Switch LiveAudio to source provider solution.

content/renderer/media/rtc_peer_connection_handler.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/rtc_peer_connection_handler.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/stl_util.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_dependency_factory.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #include "content/renderer/media/remote_media_stream_impl.h"
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
 #include "content/renderer/media/rtc_media_constraints.h"
+#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_thread_impl.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 // TODO(hta): Move the following include to WebRTCStatsRequest.h file.
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
 #include "third_party/WebKit/public/platform/WebRTCPeerConnectionHandlerClient.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
@@ skipping 488 matching lines @@
 }
 
 bool RTCPeerConnectionHandler::addStream(
     const WebKit::WebMediaStream& stream,
     const WebKit::WebMediaConstraints& options) {
   RTCMediaConstraints constraints(options);
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackAddStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
+
+  // A media stream is connected to a peer connection, enable the
+  // peer connection mode for the capturer.
+  WebRtcAudioDeviceImpl* audio_device =
+      dependency_factory_->GetWebRtcAudioDevice();
+  if (audio_device) {
+    WebRtcAudioCapturer* capturer = audio_device->GetDefaultCapturer();
+    if (capturer)
+      capturer->EnablePeerConnectionMode();
+  }
+
   return AddStream(stream, &constraints);
 }
 
 void RTCPeerConnectionHandler::removeStream(
     const WebKit::WebMediaStream& stream) {
   RemoveStream(stream);
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackRemoveStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
 }
@@ skipping 226 matching lines @@
   webrtc::SessionDescriptionInterface* native_desc =
       dependency_factory_->CreateSessionDescription(type, sdp, error);
 
   LOG_IF(ERROR, !native_desc) << "Failed to create native session description."
                               << " Type: " << type << " SDP: " << sdp;
 
   return native_desc;
 }
 
 }  // namespace content