Fix test bug introduced in r11101.

webrtc/call/rampup_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 69 matching lines @@
   send_stream_ = send_stream;
 }
 
 test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
   send_transport_ = new test::PacketTransport(sender_call, this,
                                               test::PacketTransport::kSender,
                                               forward_transport_config_);
   return send_transport_;
 }
 
+size_t RampUpTester::GetNumVideoStreams() const {
+  return num_video_streams_;
+}
+
 void RampUpTester::ModifyVideoConfigs(
     VideoSendStream::Config* send_config,
     std::vector<VideoReceiveStream::Config>* receive_configs,
     VideoEncoderConfig* encoder_config) {
   send_config->suspend_below_min_bitrate = true;
 
   if (num_video_streams_ == 1) {
     encoder_config->streams[0].target_bitrate_bps =
         encoder_config->streams[0].max_bitrate_bps = 2000000;
     // For single stream rampup until 1mbps
@@ skipping 397 matching lines @@
                     true);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTransportSequenceNumber, false, false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc