Files
platform-external-webrtc/webrtc/p2p/BUILD.gn
zhihuang 86abd6f2fd Add an abstract class for IceTransport
P2PTransportChannl will be renamed to IceTransport and this class will be
the base class of IceTransport. By doing this, the Dtls related methods
can be separated from the IceTransport.

For more detail, https://docs.google.com/document/d/1g9RA0s4RV7hFAcWiAM2b6H5ZohAVpBqcYienEDj6IcY/edit

BUG=none

Review-Url: https://codereview.webrtc.org/2577183004
Cr-Commit-Position: refs/heads/master@{#15690}
2016-12-19 19:54:05 +00:00

159 lines
4.0 KiB
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# 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") {
public_deps = [
":libstunprober",
":rtc_p2p",
]
}
config("rtc_p2p_inherited_config") {
defines = [ "FEATURE_ENABLE_VOICEMAIL" ]
}
rtc_static_library("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/icetransportinternal.h",
"base/jseptransport.cc",
"base/jseptransport.h",
"base/p2pconstants.cc",
"base/p2pconstants.h",
"base/p2ptransportchannel.cc",
"base/p2ptransportchannel.h",
"base/packetsocketfactory.h",
"base/packettransportinterface.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/session.cc",
"base/session.h",
"base/sessiondescription.cc",
"base/sessiondescription.h",
"base/stun.cc",
"base/stun.h",
"base/stunport.cc",
"base/stunport.h",
"base/stunrequest.cc",
"base/stunrequest.h",
"base/tcpport.cc",
"base/tcpport.h",
"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/udpport.h",
"base/udptransportchannel.cc",
"base/udptransportchannel.h",
"client/basicportallocator.cc",
"client/basicportallocator.h",
"client/socketmonitor.cc",
"client/socketmonitor.h",
]
defines = [ "FEATURE_ENABLE_SSL" ]
deps = [
"../base:rtc_base",
]
if (rtc_build_expat) {
deps += [ "//third_party/expat" ]
public_deps = [
"//third_party/expat",
]
}
public_configs = [ ":rtc_p2p_inherited_config" ]
if (build_with_chromium) {
if (is_nacl) {
deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
}
} else {
sources += [
"base/relayserver.cc",
"base/relayserver.h",
"base/stunserver.cc",
"base/stunserver.h",
"base/turnserver.cc",
"base/turnserver.h",
]
defines += [
"FEATURE_ENABLE_VOICEMAIL",
"FEATURE_ENABLE_PSTN",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
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" ]
}
}
rtc_static_library("libstunprober") {
sources = [
"stunprober/stunprober.cc",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
"..:webrtc_common",
"../base:rtc_base",
]
}