Moving SwitchOutputDevice out of OutputDevice interface, eliminating OutputDevice

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/macros.h"
 #include "base/metrics/histogram.h"
@@ skipping 157 matching lines @@
   if (media::ToAudioSampleRate(device_info_.device.input.sample_rate, &asr)) {
     UMA_HISTOGRAM_ENUMERATION(
         "WebRTC.AudioInputSampleRate", asr, media::kAudioSampleRateMax + 1);
   } else {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputSampleRateUnexpected",
                          device_info_.device.input.sample_rate);
   }
 
   // Create and configure the default audio capturing source.
   SetCapturerSourceInternal(
-      AudioDeviceFactory::NewInputDevice(render_frame_id_), channel_layout,
-      device_info_.device.input.sample_rate);
+      AudioDeviceFactory::NewAudioCapturerSource(render_frame_id_),
+      channel_layout, device_info_.device.input.sample_rate);
 
   // Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware
   // information from the capturer.
   if (audio_device_)
     audio_device_->AddAudioCapturer(this);
 
   return true;
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer(
@@ skipping 154 matching lines @@
   }
 
   // Do nothing if the current buffer size is the WebRtc native buffer size.
   if (GetBufferSize(input_params.sample_rate()) ==
           input_params.frames_per_buffer()) {
     return;
   }
 
   // Create a new audio stream as source which will open the hardware using
   // WebRtc native buffer size.
-  SetCapturerSourceInternal(AudioDeviceFactory::NewInputDevice(render_frame_id),
-                            input_params.channel_layout(),
-                            input_params.sample_rate());
+  SetCapturerSourceInternal(
+      AudioDeviceFactory::NewAudioCapturerSource(render_frame_id),
+      input_params.channel_layout(), input_params.sample_rate());
 }
 
 void WebRtcAudioCapturer::Start() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Start()";
   base::AutoLock auto_lock(lock_);
   if (running_ || !source_.get())
     return;
 
   // Start the data source, i.e., start capturing data from the current source.
@@ skipping 190 matching lines @@
 
 void WebRtcAudioCapturer::SetCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
   SetCapturerSourceInternal(source, params.channel_layout(),
                             params.sample_rate());
 }
 
 }  // namespace content