Removed unused files from libjingle.gyp

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 290 matching lines @@
       "../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/videocapturertracksource.cc",
       "../webrtc/api/videocapturertracksource.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/videotracksource.cc",
       "../webrtc/api/videotracksource.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/codec.cc",
@@ skipping 117 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.