Webrtc deps roll.

content/renderer/media/webrtc_audio_capturer.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_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 210 matching lines @@
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer(
     int render_view_id,
     const StreamDeviceInfo& device_info,
     const blink::WebMediaConstraints& constraints,
     WebRtcAudioDeviceImpl* audio_device,
     MediaStreamAudioSource* audio_source)
     : constraints_(constraints),
       audio_processor_(
-          new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+          new rtc::RefCountedObject<MediaStreamAudioProcessor>(
               constraints, device_info.device.input.effects, audio_device)),
       running_(false),
       render_view_id_(render_view_id),
       device_info_(device_info),
       volume_(0),
       peer_connection_mode_(false),
       key_pressed_(false),
       need_audio_processing_(false),
       audio_device_(audio_device),
       audio_source_(audio_source),
@@ skipping 367 matching lines @@
 
 void WebRtcAudioCapturer::SetCapturerSourceForTesting(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSource(source, params.channel_layout(),
                     static_cast<float>(params.sample_rate()));
 }
 
 }  // namespace content