Deflake ChangingNetworkRoute test.

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 1256 matching lines @@
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for start bitrate to be exceeded.";
 
       bitrate_config.start_bitrate_bps = -1;
       bitrate_config.max_bitrate_bps = kNewMaxBitrateBps;
       call_->SetBitrateConfig(bitrate_config);
       // TODO(holmer): We should set the last sent packet id here and verify
       // that we correctly ignore any packet loss reported prior to that id.
       ++new_route.local_network_id;
       call_->OnNetworkRouteChanged("transport", new_route);
-      EXPECT_EQ(kStartBitrateBps, call_->GetStats().send_bandwidth_bps);
+      EXPECT_GT(call_->GetStats().send_bandwidth_bps, kStartBitrateBps);

[the sun, 2016/10/19 07:46:07]

not _GE?

[stefan-webrtc, 2016/10/25 10:37:48]

No, zero is invalid and is what we accidentally set it to in a previous bug.

[stefan-webrtc, 2016/10/25 10:58:07]

I misread your comment. Sorry... Yes, it should have been GE... :)

     }
 
    private:
     Call* call_;
   } test;
 
   RunBaseTest(&test);
 }
 
 class MaxPaddingSetTest : public test::SendTest {
@@ skipping 1600 matching lines @@
         observation_complete_.Set();
       }
     }
   } test;
 
   RunBaseTest(&test);
 }
 #endif  // !defined(RTC_DISABLE_VP9)
 
 }  // namespace webrtc