Reland: Fix race / crash in OnNetworkRouteChanged().

webrtc/video/video_send_stream_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>  // max
@@ skipping 1117 matching lines @@
     std::unique_ptr<RtpRtcp> rtp_rtcp_;
     std::unique_ptr<internal::TransportAdapter> feedback_transport_;
     RateLimiter retranmission_rate_limiter_;
     VideoSendStream* stream_;
     bool bitrate_capped_;
   } test;
 
   RunBaseTest(&test);
 }
 
-TEST_F(VideoSendStreamTest, DISABLED_ChangingNetworkRoute) {
+TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
+  static const int kStartBitrateBps = 300000;
+  static const int kNewMaxBitrateBps = 1234567;
+  static const uint8_t kExtensionId = 13;
   class ChangingNetworkRouteTest : public test::EndToEndTest {
    public:
-    const int kStartBitrateBps = 300000;
-    const int kNewMaxBitrateBps = 1234567;
-
     ChangingNetworkRouteTest()
-        : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
-          call_(nullptr) {}
+        : EndToEndTest(test::CallTest::kDefaultTimeoutMs), call_(nullptr) {
+      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+          kRtpExtensionTransportSequenceNumber, kExtensionId));
+    }
 
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       call_ = sender_call;
     }
 
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+      (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+      (*receive_configs)[0].rtp.transport_cc = true;
+    }
+
+    void ModifyAudioConfigs(
+        AudioSendStream::Config* send_config,
+        std::vector<AudioReceiveStream::Config>* receive_configs) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+      (*receive_configs)[0].rtp.extensions.clear();
+      (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+      (*receive_configs)[0].rtp.transport_cc = true;
+    }
+
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (call_->GetStats().send_bandwidth_bps > kStartBitrateBps) {
         observation_complete_.Set();
       }
 
       return SEND_PACKET;
     }
 
     void PerformTest() override {
       rtc::NetworkRoute new_route(true, 10, 20, -1);
@@ skipping 1435 matching lines @@
         observation_complete_.Set();
       }
     }
   } test;
 
   RunBaseTest(&test);
 }
 #endif  // !defined(RTC_DISABLE_VP9)
 
 }  // namespace webrtc