Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer.

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 1095 matching lines @@
       InitWithDtmfCodec();
     } else {
       Init();
     }
     SendAudioVideoStream1();
     CreateAndSetRemoteOfferAndLocalAnswer();
     EXPECT_FALSE(session_->CanInsertDtmf(""));
     EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
   }
 
+  bool DescriptionContainsRtxCodec(const SessionDescriptionInterface* desc) {
+    for (const auto& content : desc->description()->contents()) {
+      if (static_cast<cricket::MediaContentDescription*>(content.description)
+              ->type() == cricket::MEDIA_TYPE_VIDEO) {
+        const auto* mdesc =
+            static_cast<cricket::VideoContentDescription*>(content.description);
+        for (const auto& codec : mdesc->codecs()) {
+          if (codec.name == "rtx") {
+            return true;
+          }
+        }
+      }
+    }
+    return false;
+  }
   // Helper class to configure loopback network and verify Best
   // Connection using right IP protocol for TestLoopbackCall
   // method. LoopbackNetworkManager applies firewall rules to block
   // all ping traffic once ICE completed, and remove them to observe
   // ICE reconnected again. This LoopbackNetworkConfiguration struct
   // verifies the best connection is using the right IP protocol after
   // initial ICE convergences.
 
   class LoopbackNetworkConfiguration {
    public:
@@ skipping 3151 matching lines @@
   candidate1.set_address(rtc::SocketAddress("1.1.1.1", 5000));
   candidate1.set_component(1);
   JsepIceCandidate ice_candidate(kMediaContentName1, kMediaContentIndex1,
                                  candidate1);
   EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate));
 
   answer = CreateAnswer();
   SetLocalDescriptionWithoutError(answer);
 }
 
-// Flaky on Win and Mac only. See webrtc:4943
-#if defined(WEBRTC_WIN) || defined(WEBRTC_MAC)
-#define MAYBE_TestRtxRemovedByCreateAnswer DISABLED_TestRtxRemovedByCreateAnswer
-#else
-#define MAYBE_TestRtxRemovedByCreateAnswer TestRtxRemovedByCreateAnswer
-#endif
 // Tests that RTX codec is removed from the answer when it isn't supported
 // by local side.
-TEST_F(WebRtcSessionTest, MAYBE_TestRtxRemovedByCreateAnswer) {
+TEST_F(WebRtcSessionTest, TestRtxRemovedByCreateAnswer) {
   Init();
   SendAudioVideoStream1();
   std::string offer_sdp(kSdpWithRtx);
 
   SessionDescriptionInterface* offer =
       CreateSessionDescription(JsepSessionDescription::kOffer, offer_sdp, NULL);
   EXPECT_TRUE(offer->ToString(&offer_sdp));
 
   // Offer SDP contains the RTX codec.
-  EXPECT_TRUE(offer_sdp.find("rtx") != std::string::npos);
+  EXPECT_TRUE(DescriptionContainsRtxCodec(offer));
   SetRemoteDescriptionWithoutError(offer);
 
   SessionDescriptionInterface* answer = CreateAnswer();
-  std::string answer_sdp;
-  answer->ToString(&answer_sdp);
-  // Answer SDP removes the unsupported RTX codec.
-  EXPECT_TRUE(answer_sdp.find("rtx") == std::string::npos);
   SetLocalDescriptionWithoutError(answer);
+  // Answer SDP does not contain the RTX codec.
+  EXPECT_FALSE(DescriptionContainsRtxCodec(answer));

[Taylor Brandstetter, 2016/04/23 01:25:40]

nit: Can you move this back to before "SetLocalDescriptionWithoutError" for
consistency? Also, "SetLocalDescription" takes ownership of the
SessionDescription passed into it, so technically it's not safe to use
afterwards.

 }
 
 // This verifies that the voice channel after bundle has both options from video
 // and voice channels.
 TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
   SendAudioVideoStream1();
 
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.use_rtp_mux = true;
@@ skipping 79 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));