Switch LiveAudio to source provider solution.

content/renderer/media/rtc_peer_connection_handler.cc

Index: content/renderer/media/rtc_peer_connection_handler.cc
diff --git a/content/renderer/media/rtc_peer_connection_handler.cc b/content/renderer/media/rtc_peer_connection_handler.cc
index 59413582383825b38a163a688f0f7c2c11ac4ed5..af2fcf7e0b75204af564103900a368de3725c4ab 100644
--- a/content/renderer/media/rtc_peer_connection_handler.cc
+++ b/content/renderer/media/rtc_peer_connection_handler.cc
@@ -20,6 +20,8 @@
 #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.
@@ -530,6 +532,17 @@ bool RTCPeerConnectionHandler::addStream(
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackAddStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
+
+  // A media stream is connected to a peer connection, enable the native mode

[tommi (sloooow) - chröme, 2013/09/06 11:20:30]

s/the native mode/peer connection mode

[no longer working on chromium, 2013/09/10 12:43:15]

Done.

+  // 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);
 }