Add a virtual beamforming audio device on ChromeOS.

content/renderer/media/webrtc_audio_renderer.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 "content/renderer/media/webrtc_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 181 matching lines @@
       source_render_frame_id_(source_render_frame_id),
       session_id_(session_id),
       signaling_thread_(signaling_thread),
       media_stream_(media_stream),
       source_(NULL),
       play_ref_count_(0),
       start_ref_count_(0),
       audio_delay_milliseconds_(0),
       fifo_delay_milliseconds_(0),
       sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                   media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
+                   media::CHANNEL_LAYOUT_STEREO,
+                   sample_rate,
+                   16,
                    frames_per_buffer,
+                   std::vector<media::Point>(),
                    GetCurrentDuckingFlag(source_render_frame_id)),
       render_callback_count_(0) {
   WebRtcLogMessage(base::StringPrintf(
       "WAR::WAR. source_render_frame_id=%d"
       ", session_id=%d, sample_rate=%d, frames_per_buffer=%d, effects=%i",
       source_render_frame_id, session_id, sample_rate, frames_per_buffer,
       sink_params_.effects()));
 }
 
 WebRtcAudioRenderer::~WebRtcAudioRenderer() {
@@ skipping 355 matching lines @@
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
 
 }  // namespace content