Remove contention between RTCP packets and encoding.

webrtc/video/rampup_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.
  */
 
@@ skipping 255 matching lines @@
   ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, "bytes");
   if (ramp_up_finished_ms_ >= 0) {
     ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_,
                  "milliseconds");
   }
 }
 
 void RampUpTester::PerformTest() {
   test_start_ms_ = clock_->TimeInMilliseconds();
   poller_thread_->Start();
-  if (Wait() != kEventSignaled) {
-    printf("Timed out while waiting for ramp-up to complete.");
-    return;
-  }
+  EXPECT_EQ(kEventSignaled, Wait())
+      << "Timed out while waiting for ramp-up to complete.";
   TriggerTestDone();
   poller_thread_->Stop();
 }
 
 RampUpDownUpTester::RampUpDownUpTester(size_t num_streams,
                                        unsigned int start_bitrate_bps,
                                        const std::string& extension_type,
                                        bool rtx,
                                        bool red)
     : RampUpTester(num_streams, start_bitrate_bps, extension_type, rtx, red),
@@ skipping 210 matching lines @@
   RampUpTester test(3, 0, RtpExtension::kTransportSequenceNumber, true, true);
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTransportSequenceNumber, false, false);
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 }  // namespace webrtc