Move the call to enable the WebRTC event log from PeerConnectionFactory to PeerConnection.

content/renderer/media/media_stream_audio_processor_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <stddef.h>
 #include <stdint.h>
 
 #include <vector>
 
 #include "base/files/file_path.h"
@@ skipping 465 matching lines @@
   audio_processor->Stop();
 }
 
 // Test that if we have an AEC dump message filter created, we are getting it
 // correctly in MSAP. Any IPC messages will be deleted since no sender in the
 // filter will be created.
 TEST_F(MediaStreamAudioProcessorTest, GetAecDumpMessageFilter) {
   base::MessageLoopForUI message_loop;
   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_(
       new AecDumpMessageFilter(message_loop.task_runner(),
-                               message_loop.task_runner(), nullptr));
+                               message_loop.task_runner()));
 
   MockConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), input_device_params_,
           webrtc_audio_device.get()));
 
   EXPECT_TRUE(audio_processor->aec_dump_message_filter_.get());
@@ skipping 90 matching lines @@
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
 
   // Stop |audio_processor| so that it removes itself from
   // |webrtc_audio_device| and clears its pointer to it.
   audio_processor->Stop();
 }
 
 }  // namespace content