Chromium Code Reviews Chromium Code Reviews
Issue 1246913005:
TransportController refactoring (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master| Index: talk/app/webrtc/webrtcsession_unittest.cc |
| diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc |
| index 2c52d1301e515c47f1dcc366df1c9c3ffcbc4d8c..4ff21b630883399ea60d79d9026b145dc0f08b0c 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); |
| @@ -968,15 +1003,10 @@ 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 +1031,7 @@ class WebRtcSessionTest |
| // initial ICE convergences. |
| class LoopbackNetworkConfiguration { |
| - public: |
| + public: |
| LoopbackNetworkConfiguration() |
| : test_ipv6_network_(false), |
| test_extra_ipv4_network_(false), |
| @@ -1146,11 +1176,8 @@ 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); |
| @@ -1220,7 +1247,8 @@ class WebRtcSessionTest |
| } |
| void VerifyMultipleAsyncCreateDescriptionAfterInit( |
| - bool success, CreateSessionDescriptionRequest::Type type) { |
| + bool success, |
| + CreateSessionDescriptionRequest::Type type) { |
| CHECK(session_); |
| SetFactoryDtlsSrtp(); |
| if (type == CreateSessionDescriptionRequest::kAnswer) { |
| @@ -1704,15 +1732,14 @@ 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. |
| @@ -2014,7 +2041,7 @@ 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(); |
| @@ -2026,7 +2053,7 @@ 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. |
| @@ -2070,23 +2097,17 @@ 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); |
| } |
| @@ -2128,8 +2149,14 @@ 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->content_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->content_name(), "video_content_name"); |
| EXPECT_TRUE((video_channel_ = media_engine_->GetVideoChannel(0)) != NULL); |
| EXPECT_TRUE((voice_channel_ = media_engine_->GetVoiceChannel(0)) != NULL); |
| } |
| @@ -2689,20 +2716,23 @@ 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; |
| + } |
| } |
| } |
| } |
| @@ -2755,7 +2785,7 @@ 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(); |
| @@ -2766,19 +2796,19 @@ TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) { |
| SessionDescriptionInterface* offer = CreateOffer(options); |
| SetLocalDescriptionWithoutError(offer); |
| - EXPECT_NE(session_->GetTransportProxy("audio")->impl(), |
| - session_->GetTransportProxy("video")->impl()); |
| + EXPECT_NE(session_->voice_channel()->transport_channel(), |
|
pthatcher1
2015/08/31 22:01:35
Why not use voice voice_rtp_transport_channel() an
Taylor Brandstetter
2015/09/01 23:53:30
We should; done.
|
| + session_->video_channel()->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_channel()->transport_channel(), |
| + session_->video_channel()->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(); |
| @@ -2789,8 +2819,8 @@ TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) { |
| SessionDescriptionInterface* offer = CreateOffer(options); |
| SetLocalDescriptionWithoutError(offer); |
| - EXPECT_NE(session_->GetTransportProxy("audio")->impl(), |
| - session_->GetTransportProxy("video")->impl()); |
| + EXPECT_NE(session_->voice_channel()->transport_channel(), |
| + session_->video_channel()->transport_channel()); |
| mediastream_signaling_.SendAudioVideoStream2(); |
| @@ -2804,8 +2834,8 @@ 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_channel()->transport_channel(), |
| + session_->video_channel()->transport_channel()); |
| } |
| // kBundlePolicyMaxBundle policy with BUNDLE in the answer. |
| @@ -2819,16 +2849,51 @@ TEST_F(WebRtcSessionTest, TestMaxBundleBundleInAnswer) { |
| SessionDescriptionInterface* offer = CreateOffer(options); |
| SetLocalDescriptionWithoutError(offer); |
| - EXPECT_EQ(session_->GetTransportProxy("audio")->impl(), |
| - session_->GetTransportProxy("video")->impl()); |
| + EXPECT_EQ(session_->voice_channel()->transport_channel(), |
| + session_->video_channel()->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); |
| + |
| + std::string voice_content_name = session_->voice_channel()->content_name(); |
| + std::string video_content_name = session_->video_channel()->content_name(); |
| + EXPECT_EQ(session_->voice_channel()->transport_channel(), |
| + session_->video_channel()->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_channel()->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. |
| @@ -2842,8 +2907,8 @@ TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInAnswer) { |
| SessionDescriptionInterface* offer = CreateOffer(options); |
| SetLocalDescriptionWithoutError(offer); |
| - EXPECT_EQ(session_->GetTransportProxy("audio")->impl(), |
| - session_->GetTransportProxy("video")->impl()); |
| + EXPECT_EQ(session_->voice_channel()->transport_channel(), |
| + session_->video_channel()->transport_channel()); |
| mediastream_signaling_.SendAudioVideoStream2(); |
| @@ -2857,8 +2922,8 @@ 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_channel()->transport_channel(), |
| + session_->video_channel()->transport_channel()); |
| } |
| // kBundlePolicyMaxCompat bundle policy and answer contains BUNDLE. |
| @@ -2872,8 +2937,8 @@ TEST_F(WebRtcSessionTest, TestMaxCompatBundleInAnswer) { |
| SessionDescriptionInterface* offer = CreateOffer(options); |
| SetLocalDescriptionWithoutError(offer); |
| - EXPECT_NE(session_->GetTransportProxy("audio")->impl(), |
| - session_->GetTransportProxy("video")->impl()); |
| + EXPECT_NE(session_->voice_channel()->transport_channel(), |
| + session_->video_channel()->transport_channel()); |
| mediastream_signaling_.SendAudioVideoStream2(); |
| SessionDescriptionInterface* answer = |
| @@ -2882,11 +2947,11 @@ 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_channel()->transport_channel(), |
| + session_->video_channel()->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(); |
| @@ -2896,8 +2961,8 @@ TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) { |
| SessionDescriptionInterface* offer = CreateOffer(options); |
| SetLocalDescriptionWithoutError(offer); |
| - EXPECT_NE(session_->GetTransportProxy("audio")->impl(), |
| - session_->GetTransportProxy("video")->impl()); |
| + EXPECT_NE(session_->voice_channel()->transport_channel(), |
| + session_->video_channel()->transport_channel()); |
| mediastream_signaling_.SendAudioVideoStream2(); |
| @@ -2911,8 +2976,8 @@ 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_channel()->transport_channel(), |
| + session_->video_channel()->transport_channel()); |
| } |
| // kBundlePolicyMaxbundle and then we call SetRemoteDescription first. |
| @@ -2926,8 +2991,8 @@ 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) { |
| @@ -2938,16 +3003,16 @@ 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) { |
| @@ -2958,16 +3023,16 @@ 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 |
| @@ -2988,11 +3053,11 @@ 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); |
| @@ -3255,8 +3320,8 @@ 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); |
| @@ -3265,8 +3330,6 @@ 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); |
| } |
| @@ -3301,7 +3364,7 @@ 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()); |
| @@ -3654,8 +3717,8 @@ 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 |
| @@ -3942,6 +4005,7 @@ 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)); |
