Index: webrtc/p2p/BUILD.gn |
diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn |
new file mode 100644 |
index 0000000000000000000000000000000000000000..538781b9ffe9baced9e2826f1d2903f85c3c09de |
--- /dev/null |
+++ b/webrtc/p2p/BUILD.gn |
@@ -0,0 +1,138 @@ |
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
+# |
+# Use of this source code is governed by a BSD-style license |
+# that can be found in the LICENSE file in the root of the source |
+# tree. An additional intellectual property rights grant can be found |
+# in the file PATENTS. All contributing project authors may |
+# be found in the AUTHORS file in the root of the source tree. |
+ |
+import("../build/webrtc.gni") |
+ |
+group("p2p") { |
+ deps = [ |
+ ":rtc_p2p", |
+ ] |
+} |
+ |
+config("rtc_p2p_inherited_config") { |
+ defines = [ "FEATURE_ENABLE_VOICEMAIL" ] |
+} |
+ |
+source_set("rtc_p2p") { |
+ sources = [ |
+ "base/asyncstuntcpsocket.cc", |
+ "base/asyncstuntcpsocket.h", |
+ "base/basicpacketsocketfactory.cc", |
+ "base/basicpacketsocketfactory.h", |
+ "base/candidate.h", |
+ "base/common.h", |
+ "base/dtlstransportchannel.cc", |
+ "base/dtlstransportchannel.h", |
+ "base/p2pconstants.cc", |
+ "base/p2pconstants.h", |
+ "base/p2ptransport.cc", |
+ "base/p2ptransport.h", |
+ "base/p2ptransportchannel.cc", |
+ "base/p2ptransportchannel.h", |
+ "base/packetsocketfactory.h", |
+ "base/port.cc", |
+ "base/port.h", |
+ "base/portallocator.cc", |
+ "base/portallocator.h", |
+ "base/portinterface.h", |
+ "base/pseudotcp.cc", |
+ "base/pseudotcp.h", |
+ "base/relayport.cc", |
+ "base/relayport.h", |
+ "base/relayserver.cc", |
+ "base/relayserver.h", |
+ "base/sessiondescription.cc", |
+ "base/sessiondescription.h", |
+ "base/sessionid.h", |
+ "base/stun.cc", |
+ "base/stun.h", |
+ "base/stunport.cc", |
+ "base/stunport.h", |
+ "base/stunrequest.cc", |
+ "base/stunrequest.h", |
+ "base/stunserver.cc", |
+ "base/stunserver.h", |
+ "base/tcpport.cc", |
+ "base/tcpport.h", |
+ "base/transport.cc", |
+ "base/transport.h", |
+ "base/transportchannel.cc", |
+ "base/transportchannel.h", |
+ "base/transportchannelimpl.h", |
+ "base/transportcontroller.cc", |
+ "base/transportcontroller.h", |
+ "base/transportdescription.cc", |
+ "base/transportdescription.h", |
+ "base/transportdescriptionfactory.cc", |
+ "base/transportdescriptionfactory.h", |
+ "base/transportinfo.h", |
+ "base/turnport.cc", |
+ "base/turnport.h", |
+ "base/turnserver.cc", |
+ "base/turnserver.h", |
+ "base/udpport.h", |
+ "client/basicportallocator.cc", |
+ "client/basicportallocator.h", |
+ "client/httpportallocator.cc", |
+ "client/httpportallocator.h", |
+ "client/socketmonitor.cc", |
+ "client/socketmonitor.h", |
+ ] |
+ |
+ defines = [ "FEATURE_ENABLE_SSL" ] |
+ |
+ deps = [ |
+ "../base:rtc_base", |
+ "../libjingle/xmllite", |
+ ] |
+ |
+ if (rtc_build_expat) { |
+ deps += [ "//third_party/expat" ] |
+ public_deps = [ |
+ "//third_party/expat", |
+ ] |
+ } |
+ |
+ configs += [ "..:common_config" ] |
+ public_configs = [ |
+ "..:common_inherited_config", |
+ ":rtc_p2p_inherited_config", |
+ ] |
+ |
+ if (!build_with_chromium) { |
+ defines += [ |
+ "FEATURE_ENABLE_VOICEMAIL", |
+ "FEATURE_ENABLE_PSTN", |
+ ] |
+ } |
+ |
+ if (rtc_use_quic) { |
+ deps = [ |
+ "//third_party/libquic", |
+ ] |
+ sources += [ |
+ "quic/quicconnectionhelper.cc", |
+ "quic/quicconnectionhelper.h", |
+ "quic/quicsession.cc", |
+ "quic/quicsession.h", |
+ "quic/quictransport.cc", |
+ "quic/quictransport.h", |
+ "quic/quictransportchannel.cc", |
+ "quic/quictransportchannel.h", |
+ "quic/reliablequicstream.cc", |
+ "quic/reliablequicstream.h", |
+ ] |
+ public_deps += [ "//third_party/libquic" ] |
+ } |
+ |
+ if (is_clang) { |
+ # Suppress warnings from Chrome's Clang plugins. |
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. |
+ configs -= [ "//build/config/clang:find_bad_constructs" ] |
+ } |
+} |