Reduce flakiness in EndToEnd probing tests.

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.
  */
 
@@ skipping 214 matching lines @@
 
   // TODO(brandtr): Update this when we support multistream protection.
   if (num_flexfec_streams > 0) {
     video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
     video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
     video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
   }
 }
 
 void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
-  RTC_DCHECK(video_receive_configs_.empty());
-  RTC_DCHECK(allocated_decoders_.empty());
+  video_receive_configs_.clear();
+  allocated_decoders_.clear();
   if (num_video_streams_ > 0) {
     RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
     VideoReceiveStream::Config video_config(rtcp_send_transport);
     video_config.rtp.remb = false;
     video_config.rtp.transport_cc = true;
     video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     for (const RtpExtension& extension : video_send_config_.rtp.extensions)
       video_config.rtp.extensions.push_back(extension);
     video_config.renderer = &fake_renderer_;
     for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
@@ skipping 277 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc