Fix race / crash in OnNetworkRouteChanged().

webrtc/video/video_send_stream_tests.cc

Index: webrtc/video/video_send_stream_tests.cc
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index ba15b17c85b229604bd7ba082e836b00ac54fc4f..04b954ac12ef0d47e15c1972ccf2ce09e700b7e2 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1136,19 +1136,43 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
 }
 
 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;
+    }

[stefan-webrtc, 2016/09/27 10:58:55]

Changes to enable send-side BWE for this test. This triggered the crash I'm
fixing.

+
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (call_->GetStats().send_bandwidth_bps > kStartBitrateBps) {
         observation_complete_.Set();