Add a virtual beamforming audio device on ChromeOS.

content/renderer/media/webrtc_local_audio_renderer.cc

Index: content/renderer/media/webrtc_local_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
index 55282c8d62f7fa5c605637a0664fb3dc4d08e694..71488c3124e6be4d526ec3c6ed045538c1d35484 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -263,12 +263,12 @@ void WebRtcLocalAudioRenderer::ReconfigureSink(
 
   source_params_ = params;
 
-  sink_params_ = media::AudioParameters(source_params_.format(),
-      source_params_.channel_layout(), source_params_.sample_rate(),
-      source_params_.bits_per_sample(),
+  sink_params_ = media::AudioParameters(
+      source_params_.format(), source_params_.channel_layout(),
+      source_params_.sample_rate(), source_params_.bits_per_sample(),
       WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(),
                                                 frames_per_buffer_),
-      source_params_.effects());
+      std::vector<media::Point>(), source_params_.effects());
 
   {
     // Note: The max buffer is fairly large, but will rarely be used.