Roll WebRTC 11486:11495, Libjingle 11485:11495

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/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",
+      "../webrtc/media/base/hybriddataengine.h",
+      "../webrtc/media/base/mediachannel.h",
+      "../webrtc/media/base/mediaengine.cc",
+      "../webrtc/media/base/mediaengine.h",
+      "../webrtc/media/base/rtpdataengine.cc",
+      "../webrtc/media/base/rtpdataengine.h",
+      "../webrtc/media/base/rtpdump.cc",
+      "../webrtc/media/base/rtpdump.h",
+      "../webrtc/media/base/rtputils.cc",
+      "../webrtc/media/base/rtputils.h",
+      "../webrtc/media/base/streamparams.cc",
+      "../webrtc/media/base/streamparams.h",
+      "../webrtc/media/base/turnutils.cc",
+      "../webrtc/media/base/turnutils.h",
+      "../webrtc/media/base/videoadapter.cc",
+      "../webrtc/media/base/videoadapter.h",
+      "../webrtc/media/base/videocapturer.cc",
+      "../webrtc/media/base/videocapturer.h",
+      "../webrtc/media/base/videocommon.cc",
+      "../webrtc/media/base/videocommon.h",
+      "../webrtc/media/base/videoframe.cc",
+      "../webrtc/media/base/videoframe.h",
+      "../webrtc/media/base/videoframefactory.cc",
+      "../webrtc/media/base/videoframefactory.h",
+      "../webrtc/media/devices/dummydevicemanager.cc",
+      "../webrtc/media/devices/dummydevicemanager.h",
+      "../webrtc/media/devices/filevideocapturer.cc",
+      "../webrtc/media/devices/filevideocapturer.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",
@@ skipping 52 matching lines @@
       "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/media/base/audiorenderer.h",
-      "source/talk/media/base/capturemanager.cc",
-      "source/talk/media/base/capturemanager.h",
-      "source/talk/media/base/capturerenderadapter.cc",
-      "source/talk/media/base/capturerenderadapter.h",
-      "source/talk/media/base/codec.cc",
-      "source/talk/media/base/codec.h",
-      "source/talk/media/base/constants.cc",
-      "source/talk/media/base/constants.h",
-      "source/talk/media/base/cryptoparams.h",
-      "source/talk/media/base/hybriddataengine.h",
-      "source/talk/media/base/mediachannel.h",
-      "source/talk/media/base/mediaengine.cc",
-      "source/talk/media/base/mediaengine.h",
-      "source/talk/media/base/rtpdataengine.cc",
-      "source/talk/media/base/rtpdataengine.h",
-      "source/talk/media/base/rtpdump.cc",
-      "source/talk/media/base/rtpdump.h",
-      "source/talk/media/base/rtputils.cc",
-      "source/talk/media/base/rtputils.h",
-      "source/talk/media/base/streamparams.cc",
-      "source/talk/media/base/streamparams.h",
-      "source/talk/media/base/turnutils.cc",
-      "source/talk/media/base/turnutils.h",
-      "source/talk/media/base/videoadapter.cc",
-      "source/talk/media/base/videoadapter.h",
-      "source/talk/media/base/videocapturer.cc",
-      "source/talk/media/base/videocapturer.h",
-      "source/talk/media/base/videocommon.cc",
-      "source/talk/media/base/videocommon.h",
-      "source/talk/media/base/videoframe.cc",
-      "source/talk/media/base/videoframe.h",
-      "source/talk/media/base/videoframefactory.cc",
-      "source/talk/media/base/videoframefactory.h",
-      "source/talk/media/devices/dummydevicemanager.cc",
-      "source/talk/media/devices/dummydevicemanager.h",
-      "source/talk/media/devices/filevideocapturer.cc",
-      "source/talk/media/devices/filevideocapturer.h",
-      "source/talk/media/webrtc/webrtccommon.h",
-      "source/talk/media/webrtc/webrtcvideoframe.cc",
-      "source/talk/media/webrtc/webrtcvideoframe.h",
-      "source/talk/media/webrtc/webrtcvideoframefactory.cc",
-      "source/talk/media/webrtc/webrtcvideoframefactory.h",
-      "source/talk/media/webrtc/webrtcvoe.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 21 matching lines @@
       ":libjingle",
       "//third_party/libsrtp",
       "//third_party/webrtc/modules/media_file",
       "//third_party/webrtc/modules/video_capture",
       "//third_party/webrtc/modules/video_render",
     ]
 
     if (!is_ios) {
       # TODO(mallinath) - Enable SCTP for iOS.
       sources += [
-        "source/talk/media/sctp/sctpdataengine.cc",
-        "source/talk/media/sctp/sctpdataengine.h",
+        "../webrtc/media/sctp/sctpdataengine.cc",
+        "../webrtc/media/sctp/sctpdataengine.h",
       ]
       defines = [ "HAVE_SCTP" ]
       deps += [ "//third_party/usrsctp" ]
     }
   }
 
   # 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("libpeerconnection") {
     sources = [
-      "source/talk/media/webrtc/simulcast.cc",
-      "source/talk/media/webrtc/simulcast.h",
-      "source/talk/media/webrtc/webrtcmediaengine.cc",
-      "source/talk/media/webrtc/webrtcmediaengine.h",
-      "source/talk/media/webrtc/webrtcvideoengine2.cc",
-      "source/talk/media/webrtc/webrtcvideoengine2.h",
-      "source/talk/media/webrtc/webrtcvoiceengine.cc",
-      "source/talk/media/webrtc/webrtcvoiceengine.h",
+      "../webrtc/media/webrtc/simulcast.cc",
+      "../webrtc/media/webrtc/simulcast.h",
+      "../webrtc/media/webrtc/webrtcmediaengine.cc",
+      "../webrtc/media/webrtc/webrtcmediaengine.h",
+      "../webrtc/media/webrtc/webrtcvideoengine2.cc",
+      "../webrtc/media/webrtc/webrtcvideoengine2.h",
+      "../webrtc/media/webrtc/webrtcvoiceengine.cc",
+      "../webrtc/media/webrtc/webrtcvoiceengine.h",
     ]
 
     configs += [ ":jingle_unexported_configs" ]
     public_configs = [ ":jingle_public_configs" ]
     configs -= [ "//build/config/compiler:chromium_code" ]
     configs += [ "//build/config/compiler:no_chromium_code" ]
 
     deps = [
       # TODO(GYP): crbug.com/481633. Consider depending on :libjingle_webrtc
       # instead.
@@ skipping 10 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.