Feed audio constraints over to WebRtcLocalAudioTrack

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/webrtc_audio_capturer.h"
@@ skipping 390 matching lines @@
 MockMediaStreamDependencyFactory::CreateLocalVideoSource(
     int video_session_id,
     bool is_screencast,
     const webrtc::MediaConstraintsInterface* constraints) {
   last_video_source_ = new talk_base::RefCountedObject<MockVideoSource>();
   return last_video_source_;
 }
 
 scoped_refptr<WebRtcAudioCapturer>
 MockMediaStreamDependencyFactory::CreateWebAudioSource(
-    WebKit::WebMediaStreamSource* source) {
+    WebKit::WebMediaStreamSource* source,
+    RTCMediaConstraints* constraints) {
   return NULL;
 }
 
 scoped_refptr<webrtc::MediaStreamInterface>
 MockMediaStreamDependencyFactory::CreateLocalMediaStream(
     const std::string& label) {
   DCHECK(mock_pc_factory_created_);
   return new talk_base::RefCountedObject<MockMediaStream>(label);
 }
 
@@ skipping 18 matching lines @@
       new talk_base::RefCountedObject<MockVideoSource>();
   source->SetVideoCapturer(capturer);
 
   return new talk_base::RefCountedObject<MockLocalVideoTrack>(id, source.get());
 }
 
 scoped_refptr<webrtc::AudioTrackInterface>
 MockMediaStreamDependencyFactory::CreateLocalAudioTrack(
     const std::string& id,
     const scoped_refptr<WebRtcAudioCapturer>& capturer,
-    webrtc::AudioSourceInterface* source) {
+    webrtc::AudioSourceInterface* source,
+    const webrtc::MediaConstraintsInterface* constraints) {
   DCHECK(mock_pc_factory_created_);
   DCHECK(!capturer.get());
   return WebRtcLocalAudioTrack::Create(
-      id, WebRtcAudioCapturer::CreateCapturer(), source);
+      id, WebRtcAudioCapturer::CreateCapturer(), 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(
     int render_view_id, const StreamDeviceInfo& device_info) {
   return WebRtcAudioCapturer::CreateCapturer();
 }
 
 }  // namespace content