Disable reduced-size RTCP in default config.

webrtc/video/bitrate_estimator_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 <functional>
@@ skipping 133 matching lines @@
     // Encoders will be set separately per stream.
     send_config_.encoder_settings.encoder = nullptr;
     send_config_.encoder_settings.payload_name = "FAKE";
     send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
     encoder_config_.streams = test::CreateVideoStreams(1);
 
     receive_config_ = VideoReceiveStream::Config();
     // receive_config_.decoders will be set by every stream separately.
     receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
     receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
+    receive_config_.rtp.remb = true;
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
   }
 
   virtual void TearDown() {
     std::for_each(streams_.begin(), streams_.end(),
         std::mem_fun(&Stream::StopSending));
 
@@ skipping 205 matching lines @@
   receiver_trace_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_trace_.PushExpectedLogLine(
       "RemoteBitrateEstimatorFactory: Instantiating.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
 }
 }  // namespace webrtc