TransportController refactoring.

talk/app/webrtc/webrtcsession_unittest.cc

Index: talk/app/webrtc/webrtcsession_unittest.cc
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index 8bd97e59ce5f6b67a681ada2454cefada0065e21..6dc0fc71cd753b3ad894e30e1463d7df35f6b77e 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -25,6 +25,8 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+#include <vector>
+
 #include "talk/app/webrtc/audiotrack.h"
 #include "talk/app/webrtc/fakemetricsobserver.h"
 #include "talk/app/webrtc/jsepicecandidate.h"
@@ -163,8 +165,8 @@ static void InjectAfter(const std::string& line,
                         const std::string& newlines,
                         std::string* message) {
   const std::string tmp = line + newlines;
-  rtc::replace_substrs(line.c_str(), line.length(),
-                       tmp.c_str(), tmp.length(), message);
+  rtc::replace_substrs(line.c_str(), line.length(), tmp.c_str(), tmp.length(),
+                       message);
 }
 
 class MockIceObserver : public webrtc::IceObserver {
@@ -244,12 +246,52 @@ class WebRtcSessionForTest : public webrtc::WebRtcSession {
   }
   virtual ~WebRtcSessionForTest() {}
 
-  using cricket::BaseSession::GetTransportProxy;
+  // 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() {
+    return rtp_transport_channel(voice_channel());
+  }
+
+  cricket::TransportChannel* voice_rtcp_transport_channel() {
+    return rtcp_transport_channel(voice_channel());
+  }
+
+  cricket::TransportChannel* video_rtp_transport_channel() {
+    return rtp_transport_channel(video_channel());
+  }
+
+  cricket::TransportChannel* video_rtcp_transport_channel() {
+    return rtcp_transport_channel(video_channel());
+  }
+
+  cricket::TransportChannel* data_rtp_transport_channel() {
+    return rtp_transport_channel(data_channel());
+  }
+
+  cricket::TransportChannel* data_rtcp_transport_channel() {
+    return rtcp_transport_channel(data_channel());
+  }
+
   using webrtc::WebRtcSession::SetAudioPlayout;
   using webrtc::WebRtcSession::SetAudioSend;
   using webrtc::WebRtcSession::SetCaptureDevice;
   using webrtc::WebRtcSession::SetVideoPlayout;
   using webrtc::WebRtcSession::SetVideoSend;
+
+ private:
+  cricket::TransportChannel* rtp_transport_channel(cricket::BaseChannel* ch) {
+    if (!ch) {
+      return nullptr;
+    }
+    return ch->transport_channel();
+  }
+
+  cricket::TransportChannel* rtcp_transport_channel(cricket::BaseChannel* ch) {
+    if (!ch) {
+      return nullptr;
+    }
+    return ch->rtcp_transport_channel();
+  }
 };
 
 class WebRtcSessionCreateSDPObserverForTest
@@ -375,9 +417,9 @@ class WebRtcSessionTest
     EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
         observer_.ice_gathering_state_);
 
-    EXPECT_TRUE(session_->Initialize(
-        options_, constraints_.get(), dtls_identity_store.Pass(),
-        rtc_configuration));
+    EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
+                                     dtls_identity_store.Pass(),
+                                     rtc_configuration));
     session_->set_metrics_observer(metrics_observer_);
   }
 
@@ -490,13 +532,6 @@ class WebRtcSessionTest
             session_->video_channel() != NULL);
   }
 
-  void CheckTransportChannels() const {
-    EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 1) != NULL);
-    EXPECT_TRUE(session_->GetChannel(cricket::CN_AUDIO, 2) != NULL);
-    EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 1) != NULL);
-    EXPECT_TRUE(session_->GetChannel(cricket::CN_VIDEO, 2) != NULL);
-  }
-
   void VerifyCryptoParams(const cricket::SessionDescription* sdp) {
     ASSERT_TRUE(session_.get() != NULL);
     const cricket::ContentInfo* content = cricket::GetFirstAudioContent(sdp);
@@ -722,6 +757,7 @@ class WebRtcSessionTest
   }
   void SetLocalDescriptionWithoutError(SessionDescriptionInterface* desc) {
     EXPECT_TRUE(session_->SetLocalDescription(desc, NULL));
+    session_->MaybeStartGathering();
   }
   void SetLocalDescriptionExpectState(SessionDescriptionInterface* desc,
                                       BaseSession::State expected_state) {
@@ -968,15 +1004,10 @@ class WebRtcSessionTest
     SetRemoteDescriptionWithoutError(new_answer);
     EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
     EXPECT_EQ(expected_candidate_num, observer_.mline_0_candidates_.size());
-    EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
-    for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
-      cricket::Candidate c0 = observer_.mline_0_candidates_[i];
-      cricket::Candidate c1 = observer_.mline_1_candidates_[i];
-      if (bundle) {
-        EXPECT_TRUE(c0.IsEquivalent(c1));
-      } else {
-        EXPECT_FALSE(c0.IsEquivalent(c1));
-      }
+    if (bundle) {
+      EXPECT_EQ(0, observer_.mline_1_candidates_.size());
+    } else {
+      EXPECT_EQ(expected_candidate_num, observer_.mline_1_candidates_.size());
     }
   }
   // Tests that we can only send DTMF when the dtmf codec is supported.
@@ -1001,7 +1032,7 @@ class WebRtcSessionTest
   // initial ICE convergences.
 
   class LoopbackNetworkConfiguration {
-  public:
+   public:
     LoopbackNetworkConfiguration()
         : test_ipv6_network_(false),
           test_extra_ipv4_network_(false),
@@ -1150,11 +1181,8 @@ class WebRtcSessionTest
 
     // Clearing the rules, session should move back to completed state.
     loopback_network_manager.ClearRules(fss_.get());
-    // Session is automatically calling OnSignalingReady after creation of
-    // new portallocator session which will allocate new set of candidates.
 
     LOG(LS_INFO) << "Firewall Rules cleared";
-
     EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                    observer_.ice_connection_state_,
                    kIceCandidatesTimeout);
@@ -1707,15 +1735,14 @@ TEST_P(WebRtcSessionTest, TestSetLocalNonDtlsAnswerWhenDtlsOn) {
 // a DTLS fingerprint when DTLS is required.
 TEST_P(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  // Enable both SDES and DTLS, so that offer won't be outright rejected as a
-  // result of using the "UDP/TLS/RTP/SAVPF" profile.
   InitWithDtls(GetParam());
-  session_->SetSdesPolicy(cricket::SEC_ENABLED);
   SessionDescriptionInterface* offer = CreateOffer();
   cricket::MediaSessionOptions options;
   options.recv_video = true;
+  rtc::scoped_ptr<SessionDescriptionInterface> temp_offer(
+      CreateRemoteOffer(options, cricket::SEC_ENABLED));
   JsepSessionDescription* answer =
-      CreateRemoteAnswer(offer, options, cricket::SEC_ENABLED);
+      CreateRemoteAnswer(temp_offer.get(), options, cricket::SEC_ENABLED);
 
   // SetRemoteDescription and SetLocalDescription will take the ownership of
   // the offer and answer.
@@ -2017,7 +2044,7 @@ TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
   EXPECT_LT(0u, candidates->count());
   candidates = local_desc->candidates(1);
   ASSERT_TRUE(candidates != NULL);
-  EXPECT_LT(0u, candidates->count());
+  EXPECT_EQ(0u, candidates->count());
 
   // Update the session descriptions.
   mediastream_signaling_.SendAudioVideoStream1();
@@ -2029,7 +2056,7 @@ TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
   EXPECT_LT(0u, candidates->count());
   candidates = local_desc->candidates(1);
   ASSERT_TRUE(candidates != NULL);
-  EXPECT_LT(0u, candidates->count());
+  EXPECT_EQ(0u, candidates->count());
 }
 
 // Test that we can set a remote session description with remote candidates.
@@ -2073,23 +2100,17 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
   // Wait until at least one local candidate has been collected.
   EXPECT_TRUE_WAIT(0u < observer_.mline_0_candidates_.size(),
                    kIceCandidatesTimeout);
-  EXPECT_TRUE_WAIT(0u < observer_.mline_1_candidates_.size(),
-                   kIceCandidatesTimeout);
 
   rtc::scoped_ptr<SessionDescriptionInterface> local_offer(CreateOffer());
 
   ASSERT_TRUE(local_offer->candidates(kMediaContentIndex0) != NULL);
   EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex0)->count());
-  ASSERT_TRUE(local_offer->candidates(kMediaContentIndex1) != NULL);
-  EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex1)->count());
 
   SessionDescriptionInterface* remote_offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(remote_offer);
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   ASSERT_TRUE(answer->candidates(kMediaContentIndex0) != NULL);
   EXPECT_LT(0u, answer->candidates(kMediaContentIndex0)->count());
-  ASSERT_TRUE(answer->candidates(kMediaContentIndex1) != NULL);
-  EXPECT_LT(0u, answer->candidates(kMediaContentIndex1)->count());
   SetLocalDescriptionWithoutError(answer);
 }
 
@@ -2131,8 +2152,14 @@ TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
       CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
 
-  EXPECT_TRUE(session_->GetTransportProxy("audio_content_name") != NULL);
-  EXPECT_TRUE(session_->GetTransportProxy("video_content_name") != NULL);
+  cricket::TransportChannel* 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 =
+      session_->video_rtp_transport_channel();
+  EXPECT_TRUE(video_transport_channel != NULL);
+  EXPECT_EQ(video_transport_channel->transport_name(), "video_content_name");
   EXPECT_TRUE((video_channel_ = media_engine_->GetVideoChannel(0)) != NULL);
   EXPECT_TRUE((voice_channel_ = media_engine_->GetVoiceChannel(0)) != NULL);
 }
@@ -2692,20 +2719,23 @@ TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) {
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
-  cricket::Transport* t = session_->GetTransport("audio");
+  cricket::BaseChannel* voice_channel = session_->voice_channel();
+  ASSERT(voice_channel != NULL);
 
   // Checks if one of the transport channels contains a connection using a given
   // port.
-  auto connection_with_remote_port = [t](int port) {
-    cricket::TransportStats stats;
-    t->GetStats(&stats);
-    for (auto& chan_stat : stats.channel_stats) {
-      for (auto& conn_info : chan_stat.connection_infos) {
-        if (conn_info.remote_candidate.address().port() == port) {
-          return true;
+  auto connection_with_remote_port = [this, voice_channel](int port) {
+    cricket::SessionStats stats;
+    session_->GetChannelTransportStats(voice_channel, &stats);
+    for (auto& kv : stats.transport_stats) {
+      for (auto& chan_stat : kv.second.channel_stats) {
+        for (auto& conn_info : chan_stat.connection_infos) {
+          if (conn_info.remote_candidate.address().port() == port) {
+            return true;
+          }
         }
       }
     }
@@ -2758,7 +2788,7 @@ TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) {
   EXPECT_FALSE(connection_with_remote_port(6000));
 }
 
-// kBundlePolicyBalanced bundle policy and answer contains BUNDLE.
+// kBundlePolicyBalanced BUNDLE policy and answer contains BUNDLE.
 TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
   mediastream_signaling_.SendAudioVideoStream1();
@@ -2769,19 +2799,19 @@ TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 }
 
-// kBundlePolicyBalanced bundle policy but no BUNDLE in the answer.
+// kBundlePolicyBalanced BUNDLE policy but no BUNDLE in the answer.
 TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
   mediastream_signaling_.SendAudioVideoStream1();
@@ -2792,8 +2822,8 @@ TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
 
@@ -2807,8 +2837,8 @@ TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) {
   modified_answer->Initialize(answer_copy, "1", "1");
   SetRemoteDescriptionWithoutError(modified_answer);  //
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyMaxBundle policy with BUNDLE in the answer.
@@ -2822,16 +2852,49 @@ TEST_F(WebRtcSessionTest, TestMaxBundleBundleInAnswer) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
+}
+
+// kBundlePolicyMaxBundle policy with BUNDLE in the answer, but no
+// audio content in the answer.
+TEST_F(WebRtcSessionTest, TestMaxBundleRejectAudio) {
+  InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
+  mediastream_signaling_.SendAudioVideoStream1();
+
+  PeerConnectionInterface::RTCOfferAnswerOptions options;
+  options.use_rtp_mux = true;
+
+  SessionDescriptionInterface* offer = CreateOffer(options);
+  SetLocalDescriptionWithoutError(offer);
+
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
+
+  mediastream_signaling_.SendAudioVideoStream2();
+  cricket::MediaSessionOptions recv_options;
+  recv_options.recv_audio = false;
+  recv_options.recv_video = true;
+  SessionDescriptionInterface* answer =
+      CreateRemoteAnswer(session_->local_description(), recv_options);
+  SetRemoteDescriptionWithoutError(answer);
+
+  EXPECT_TRUE(NULL == session_->voice_channel());
+  EXPECT_TRUE(NULL != session_->video_rtp_transport_channel());
+
+  session_->Terminate();
+  EXPECT_TRUE(NULL == session_->voice_rtp_transport_channel());
+  EXPECT_TRUE(NULL == session_->voice_rtcp_transport_channel());
+  EXPECT_TRUE(NULL == session_->video_rtp_transport_channel());
+  EXPECT_TRUE(NULL == session_->video_rtcp_transport_channel());
 }
 
 // kBundlePolicyMaxBundle policy but no BUNDLE in the answer.
@@ -2845,8 +2908,8 @@ TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInAnswer) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
 
@@ -2860,8 +2923,45 @@ TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInAnswer) {
   modified_answer->Initialize(answer_copy, "1", "1");
   SetRemoteDescriptionWithoutError(modified_answer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
+}
+
+// kBundlePolicyMaxBundle policy with BUNDLE in the remote offer.
+TEST_F(WebRtcSessionTest, TestMaxBundleBundleInRemoteOffer) {
+  InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
+  mediastream_signaling_.SendAudioVideoStream1();
+
+  SessionDescriptionInterface* offer = CreateRemoteOffer();
+  SetRemoteDescriptionWithoutError(offer);
+
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
+
+  mediastream_signaling_.SendAudioVideoStream2();
+  SessionDescriptionInterface* answer = CreateAnswer(nullptr);
+  SetLocalDescriptionWithoutError(answer);
+
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
+}
+
+// kBundlePolicyMaxBundle policy but no BUNDLE in the remote offer.
+TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInRemoteOffer) {
+  InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxBundle);
+  mediastream_signaling_.SendAudioVideoStream1();
+
+  // Remove BUNDLE from the offer.
+  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
+  cricket::SessionDescription* offer_copy = offer->description()->Copy();
+  offer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
+  JsepSessionDescription* modified_offer =
+      new JsepSessionDescription(JsepSessionDescription::kOffer);
+  modified_offer->Initialize(offer_copy, "1", "1");
+
+  // Expect an error when applying the remote description
+  SetRemoteDescriptionExpectError(JsepSessionDescription::kOffer,
+                                  kCreateChannelFailed, modified_offer);
 }
 
 // kBundlePolicyMaxCompat bundle policy and answer contains BUNDLE.
@@ -2875,8 +2975,8 @@ TEST_F(WebRtcSessionTest, TestMaxCompatBundleInAnswer) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
@@ -2885,11 +2985,11 @@ TEST_F(WebRtcSessionTest, TestMaxCompatBundleInAnswer) {
 
   // This should lead to an audio-only call but isn't implemented
   // correctly yet.
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 }
 
-// kBundlePolicyMaxCompat bundle policy but no BUNDLE in the answer.
+// kBundlePolicyMaxCompat BUNDLE policy but no BUNDLE in the answer.
 TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyMaxCompat);
   mediastream_signaling_.SendAudioVideoStream1();
@@ -2899,8 +2999,8 @@ TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
 
@@ -2914,8 +3014,8 @@ TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) {
   modified_answer->Initialize(answer_copy, "1", "1");
   SetRemoteDescriptionWithoutError(modified_answer);  //
 
-  EXPECT_NE(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 }
 
 // kBundlePolicyMaxbundle and then we call SetRemoteDescription first.
@@ -2929,8 +3029,8 @@ TEST_F(WebRtcSessionTest, TestMaxBundleWithSetRemoteDescriptionFirst) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetRemoteDescriptionWithoutError(offer);
 
-  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
-            session_->GetTransportProxy("video")->impl());
+  EXPECT_EQ(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 }
 
 TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
@@ -2941,16 +3041,16 @@ TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
-  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+  EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL);
+  EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL);
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
-  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+  EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL);
+  EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL);
 }
 
 TEST_F(WebRtcSessionTest, TestNegotiateRtcpMux) {
@@ -2961,16 +3061,16 @@ TEST_F(WebRtcSessionTest, TestNegotiateRtcpMux) {
   SessionDescriptionInterface* offer = CreateOffer(options);
   SetLocalDescriptionWithoutError(offer);
 
-  EXPECT_TRUE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
-  EXPECT_TRUE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+  EXPECT_TRUE(session_->voice_rtcp_transport_channel() != NULL);
+  EXPECT_TRUE(session_->video_rtcp_transport_channel() != NULL);
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
-  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+  EXPECT_TRUE(session_->voice_rtcp_transport_channel() == NULL);
+  EXPECT_TRUE(session_->video_rtcp_transport_channel() == NULL);
 }
 
 // This test verifies that SetLocalDescription and SetRemoteDescription fails
@@ -2991,11 +3091,11 @@ TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
   rtc::replace_substrs(rtcp_mux.c_str(), rtcp_mux.length(),
                              xrtcp_mux.c_str(), xrtcp_mux.length(),
                              &offer_str);
-  JsepSessionDescription *local_offer =
+  JsepSessionDescription* local_offer =
       new JsepSessionDescription(JsepSessionDescription::kOffer);
   EXPECT_TRUE((local_offer)->Initialize(offer_str, NULL));
   SetLocalDescriptionOfferExpectError(kBundleWithoutRtcpMux, local_offer);
-  JsepSessionDescription *remote_offer =
+  JsepSessionDescription* remote_offer =
       new JsepSessionDescription(JsepSessionDescription::kOffer);
   EXPECT_TRUE((remote_offer)->Initialize(offer_str, NULL));
   SetRemoteDescriptionOfferExpectError(kBundleWithoutRtcpMux, remote_offer);
@@ -3258,8 +3358,8 @@ TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
                                   candidate1);
   EXPECT_TRUE(offer->AddCandidate(&ice_candidate1));
   SetRemoteDescriptionWithoutError(offer);
-  ASSERT_TRUE(session_->GetTransportProxy("audio") != NULL);
-  ASSERT_TRUE(session_->GetTransportProxy("video") != NULL);
+  ASSERT_TRUE(session_->voice_rtp_transport_channel() != NULL);
+  ASSERT_TRUE(session_->video_rtp_transport_channel() != NULL);
 
   // Pump for 1 second and verify that no candidates are generated.
   rtc::Thread::Current()->ProcessMessages(1000);
@@ -3268,8 +3368,6 @@ TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
 
   SessionDescriptionInterface* answer = CreateAnswer(NULL);
   SetLocalDescriptionWithoutError(answer);
-  EXPECT_TRUE(session_->GetTransportProxy("audio")->negotiated());
-  EXPECT_TRUE(session_->GetTransportProxy("video")->negotiated());
   EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
 }
 
@@ -3304,7 +3402,7 @@ TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
   // will be set as per MediaSessionDescriptionFactory.
   std::string offer_str;
   offer->ToString(&offer_str);
-  SessionDescriptionInterface *jsep_offer_str =
+  SessionDescriptionInterface* jsep_offer_str =
       CreateSessionDescription(JsepSessionDescription::kOffer, offer_str, NULL);
   SetLocalDescriptionWithoutError(jsep_offer_str);
   EXPECT_FALSE(session_->voice_channel()->secure_required());
@@ -3657,8 +3755,8 @@ TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
 TEST_P(WebRtcSessionTest,
        TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  VerifyMultipleAsyncCreateDescription(
-      GetParam(), CreateSessionDescriptionRequest::kOffer);
+  VerifyMultipleAsyncCreateDescription(GetParam(),
+                                       CreateSessionDescriptionRequest::kOffer);
 }
 
 // Verifies that CreateOffer fails when Multiple CreateOffer calls are made
@@ -3881,31 +3979,31 @@ TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
                                        rtc::Socket::Option::OPT_RCVBUF, 8000);
 
   int option_val;
-  EXPECT_TRUE(session_->video_channel()->transport_channel()->GetOption(
+  EXPECT_TRUE(session_->video_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
   EXPECT_EQ(4000, option_val);
-  EXPECT_FALSE(session_->voice_channel()->transport_channel()->GetOption(
+  EXPECT_FALSE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
 
-  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+  EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
   EXPECT_EQ(8000, option_val);
-  EXPECT_FALSE(session_->video_channel()->transport_channel()->GetOption(
+  EXPECT_FALSE(session_->video_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
 
-  EXPECT_NE(session_->voice_channel()->transport_channel(),
-            session_->video_channel()->transport_channel());
+  EXPECT_NE(session_->voice_rtp_transport_channel(),
+            session_->video_rtp_transport_channel());
 
   mediastream_signaling_.SendAudioVideoStream2();
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
-  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+  EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_SNDBUF, &option_val));
   EXPECT_EQ(4000, option_val);
 
-  EXPECT_TRUE(session_->voice_channel()->transport_channel()->GetOption(
+  EXPECT_TRUE(session_->voice_rtp_transport_channel()->GetOption(
       rtc::Socket::Option::OPT_RCVBUF, &option_val));
   EXPECT_EQ(8000, option_val);
 }
@@ -3941,6 +4039,7 @@ TEST_F(WebRtcSessionTest, CreateOffersAndShutdown) {
 // 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));
+INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
+                        WebRtcSessionTest,
+                        testing::Values(ALREADY_GENERATED,
+                                        DTLS_IDENTITY_STORE));