Removing unnecessary parameters from CreateXChannel methods.

webrtc/pc/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 2012 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 52 matching lines @@
         media_engine_(new cricket::FakeMediaEngine()),
         channel_manager_(
             std::unique_ptr<cricket::MediaEngineInterface>(media_engine_),
             rtc::Thread::Current(),
             rtc::Thread::Current()),
         fake_call_(Call::Config(&event_log_)),
         fake_media_controller_(&channel_manager_, &fake_call_),
         stream_(MediaStream::Create(kStreamLabel1)) {
     // Create channels to be used by the RtpSenders and RtpReceivers.
     channel_manager_.Init();
-    bool rtcp_mux_required = true;
     bool srtp_required = true;
     cricket::DtlsTransportInternal* rtp_transport =
         fake_transport_controller_.CreateDtlsTransport(
             cricket::CN_AUDIO, cricket::ICE_CANDIDATE_COMPONENT_RTP);
     voice_channel_ = channel_manager_.CreateVoiceChannel(
         &fake_media_controller_, rtp_transport, nullptr, rtc::Thread::Current(),
-        cricket::CN_AUDIO, nullptr, rtcp_mux_required, srtp_required,
-        cricket::AudioOptions());
+        cricket::CN_AUDIO, srtp_required, cricket::AudioOptions());
     video_channel_ = channel_manager_.CreateVideoChannel(
         &fake_media_controller_, rtp_transport, nullptr, rtc::Thread::Current(),
-        cricket::CN_VIDEO, nullptr, rtcp_mux_required, srtp_required,
-        cricket::VideoOptions());
+        cricket::CN_VIDEO, srtp_required, cricket::VideoOptions());
     voice_channel_->Enable(true);
     video_channel_->Enable(true);
     voice_media_channel_ = media_engine_->GetVoiceChannel(0);
     video_media_channel_ = media_engine_->GetVideoChannel(0);
     RTC_CHECK(voice_channel_);
     RTC_CHECK(video_channel_);
     RTC_CHECK(voice_media_channel_);
     RTC_CHECK(video_media_channel_);
 
     // Create streams for predefined SSRCs. Streams need to exist in order
@@ skipping 703 matching lines @@
 // Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
 // destroyed, which is needed for the DTMF sender.
 TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
   CreateAudioRtpSender();
   EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
   audio_rtp_sender_ = nullptr;
   EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
 }
 
 }  // namespace webrtc