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Side by Side Diff: webrtc/api/webrtcsession_unittest.cc

Issue 1919523002: Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rename the functions. Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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1106 InitWithDtmfCodec(); 1106 InitWithDtmfCodec();
1107 } else { 1107 } else {
1108 Init(); 1108 Init();
1109 } 1109 }
1110 SendAudioVideoStream1(); 1110 SendAudioVideoStream1();
1111 CreateAndSetRemoteOfferAndLocalAnswer(); 1111 CreateAndSetRemoteOfferAndLocalAnswer();
1112 EXPECT_FALSE(session_->CanInsertDtmf("")); 1112 EXPECT_FALSE(session_->CanInsertDtmf(""));
1113 EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1)); 1113 EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
1114 } 1114 }
1115 1115
1116 bool ContainsVideoCodecWithName(const SessionDescriptionInterface* desc,
1117 const std::string& codec_name) {
1118 for (const auto& content : desc->description()->contents()) {
1119 if (static_cast<cricket::MediaContentDescription*>(content.description)
1120 ->type() == cricket::MEDIA_TYPE_VIDEO) {
1121 const auto* mdesc =
1122 static_cast<cricket::VideoContentDescription*>(content.description);
1123 for (const auto& codec : mdesc->codecs()) {
1124 if (codec.name == codec_name) {
1125 return true;
1126 }
1127 }
1128 }
1129 }
1130 return false;
1131 }
1116 // Helper class to configure loopback network and verify Best 1132 // Helper class to configure loopback network and verify Best
1117 // Connection using right IP protocol for TestLoopbackCall 1133 // Connection using right IP protocol for TestLoopbackCall
1118 // method. LoopbackNetworkManager applies firewall rules to block 1134 // method. LoopbackNetworkManager applies firewall rules to block
1119 // all ping traffic once ICE completed, and remove them to observe 1135 // all ping traffic once ICE completed, and remove them to observe
1120 // ICE reconnected again. This LoopbackNetworkConfiguration struct 1136 // ICE reconnected again. This LoopbackNetworkConfiguration struct
1121 // verifies the best connection is using the right IP protocol after 1137 // verifies the best connection is using the right IP protocol after
1122 // initial ICE convergences. 1138 // initial ICE convergences.
1123 1139
1124 class LoopbackNetworkConfiguration { 1140 class LoopbackNetworkConfiguration {
1125 public: 1141 public:
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4270 candidate1.set_address(rtc::SocketAddress("1.1.1.1", 5000)); 4286 candidate1.set_address(rtc::SocketAddress("1.1.1.1", 5000));
4271 candidate1.set_component(1); 4287 candidate1.set_component(1);
4272 JsepIceCandidate ice_candidate(kMediaContentName1, kMediaContentIndex1, 4288 JsepIceCandidate ice_candidate(kMediaContentName1, kMediaContentIndex1,
4273 candidate1); 4289 candidate1);
4274 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate)); 4290 EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate));
4275 4291
4276 answer = CreateAnswer(); 4292 answer = CreateAnswer();
4277 SetLocalDescriptionWithoutError(answer); 4293 SetLocalDescriptionWithoutError(answer);
4278 } 4294 }
4279 4295
4280 // Flaky on Win and Mac only. See webrtc:4943
4281 #if defined(WEBRTC_WIN) || defined(WEBRTC_MAC)
4282 #define MAYBE_TestRtxRemovedByCreateAnswer DISABLED_TestRtxRemovedByCreateAnswer
4283 #else
4284 #define MAYBE_TestRtxRemovedByCreateAnswer TestRtxRemovedByCreateAnswer
4285 #endif
4286 // Tests that RTX codec is removed from the answer when it isn't supported 4296 // Tests that RTX codec is removed from the answer when it isn't supported
4287 // by local side. 4297 // by local side.
4288 TEST_F(WebRtcSessionTest, MAYBE_TestRtxRemovedByCreateAnswer) { 4298 TEST_F(WebRtcSessionTest, TestRtxRemovedByCreateAnswer) {
4289 Init(); 4299 Init();
4290 SendAudioVideoStream1(); 4300 SendAudioVideoStream1();
4291 std::string offer_sdp(kSdpWithRtx); 4301 std::string offer_sdp(kSdpWithRtx);
4292 4302
4293 SessionDescriptionInterface* offer = 4303 SessionDescriptionInterface* offer =
4294 CreateSessionDescription(JsepSessionDescription::kOffer, offer_sdp, NULL); 4304 CreateSessionDescription(JsepSessionDescription::kOffer, offer_sdp, NULL);
4295 EXPECT_TRUE(offer->ToString(&offer_sdp)); 4305 EXPECT_TRUE(offer->ToString(&offer_sdp));
4296 4306
4297 // Offer SDP contains the RTX codec. 4307 // Offer SDP contains the RTX codec.
4298 EXPECT_TRUE(offer_sdp.find("rtx") != std::string::npos); 4308 EXPECT_TRUE(ContainsVideoCodecWithName(offer, "rtx"));
4299 SetRemoteDescriptionWithoutError(offer); 4309 SetRemoteDescriptionWithoutError(offer);
4300 4310
4301 SessionDescriptionInterface* answer = CreateAnswer(); 4311 SessionDescriptionInterface* answer = CreateAnswer();
4302 std::string answer_sdp; 4312 // Answer SDP does not contain the RTX codec.
4303 answer->ToString(&answer_sdp); 4313 EXPECT_FALSE(ContainsVideoCodecWithName(answer, "rtx"));
4304 // Answer SDP removes the unsupported RTX codec.
4305 EXPECT_TRUE(answer_sdp.find("rtx") == std::string::npos);
4306 SetLocalDescriptionWithoutError(answer); 4314 SetLocalDescriptionWithoutError(answer);
4307 } 4315 }
4308 4316
4309 // This verifies that the voice channel after bundle has both options from video 4317 // This verifies that the voice channel after bundle has both options from video
4310 // and voice channels. 4318 // and voice channels.
4311 TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) { 4319 TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
4312 InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced); 4320 InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
4313 SendAudioVideoStream1(); 4321 SendAudioVideoStream1();
4314 4322
4315 PeerConnectionInterface::RTCOfferAnswerOptions options; 4323 PeerConnectionInterface::RTCOfferAnswerOptions options;
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4396 } 4404 }
4397 4405
4398 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test 4406 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test
4399 // currently fails because upon disconnection and reconnection OnIceComplete is 4407 // currently fails because upon disconnection and reconnection OnIceComplete is
4400 // called more than once without returning to IceGatheringGathering. 4408 // called more than once without returning to IceGatheringGathering.
4401 4409
4402 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests, 4410 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
4403 WebRtcSessionTest, 4411 WebRtcSessionTest,
4404 testing::Values(ALREADY_GENERATED, 4412 testing::Values(ALREADY_GENERATED,
4405 DTLS_IDENTITY_STORE)); 4413 DTLS_IDENTITY_STORE));
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