Add task queue to Call.

webrtc/call/call_perf_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 654 matching lines @@
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       encoder_config->streams[0].min_bitrate_bps = 50000;
       encoder_config->streams[0].target_bitrate_bps =
           encoder_config->streams[0].max_bitrate_bps = 2000000;
 
-      encoder_config_ = *encoder_config;
+      encoder_config_ = encoder_config->Copy();
     }
 
     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }
 
     void PerformTest() override {
       ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
           << "Timed out before receiving an initial high bitrate.";
       encoder_config_.streams[0].width *= 2;
       encoder_config_.streams[0].height *= 2;
-      send_stream_->ReconfigureVideoEncoder(encoder_config_);
+      send_stream_->ReconfigureVideoEncoder(encoder_config_.Copy());
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for a couple of high bitrate estimates "
              "after reconfiguring the send stream.";
     }
 
    private:
     rtc::Event time_to_reconfigure_;
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc