Add task queue to Call.

webrtc/call/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 155 matching lines @@
           is_sending_receiving_(false),
           send_stream_(nullptr),
           audio_receive_stream_(nullptr),
           video_receive_stream_(nullptr),
           frame_generator_capturer_(),
           fake_encoder_(Clock::GetRealTimeClock()),
           fake_decoder_() {
       test_->video_send_config_.rtp.ssrcs[0]++;
       test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
       send_stream_ = test_->sender_call_->CreateVideoSendStream(
-          test_->video_send_config_, test_->video_encoder_config_);
+          test_->video_send_config_.Copy(),
+          test_->video_encoder_config_.Copy());
       RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
           send_stream_->Input(), test_->video_encoder_config_.streams[0].width,
           test_->video_encoder_config_.streams[0].height, 30,
           Clock::GetRealTimeClock()));
       send_stream_->Start();
       frame_generator_capturer_->Start();
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
@@ skipping 138 matching lines @@
       RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc