Roll WebRTC 11523:11548, Libjingle 11522:11545

third_party/libjingle/BUILD.gn

 # 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.
 
 import("//build/config/features.gni")
 
 # From third_party/libjingle/libjingle.gyp's target_defaults.
 config("jingle_unexported_configs") {
   defines = [
     "EXPAT_RELATIVE_PATH",
@@ skipping 278 matching lines @@
     public_configs = [ ":jingle_public_configs" ]
     public_deps = [
       ":libjingle_webrtc_common",
     ]
   }
 
   # Note: this does not support the shared library build of libpeerconnection
   # as is supported in the GYP build. It's not clear what this is used for.
   source_set("libjingle_webrtc_common") {
     sources = [
+      "../webrtc/api/audiotrack.cc",
+      "../webrtc/api/audiotrack.h",
+      "../webrtc/api/datachannel.cc",
+      "../webrtc/api/datachannel.h",
+      "../webrtc/api/dtlsidentitystore.cc",
+      "../webrtc/api/dtlsidentitystore.h",
+      "../webrtc/api/dtmfsender.cc",
+      "../webrtc/api/dtmfsender.h",
+      "../webrtc/api/jsep.h",
+      "../webrtc/api/jsepicecandidate.cc",
+      "../webrtc/api/jsepicecandidate.h",
+      "../webrtc/api/jsepsessiondescription.cc",
+      "../webrtc/api/jsepsessiondescription.h",
+      "../webrtc/api/localaudiosource.cc",
+      "../webrtc/api/localaudiosource.h",
+      "../webrtc/api/mediaconstraintsinterface.cc",
+      "../webrtc/api/mediaconstraintsinterface.h",
+      "../webrtc/api/mediacontroller.cc",
+      "../webrtc/api/mediacontroller.h",
+      "../webrtc/api/mediastream.cc",
+      "../webrtc/api/mediastream.h",
+      "../webrtc/api/mediastreamhandler.cc",
+      "../webrtc/api/mediastreamhandler.h",
+      "../webrtc/api/mediastreaminterface.h",
+      "../webrtc/api/mediastreamobserver.cc",
+      "../webrtc/api/mediastreamobserver.h",
+      "../webrtc/api/mediastreamprovider.h",
+      "../webrtc/api/mediastreamproxy.h",
+      "../webrtc/api/mediastreamtrack.h",
+      "../webrtc/api/mediastreamtrackproxy.h",
+      "../webrtc/api/notifier.h",
+      "../webrtc/api/peerconnection.cc",
+      "../webrtc/api/peerconnection.h",
+      "../webrtc/api/peerconnectionfactory.cc",
+      "../webrtc/api/peerconnectionfactory.h",
+      "../webrtc/api/peerconnectioninterface.h",
+      "../webrtc/api/portallocatorfactory.cc",
+      "../webrtc/api/portallocatorfactory.h",
+      "../webrtc/api/remoteaudiosource.cc",
+      "../webrtc/api/remoteaudiosource.h",
+      "../webrtc/api/remoteaudiotrack.cc",
+      "../webrtc/api/remoteaudiotrack.h",
+      "../webrtc/api/remotevideocapturer.cc",
+      "../webrtc/api/remotevideocapturer.h",
+      "../webrtc/api/rtpreceiver.cc",
+      "../webrtc/api/rtpreceiver.h",
+      "../webrtc/api/rtpreceiverinterface.h",
+      "../webrtc/api/rtpsender.cc",
+      "../webrtc/api/rtpsender.h",
+      "../webrtc/api/rtpsenderinterface.h",
+      "../webrtc/api/sctputils.cc",
+      "../webrtc/api/sctputils.h",
+      "../webrtc/api/statscollector.cc",
+      "../webrtc/api/statscollector.h",
+      "../webrtc/api/statstypes.cc",
+      "../webrtc/api/statstypes.h",
+      "../webrtc/api/streamcollection.h",
+      "../webrtc/api/umametrics.h",
+      "../webrtc/api/videosource.cc",
+      "../webrtc/api/videosource.h",
+      "../webrtc/api/videosourceinterface.h",
+      "../webrtc/api/videosourceproxy.h",
+      "../webrtc/api/videotrack.cc",
+      "../webrtc/api/videotrack.h",
+      "../webrtc/api/videotrackrenderers.cc",
+      "../webrtc/api/videotrackrenderers.h",
+      "../webrtc/api/webrtcsdp.cc",
+      "../webrtc/api/webrtcsdp.h",
+      "../webrtc/api/webrtcsession.cc",
+      "../webrtc/api/webrtcsession.h",
+      "../webrtc/api/webrtcsessiondescriptionfactory.cc",
+      "../webrtc/api/webrtcsessiondescriptionfactory.h",
       "../webrtc/media/base/audiorenderer.h",
       "../webrtc/media/base/capturemanager.cc",
       "../webrtc/media/base/capturemanager.h",
       "../webrtc/media/base/capturerenderadapter.cc",
       "../webrtc/media/base/capturerenderadapter.h",
       "../webrtc/media/base/codec.cc",
       "../webrtc/media/base/codec.h",
       "../webrtc/media/base/constants.cc",
       "../webrtc/media/base/constants.h",
       "../webrtc/media/base/cryptoparams.h",
@@ skipping 22 matching lines @@
       "../webrtc/media/base/videoframefactory.cc",
       "../webrtc/media/base/videoframefactory.h",
       "../webrtc/media/devices/dummydevicemanager.cc",
       "../webrtc/media/devices/dummydevicemanager.h",
       "../webrtc/media/webrtc/webrtccommon.h",
       "../webrtc/media/webrtc/webrtcvideoframe.cc",
       "../webrtc/media/webrtc/webrtcvideoframe.h",
       "../webrtc/media/webrtc/webrtcvideoframefactory.cc",
       "../webrtc/media/webrtc/webrtcvideoframefactory.h",
       "../webrtc/media/webrtc/webrtcvoe.h",
-      "source/talk/app/webrtc/audiotrack.cc",
-      "source/talk/app/webrtc/audiotrack.h",
-      "source/talk/app/webrtc/datachannel.cc",
-      "source/talk/app/webrtc/datachannel.h",
-      "source/talk/app/webrtc/dtlsidentitystore.cc",
-      "source/talk/app/webrtc/dtlsidentitystore.h",
-      "source/talk/app/webrtc/dtmfsender.cc",
-      "source/talk/app/webrtc/dtmfsender.h",
-      "source/talk/app/webrtc/jsep.h",
-      "source/talk/app/webrtc/jsepicecandidate.cc",
-      "source/talk/app/webrtc/jsepicecandidate.h",
-      "source/talk/app/webrtc/jsepsessiondescription.cc",
-      "source/talk/app/webrtc/jsepsessiondescription.h",
-      "source/talk/app/webrtc/localaudiosource.cc",
-      "source/talk/app/webrtc/localaudiosource.h",
-      "source/talk/app/webrtc/mediaconstraintsinterface.cc",
-      "source/talk/app/webrtc/mediaconstraintsinterface.h",
-      "source/talk/app/webrtc/mediacontroller.cc",
-      "source/talk/app/webrtc/mediacontroller.h",
-      "source/talk/app/webrtc/mediastream.cc",
-      "source/talk/app/webrtc/mediastream.h",
-      "source/talk/app/webrtc/mediastreamhandler.cc",
-      "source/talk/app/webrtc/mediastreamhandler.h",
-      "source/talk/app/webrtc/mediastreaminterface.h",
-      "source/talk/app/webrtc/mediastreamobserver.cc",
-      "source/talk/app/webrtc/mediastreamobserver.h",
-      "source/talk/app/webrtc/mediastreamprovider.h",
-      "source/talk/app/webrtc/mediastreamproxy.h",
-      "source/talk/app/webrtc/mediastreamtrack.h",
-      "source/talk/app/webrtc/mediastreamtrackproxy.h",
-      "source/talk/app/webrtc/notifier.h",
-      "source/talk/app/webrtc/peerconnection.cc",
-      "source/talk/app/webrtc/peerconnection.h",
-      "source/talk/app/webrtc/peerconnectionfactory.cc",
-      "source/talk/app/webrtc/peerconnectionfactory.h",
-      "source/talk/app/webrtc/peerconnectioninterface.h",
-      "source/talk/app/webrtc/portallocatorfactory.cc",
-      "source/talk/app/webrtc/portallocatorfactory.h",
-      "source/talk/app/webrtc/remoteaudiosource.cc",
-      "source/talk/app/webrtc/remoteaudiosource.h",
-      "source/talk/app/webrtc/remoteaudiotrack.cc",
-      "source/talk/app/webrtc/remoteaudiotrack.h",
-      "source/talk/app/webrtc/remotevideocapturer.cc",
-      "source/talk/app/webrtc/remotevideocapturer.h",
-      "source/talk/app/webrtc/rtpreceiver.cc",
-      "source/talk/app/webrtc/rtpreceiver.h",
-      "source/talk/app/webrtc/rtpreceiverinterface.h",
-      "source/talk/app/webrtc/rtpsender.cc",
-      "source/talk/app/webrtc/rtpsender.h",
-      "source/talk/app/webrtc/rtpsenderinterface.h",
-      "source/talk/app/webrtc/sctputils.cc",
-      "source/talk/app/webrtc/sctputils.h",
-      "source/talk/app/webrtc/statscollector.cc",
-      "source/talk/app/webrtc/statscollector.h",
-      "source/talk/app/webrtc/statstypes.cc",
-      "source/talk/app/webrtc/statstypes.h",
-      "source/talk/app/webrtc/streamcollection.h",
-      "source/talk/app/webrtc/umametrics.h",
-      "source/talk/app/webrtc/videosource.cc",
-      "source/talk/app/webrtc/videosource.h",
-      "source/talk/app/webrtc/videosourceinterface.h",
-      "source/talk/app/webrtc/videosourceproxy.h",
-      "source/talk/app/webrtc/videotrack.cc",
-      "source/talk/app/webrtc/videotrack.h",
-      "source/talk/app/webrtc/videotrackrenderers.cc",
-      "source/talk/app/webrtc/videotrackrenderers.h",
-      "source/talk/app/webrtc/webrtcsdp.cc",
-      "source/talk/app/webrtc/webrtcsdp.h",
-      "source/talk/app/webrtc/webrtcsession.cc",
-      "source/talk/app/webrtc/webrtcsession.h",
-      "source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc",
-      "source/talk/app/webrtc/webrtcsessiondescriptionfactory.h",
       "source/talk/session/media/audiomonitor.cc",
       "source/talk/session/media/audiomonitor.h",
       "source/talk/session/media/bundlefilter.cc",
       "source/talk/session/media/bundlefilter.h",
       "source/talk/session/media/channel.cc",
       "source/talk/session/media/channel.h",
       "source/talk/session/media/channelmanager.cc",
       "source/talk/session/media/channelmanager.h",
       "source/talk/session/media/currentspeakermonitor.cc",
       "source/talk/session/media/currentspeakermonitor.h",
@@ skipping 71 matching lines @@
       "$p2p_dir/stunprober/stunprober.cc",
     ]
 
     deps = [
       ":libjingle_webrtc_common",
       "//third_party/webrtc/base:rtc_base",
     ]
   }
 }  # enable_webrtc
 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.