Enable platform echo cancellation through the AudioRecord path.

content/renderer/media/webrtc_audio_device_unittest.cc

 // Copyright (c) 2012 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 <vector>
 
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/path_service.h"
@@ skipping 101 matching lines @@
       WebRtcAudioCapturer::CreateCapturer());
 
   media::AudioHardwareConfig* hardware_config =
       RenderThreadImpl::current()->GetAudioHardwareConfig();
 
   // Use native capture sample rate and channel configuration to get some
   // action in this test.
   int sample_rate = hardware_config->GetInputSampleRate();
   media::ChannelLayout channel_layout =
       hardware_config->GetInputChannelLayout();
+  media::AudioParameters::PlatformEffects effects;
   if (!capturer->Initialize(kRenderViewId, channel_layout, sample_rate, 0, 1,
-                            media::AudioManagerBase::kDefaultDeviceId, 0 ,0)) {
+                            media::AudioManagerBase::kDefaultDeviceId, 0, 0,
+                            effects)) {
     return false;
   }
 
   // Add the capturer to the WebRtcAudioDeviceImpl.
   webrtc_audio_device->AddAudioCapturer(capturer);
 
   return true;
 }
 
 // Create and start a local audio track. Starting the audio track will connect
@@ skipping 835 matching lines @@
     LOG(WARNING) << "Test disabled due to the test hangs on WinXP.";
     return;
   }
 #endif
   int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);
   PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",
                     "t", latency);
 }
 
 }  // namespace content