Fix capture ntp time issue introduced with r11187.

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.
  */
 #include <algorithm>
@@ skipping 387 matching lines @@
           capturer_(nullptr),
           rtp_start_timestamp_set_(false),
           rtp_start_timestamp_(0) {}
 
    private:
     test::PacketTransport* CreateSendTransport(Call* sender_call) override {
       return new test::PacketTransport(
           sender_call, this, test::PacketTransport::kSender, net_config_);
     }
 
+    test::PacketTransport* CreateReceiveTransport() override {
+      return new test::PacketTransport(
+          nullptr, this, test::PacketTransport::kReceiver, net_config_);
+    }
+
     void RenderFrame(const VideoFrame& video_frame,
                      int time_to_render_ms) override {
       rtc::CritScope lock(&crit_);
       if (video_frame.ntp_time_ms() <= 0) {
         // Haven't got enough RTCP SR in order to calculate the capture ntp
         // time.
         return;
       }
 
       int64_t now_ms = clock_->TimeInMilliseconds();
@@ skipping 327 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc