Use rtc::PacketTransportInterface in WebrtcSession unit test.

webrtc/api/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 22 matching lines @@
 #include "webrtc/base/sslidentity.h"
 #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/p2p/base/packettransportinterface.h"
 #include "webrtc/p2p/base/stunserver.h"
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/base/transportchannel.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/pc/mediasession.h"
 
 #define MAYBE_SKIP_TEST(feature)                    \
   if (!(feature())) {                               \
@@ skipping 172 matching lines @@
                       worker_thread,
                       signaling_thread,
                       port_allocator,
                       std::move(transport_controller)) {
     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
-  cricket::TransportChannel* voice_rtp_transport_channel() {
+  rtc::PacketTransportInterface* voice_rtp_transport_channel() {
     return rtp_transport_channel(voice_channel());
   }
 
-  cricket::TransportChannel* voice_rtcp_transport_channel() {
+  rtc::PacketTransportInterface* voice_rtcp_transport_channel() {
     return rtcp_transport_channel(voice_channel());
   }
 
-  cricket::TransportChannel* video_rtp_transport_channel() {
+  rtc::PacketTransportInterface* video_rtp_transport_channel() {
     return rtp_transport_channel(video_channel());
   }
 
-  cricket::TransportChannel* video_rtcp_transport_channel() {
+  rtc::PacketTransportInterface* video_rtcp_transport_channel() {
     return rtcp_transport_channel(video_channel());
   }
 
-  cricket::TransportChannel* data_rtp_transport_channel() {
+  rtc::PacketTransportInterface* data_rtp_transport_channel() {
     return rtp_transport_channel(data_channel());
   }
 
-  cricket::TransportChannel* data_rtcp_transport_channel() {
+  rtc::PacketTransportInterface* data_rtcp_transport_channel() {
     return rtcp_transport_channel(data_channel());
   }
 
  private:
-  cricket::TransportChannel* rtp_transport_channel(cricket::BaseChannel* ch) {
+  rtc::PacketTransportInterface* rtp_transport_channel(
+      cricket::BaseChannel* ch) {
     if (!ch) {
       return nullptr;
     }
     return ch->transport_channel();
   }
 
-  cricket::TransportChannel* rtcp_transport_channel(cricket::BaseChannel* ch) {
+  rtc::PacketTransportInterface* rtcp_transport_channel(
+      cricket::BaseChannel* ch) {
     if (!ch) {
       return nullptr;
     }
     return ch->rtcp_transport_channel();
   }
 };
 
 class WebRtcSessionCreateSDPObserverForTest
     : public rtc::RefCountedObject<CreateSessionDescriptionObserver> {
  public:
@@ skipping 2140 matching lines @@
                              &sdp);
 
   SessionDescriptionInterface* modified_offer =
       CreateSessionDescription(JsepSessionDescription::kOffer, sdp, NULL);
 
   SetRemoteDescriptionWithoutError(modified_offer);
 
   SessionDescriptionInterface* answer = CreateAnswer();
   SetLocalDescriptionWithoutError(answer);
 
-  cricket::TransportChannel* voice_transport_channel =
+  rtc::PacketTransportInterface* voice_transport_channel =
       session_->voice_rtp_transport_channel();
   EXPECT_TRUE(voice_transport_channel != NULL);
-  EXPECT_EQ(voice_transport_channel->transport_name(), "audio_content_name");
-  cricket::TransportChannel* video_transport_channel =
+  EXPECT_EQ(voice_transport_channel->debug_name(),
+            "audio_content_name " +
+                std::to_string(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  rtc::PacketTransportInterface* video_transport_channel =
       session_->video_rtp_transport_channel();
   ASSERT_TRUE(video_transport_channel != NULL);
-  EXPECT_EQ(video_transport_channel->transport_name(), "video_content_name");
+  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
 // the send streams when no constraints have been set.
 TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
   Init();
   std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
@@ skipping 1902 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));