webrtc_m130/p2p/BUILD.gn
Guido Urdaneta 604427b875 Revert "TurnCustomizer - an interface for modifying stun messages sent by TurnPort"
This reverts commit b23ed7f1af467a228cbdc63e839cac8856e9df8d.

Reason for revert: Breaks Chromium FYI build

Sample error log:

../../remoting/test/fake_port_allocator.cc:52:7: error: no matching constructor for initialization of 'cricket::BasicPortAllocator'
    : BasicPortAllocator(network_manager, socket_factory),
      ^                  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../third_party/webrtc/p2p/client/basicportallocator.h:32:12: note: candidate constructor not viable: requires single argument 'network_manager', but 2 arguments were provided
  explicit BasicPortAllocator(rtc::NetworkManager* network_manager);
           ^
../../third_party/webrtc/p2p/client/basicportallocator.h:27:7: note: candidate constructor (the implicit copy constructor) not viable: requires 1 argument, but 2 were provided
class BasicPortAllocator : public PortAllocator {
      ^
../../third_party/webrtc/p2p/client/basicportallocator.h:29:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
  BasicPortAllocator(rtc::NetworkManager* network_manager,
  ^
../../third_party/webrtc/p2p/client/basicportallocator.h:33:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
  BasicPortAllocator(rtc::NetworkManager* network_manager,
  ^
../../third_party/webrtc/p2p/client/basicportallocator.h:36:3: note: candidate constructor not viable: requires 5 arguments, but 2 were provided
  BasicPortAllocator(rtc::NetworkManager* network_manager,

Original change's description:
> TurnCustomizer - an interface for modifying stun messages sent by TurnPort
> 
> This patch adds an interface that allows modification of stun messages
> sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
> and the TurnCustomizer will be invoked by TurnPort before sending
> message. This allows user to e.g add custom attributes as described
> in rtf5389.
> 
> BUG=webrtc:8313
> 
> Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
> Reviewed-on: https://webrtc-review.googlesource.com/4781
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20197}

TBR=deadbeef@webrtc.org,sakal@webrtc.org,jonaso@webrtc.org

Change-Id: I624efb22f6e3ceac1b2ff8af1ec47e4cfdde9140
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8313
Reviewed-on: https://webrtc-review.googlesource.com/7680
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20199}
2017-10-09 09:53:59 +00:00

271 lines
7.5 KiB
Plaintext

# 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("../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/dtlstransport.cc",
"base/dtlstransport.h",
"base/dtlstransportinternal.h",
"base/icetransportinternal.h",
"base/jseptransport.cc",
"base/jseptransport.h",
"base/p2pconstants.cc",
"base/p2pconstants.h",
"base/p2ptransportchannel.cc",
"base/p2ptransportchannel.h",
"base/packetlossestimator.cc",
"base/packetlossestimator.h",
"base/packetsocketfactory.h",
"base/packettransportinterface.h",
"base/packettransportinternal.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/transportdescription.cc",
"base/transportdescription.h",
"base/transportdescriptionfactory.cc",
"base/transportdescriptionfactory.h",
"base/transportinfo.h",
"base/turnport.cc",
"base/turnport.h",
"base/udpport.h",
"base/udptransport.cc",
"base/udptransport.h",
"client/basicportallocator.cc",
"client/basicportallocator.h",
"client/socketmonitor.cc",
"client/socketmonitor.h",
]
defines = []
deps = [
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:ortc_api",
"../rtc_base:rtc_base",
"../system_wrappers:field_trial_api",
]
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" ]
}
}
if (rtc_include_tests) {
rtc_source_set("p2p_test_utils") {
testonly = true
sources = [
"base/fakecandidatepair.h",
"base/fakedtlstransport.h",
"base/fakeicetransport.h",
"base/fakepackettransport.h",
"base/fakeportallocator.h",
"base/mockicetransport.h",
"base/testrelayserver.h",
"base/teststunserver.h",
"base/testturnserver.h",
]
deps = [
":rtc_p2p",
"../api:ortc_api",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../test:test_support",
"//testing/gmock",
]
}
rtc_source_set("rtc_p2p_unittests") {
testonly = true
# Skip restricting visibility on mobile platforms since the tests on those
# gets additional generated targets which would require many lines here to
# cover (which would be confusing to read and hard to maintain).
if (!is_android && !is_ios) {
visibility = [ "..:rtc_unittests" ]
}
sources = [
"base/asyncstuntcpsocket_unittest.cc",
"base/dtlstransport_unittest.cc",
"base/jseptransport_unittest.cc",
"base/p2ptransportchannel_unittest.cc",
"base/packetlossestimator_unittest.cc",
"base/port_unittest.cc",
"base/portallocator_unittest.cc",
"base/pseudotcp_unittest.cc",
"base/relayport_unittest.cc",
"base/relayserver_unittest.cc",
"base/stun_unittest.cc",
"base/stunport_unittest.cc",
"base/stunrequest_unittest.cc",
"base/stunserver_unittest.cc",
"base/tcpport_unittest.cc",
"base/transportdescriptionfactory_unittest.cc",
"base/turnport_unittest.cc",
"base/turnserver_unittest.cc",
"base/udptransport_unittest.cc",
"client/basicportallocator_unittest.cc",
]
if (rtc_use_quic) {
sources += [
"quic/quicconnectionhelper_unittest.cc",
"quic/quicsession_unittest.cc",
"quic/quictransport_unittest.cc",
"quic/quictransportchannel_unittest.cc",
"quic/reliablequicstream_unittest.cc",
]
}
deps = [
":p2p_test_utils",
":rtc_p2p",
"../api:fakemetricsobserver",
"../api:ortc_api",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../test:test_support",
"//testing/gmock",
"//testing/gtest",
]
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" ]
}
defines = [ "GTEST_RELATIVE_PATH" ]
}
}
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 = [
":rtc_p2p",
"..:webrtc_common",
"../rtc_base:rtc_base",
]
}
if (rtc_include_tests) {
rtc_source_set("libstunprober_unittests") {
testonly = true
# Skip restricting visibility on mobile platforms since the tests on those
# gets additional generated targets which would require many lines here to
# cover (which would be confusing to read and hard to maintain).
if (!is_android && !is_ios) {
visibility = [ "..:rtc_unittests" ]
}
sources = [
"stunprober/stunprober_unittest.cc",
]
deps = [
":libstunprober",
":p2p_test_utils",
":rtc_p2p",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_tests_utils",
"//testing/gmock",
"//testing/gtest",
]
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" ]
}
defines = [ "GTEST_RELATIVE_PATH" ]
}
}