Cleaned up the WebRtcAudioCapturer a bit.

content/renderer/media/mock_media_stream_dependency_factory.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/mock_media_stream_dependency_factory.h"
 
 #include "base/logging.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/renderer/media/mock_peer_connection_impl.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
@@ skipping 437 matching lines @@
 }
 
 scoped_refptr<webrtc::AudioTrackInterface>
 MockMediaStreamDependencyFactory::CreateLocalAudioTrack(
     const std::string& id,
     const scoped_refptr<WebRtcAudioCapturer>& capturer,
     WebAudioCapturerSource* webaudio_source,
     webrtc::AudioSourceInterface* source,
     const webrtc::MediaConstraintsInterface* constraints) {
   DCHECK(mock_pc_factory_created_);
-  DCHECK(!capturer.get());
+  scoped_refptr<WebRtcAudioCapturer> audio_capturer = capturer ?

[perkj_chrome, 2014/01/12 21:53:06]

This  seem strange. Why don't you know if you must create a capturer or not? 

[no longer working on chromium, 2014/01/13 16:59:59]

Previously the capturer is got from WebRtcAudioDeviceImpl, and we know for sure
that in unittests WebRtcAudioDeviceImpl does not exist.
But this CL changes the code that the capturer will be got from the existing
source. So the capturer might exist or might not, depending on how the unittests
are written.

+      capturer : WebRtcAudioCapturer::CreateCapturer(-1, StreamDeviceInfo());
   return WebRtcLocalAudioTrack::Create(
-      id, WebRtcAudioCapturer::CreateCapturer(), webaudio_source,
-      source, constraints);
+      id, audio_capturer, webaudio_source, source, constraints);
 }
 
 SessionDescriptionInterface*
 MockMediaStreamDependencyFactory::CreateSessionDescription(
     const std::string& type,
     const std::string& sdp,
     webrtc::SdpParseError* error) {
   return new MockSessionDescription(type, sdp);
 }
 
 webrtc::IceCandidateInterface*
 MockMediaStreamDependencyFactory::CreateIceCandidate(
     const std::string& sdp_mid,
     int sdp_mline_index,
     const std::string& sdp) {
   return new MockIceCandidate(sdp_mid, sdp_mline_index, sdp);
 }
 
 scoped_refptr<WebRtcAudioCapturer>
-MockMediaStreamDependencyFactory::MaybeCreateAudioCapturer(
+MockMediaStreamDependencyFactory::CreateAudioCapturer(
     int render_view_id, const StreamDeviceInfo& device_info) {
-  return WebRtcAudioCapturer::CreateCapturer();
+  return WebRtcAudioCapturer::CreateCapturer(-1, device_info);
 }
 
 }  // namespace content