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Unified Diff: talk/app/webrtc/webrtcsession_unittest.cc

Issue 1246913005: TransportController refactoring (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: More renaming, formatting and other polish Created 5 years, 4 months ago
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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..2bdaf36ae82e8f2c30957f1dc71c39a6667b6191 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->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);
}
@@ -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_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();
@@ -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_rtp_transport_channel(),
+ session_->video_rtp_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_rtp_transport_channel(),
+ session_->video_rtp_transport_channel());
}
// kBundlePolicyMaxBundle policy with BUNDLE in the answer.
@@ -2819,16 +2849,49 @@ 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.
@@ -2842,8 +2905,8 @@ 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();
@@ -2857,8 +2920,8 @@ 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_rtp_transport_channel(),
+ session_->video_rtp_transport_channel());
}
// kBundlePolicyMaxCompat bundle policy and answer contains BUNDLE.
@@ -2872,8 +2935,8 @@ 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 =
@@ -2882,11 +2945,11 @@ 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();
@@ -2896,8 +2959,8 @@ 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();
@@ -2911,8 +2974,8 @@ 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.
@@ -2926,8 +2989,8 @@ 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 +3001,16 @@ 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 +3021,16 @@ 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 +3051,11 @@ 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 +3318,8 @@ 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 +3328,6 @@ 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 +3362,7 @@ 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 +3715,8 @@ 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
@@ -3882,31 +3943,31 @@ 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);
}
@@ -3942,6 +4003,7 @@ 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));
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