Disable reduced-size RTCP in default config.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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 "webrtc/test/call_test.h"
@@ skipping 90 matching lines @@
     send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]);
   send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
 }
 
 void CallTest::CreateMatchingReceiveConfigs() {
   assert(!send_config_.rtp.ssrcs.empty());
   assert(receive_configs_.empty());
   assert(allocated_decoders_.empty());
   VideoReceiveStream::Config config;
+  config.rtp.remb = true;
   config.rtp.local_ssrc = kReceiverLocalSsrc;
   for (const RtpExtension& extension : send_config_.rtp.extensions)
     config.rtp.extensions.push_back(extension);
   for (size_t i = 0; i < send_config_.rtp.ssrcs.size(); ++i) {
     VideoReceiveStream::Decoder decoder =
         test::CreateMatchingDecoder(send_config_.encoder_settings);
     allocated_decoders_.push_back(decoder.decoder);
     config.decoders.clear();
     config.decoders.push_back(decoder);
     config.rtp.remote_ssrc = send_config_.rtp.ssrcs[i];
@@ skipping 108 matching lines @@
                            const FakeNetworkPipe::Config& config)
     : BaseTest(timeout_ms, config) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc