_removed_ = 0;
 };
 
 // Used for tests in this file to verify that WebRtcSession responds to signals
 // from the SctpTransport correctly, and calls Start with the correct
 // local/remote ports.
 class FakeSctpTransport : public cricket::SctpTransportInternal {
  public:
-  void SetTransportChannel(rtc::PacketTransportInterface* channel) override {}
+  void SetTransportChannel(rtc::PacketTransportInternal* channel) override {}
   bool Start(int local_port, int remote_port) override {
     local_port_ = local_port;
     remote_port_ = remote_port;
     return true;
   }
   bool OpenStream(int sid) override { return true; }
   bool ResetStream(int sid) override { return true; }
   bool SendData(const cricket::SendDataParams& params,
                 const rtc::CopyOnWriteBuffer& payload,
                 cricket::SendDataResult* result = nullptr) override {
     return true;
   }
   bool ReadyToSendData() override { return true; }
   void set_debug_name_for_testing(const char* debug_name) override {}
 
   int local_port() const { return local_port_; }
   int remote_port() const { return remote_port_; }
 
  private:
   int local_port_ = -1;
   int remote_port_ = -1;
 };
 
 class FakeSctpTransportFactory : public cricket::SctpTransportInternalFactory {
  public:
   std::unique_ptr<cricket::SctpTransportInternal> CreateSctpTransport(
-      rtc::PacketTransportInterface*) override {
+      rtc::PacketTransportInternal*) override {
     last_fake_sctp_transport_ = new FakeSctpTransport();
     return std::unique_ptr<cricket::SctpTransportInternal>(
         last_fake_sctp_transport_);
   }
 
   FakeSctpTransport* last_fake_sctp_transport() {
     return last_fake_sctp_transport_;
   }
 
  private:
@@ skipping 17 matching lines @@
                       signaling_thread,
                       port_allocator,
                       std::move(transport_controller),
                       std::move(sctp_factory)) {
     RegisterIceObserver(ice_observer);
   }
   virtual ~WebRtcSessionForTest() {}
 
   // Note that these methods are only safe to use if the signaling thread
   // is the same as the worker thread
-  rtc::PacketTransportInterface* voice_rtp_transport_channel() {
+  rtc::PacketTransportInternal* voice_rtp_transport_channel() {
     return rtp_transport_channel(voice_channel());
   }
 
-  rtc::PacketTransportInterface* voice_rtcp_transport_channel() {
+  rtc::PacketTransportInternal* voice_rtcp_transport_channel() {
     return rtcp_transport_channel(voice_channel());
   }
 
-  rtc::PacketTransportInterface* video_rtp_transport_channel() {
+  rtc::PacketTransportInternal* video_rtp_transport_channel() {
     return rtp_transport_channel(video_channel());
   }
 
-  rtc::PacketTransportInterface* video_rtcp_transport_channel() {
+  rtc::PacketTransportInternal* video_rtcp_transport_channel() {
     return rtcp_transport_channel(video_channel());
   }
 
  private:
-  rtc::PacketTransportInterface* rtp_transport_channel(
+  rtc::PacketTransportInternal* rtp_transport_channel(
       cricket::BaseChannel* ch) {
     if (!ch) {
       return nullptr;
     }
     return ch->rtp_dtls_transport();
   }
 
-  rtc::PacketTransportInterface* rtcp_transport_channel(
+  rtc::PacketTransportInternal* rtcp_transport_channel(
       cricket::BaseChannel* ch) {
     if (!ch) {
       return nullptr;
     }
     return ch->rtcp_dtls_transport();
   }
 };
 
 class WebRtcSessionCreateSDPObserverForTest
     : public rtc::RefCountedObject<CreateSessionDescriptionObserver> {
@@ skipping 2123 matching lines @@
                              &sdp);
 
   SessionDescriptionInterface* modified_offer =
       CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
 
   SetRemoteDescriptionWithoutError(modified_offer);
 
   SessionDescriptionInterface* answer = CreateAnswer();
   SetLocalDescriptionWithoutError(answer);
 
-  rtc::PacketTransportInterface* voice_transport_channel =
+  rtc::PacketTransportInternal* voice_transport_channel =
       session_->voice_rtp_transport_channel();
   EXPECT_TRUE(voice_transport_channel != NULL);
   EXPECT_EQ(voice_transport_channel->debug_name(),
             "audio_content_name " +
                 std::to_string(cricket::ICE_CANDIDATE_COMPONENT_RTP));
-  rtc::PacketTransportInterface* video_transport_channel =
+  rtc::PacketTransportInternal* video_transport_channel =
       session_->video_rtp_transport_channel();
   ASSERT_TRUE(video_transport_channel != NULL);
   EXPECT_EQ(video_transport_channel->debug_name(),
             "video_content_name " +
                 std::to_string(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   EXPECT_TRUE((video_channel_ = media_engine_->GetVideoChannel(0)) != NULL);
   EXPECT_TRUE((voice_channel_ = media_engine_->GetVoiceChannel(0)) != NULL);
 }
 
 // Test that an offer contains the correct media content descriptions based on
@@ skipping 1875 matching lines @@
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));