Remove PC factory options param from LocalAudioSource::Create.

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 286 matching lines @@
 // associated and disassociated with a VideoRtpReceiver.
 TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) {
   CreateVideoRtpReceiver();
   DestroyVideoRtpReceiver();
 }
 
 // Test that the AudioRtpSender applies options from the local audio source.
 TEST_F(RtpSenderReceiverTest, LocalAudioSourceOptionsApplied) {
   cricket::AudioOptions options;
   options.echo_cancellation = rtc::Optional<bool>(true);
-  auto source = LocalAudioSource::Create(
-      PeerConnectionFactoryInterface::Options(), &options);
+  auto source = LocalAudioSource::Create(&options);
   CreateAudioRtpSender(source.get());
 
   EXPECT_EQ(rtc::Optional<bool>(true),
             voice_media_channel_->options().echo_cancellation);
 
   DestroyAudioRtpSender();
 }
 
 // Test that the stream is muted when the track is disabled, and unmuted when
 // the track is enabled.
@@ skipping 481 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