RembWithSendSideBwe: Rename |event_| to |stop_event_| and set it when the test ends.

webrtc/video/end_to_end_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>
@@ skipping 2077 matching lines @@
 TEST_P(EndToEndTest, RembWithSendSideBwe) {
   class BweObserver : public test::EndToEndTest {
    public:
     BweObserver()
         : EndToEndTest(kDefaultTimeoutMs),
           sender_call_(nullptr),
           clock_(Clock::GetRealTimeClock()),
           sender_ssrc_(0),
           remb_bitrate_bps_(1000000),
           receive_transport_(nullptr),
-          event_(false, false),
+          stop_event_(false, false),
           poller_thread_(&BitrateStatsPollingThread,
                          this,
                          "BitrateStatsPollingThread"),
           state_(kWaitForFirstRampUp),
           retransmission_rate_limiter_(clock_, 1000) {}
 
     ~BweObserver() {}
 
     test::PacketTransport* CreateReceiveTransport() override {
       receive_transport_ = new test::PacketTransport(
@@ skipping 67 matching lines @@
             break;
 
           case kWaitForSecondRampUp:
             if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
               observation_complete_.Set();
             }
             break;
         }
       }
 
-      return !event_.Wait(1000);
+      return !stop_event_.Wait(1000);
     }
 
     void PerformTest() override {
       poller_thread_.Start();
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for bitrate to change according to REMB.";
+      stop_event_.Set();
       poller_thread_.Stop();
     }
 
    private:
     enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp };
 
     Call* sender_call_;
     Clock* const clock_;
     uint32_t sender_ssrc_;
     int remb_bitrate_bps_;
     std::unique_ptr<RtpRtcp> rtp_rtcp_;
     test::PacketTransport* receive_transport_;
-    rtc::Event event_;
+    rtc::Event stop_event_;
     rtc::PlatformThread poller_thread_;
     TestState state_;
     RateLimiter retransmission_rate_limiter_;
   } test;
 
   RunBaseTest(&test);
 }
 
 TEST_P(EndToEndTest, VerifyNackStats) {
   static const int kPacketNumberToDrop = 200;
@@ skipping 1973 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc