Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h

 // Copyright 2014 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_MOCK_PEER_CONNECTION_DEPENDENCY_FACTORY_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_MOCK_PEER_CONNECTION_DEPENDENCY_FACTORY_H_
 
 #include <set>
 #include <string>
 #include <vector>
@@ skipping 132 matching lines @@
   scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
       const webrtc::PeerConnectionInterface::RTCConfiguration& config,
       blink::WebFrame* frame,
       webrtc::PeerConnectionObserver* observer) override;
   scoped_refptr<webrtc::AudioSourceInterface> CreateLocalAudioSource(
       const cricket::AudioOptions& options) override;
   WebRtcVideoCapturerAdapter* CreateVideoCapturer(
       bool is_screen_capture) override;
   scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(
       cricket::VideoCapturer* capturer) override;
+  void CreateWebAudioSource(blink::WebMediaStreamSource* source) override;
   scoped_refptr<webrtc::MediaStreamInterface> CreateLocalMediaStream(
       const std::string& label) override;
   scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
       const std::string& id,
       webrtc::VideoTrackSourceInterface* source) override;
   scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
       const std::string& id,
       cricket::VideoCapturer* capturer) override;
   webrtc::SessionDescriptionInterface* CreateSessionDescription(
       const std::string& type,
       const std::string& sdp,
       webrtc::SdpParseError* error) override;
   webrtc::IceCandidateInterface* CreateIceCandidate(
       const std::string& sdp_mid,
       int sdp_mline_index,
       const std::string& sdp) override;
 
+  std::unique_ptr<WebRtcAudioCapturer> CreateAudioCapturer(
+      int render_frame_id,
+      const StreamDeviceInfo& device_info,
+      const blink::WebMediaConstraints& constraints,
+      MediaStreamAudioSource* audio_source) override;
+  void FailToCreateNextAudioCapturer() {
+    fail_to_create_next_audio_capturer_ = true;
+  }
+
   MockAudioSource* last_audio_source() { return last_audio_source_.get(); }
 
-  scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread()
-      const override;
-
  private:
-  scoped_refptr<MockAudioSource> last_audio_source_;
-  base::Thread signaling_thread_;
+  bool fail_to_create_next_audio_capturer_;
+  scoped_refptr <MockAudioSource> last_audio_source_;
 
   DISALLOW_COPY_AND_ASSIGN(MockPeerConnectionDependencyFactory);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_MOCK_PEER_CONNECTION_DEPENDENCY_FACTORY_H_