This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
233 lines
6.0 KiB
Plaintext
233 lines
6.0 KiB
Plaintext
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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group("p2p") {
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public_deps = [
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":libstunprober",
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":rtc_p2p",
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]
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}
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config("rtc_p2p_inherited_config") {
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defines = [ "FEATURE_ENABLE_VOICEMAIL" ]
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}
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rtc_static_library("rtc_p2p") {
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visibility = [ "*" ]
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sources = [
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"base/asyncstuntcpsocket.cc",
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"base/asyncstuntcpsocket.h",
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"base/basicpacketsocketfactory.cc",
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"base/basicpacketsocketfactory.h",
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"base/candidatepairinterface.h",
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"base/common.h",
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"base/dtlstransport.cc",
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"base/dtlstransport.h",
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"base/dtlstransportinternal.cc",
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"base/dtlstransportinternal.h",
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"base/icetransportinternal.cc",
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"base/icetransportinternal.h",
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"base/p2pconstants.cc",
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"base/p2pconstants.h",
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"base/p2ptransportchannel.cc",
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"base/p2ptransportchannel.h",
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"base/packetlossestimator.cc",
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"base/packetlossestimator.h",
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"base/packetsocketfactory.cc",
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"base/packetsocketfactory.h",
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"base/packettransportinterface.h",
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"base/packettransportinternal.cc",
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"base/packettransportinternal.h",
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"base/port.cc",
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"base/port.h",
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"base/portallocator.cc",
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"base/portallocator.h",
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"base/portinterface.cc",
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"base/portinterface.h",
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"base/pseudotcp.cc",
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"base/pseudotcp.h",
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"base/relayport.cc",
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"base/relayport.h",
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"base/session.cc",
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"base/session.h",
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"base/stun.cc",
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"base/stun.h",
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"base/stunport.cc",
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"base/stunport.h",
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"base/stunrequest.cc",
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"base/stunrequest.h",
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"base/tcpport.cc",
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"base/tcpport.h",
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"base/transport.h",
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"base/transportchannelimpl.h",
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"base/transportdescription.cc",
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"base/transportdescription.h",
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"base/transportdescriptionfactory.cc",
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"base/transportdescriptionfactory.h",
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"base/transportinfo.h",
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"base/turnport.cc",
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"base/turnport.h",
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"base/udpport.h",
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"base/udptransport.cc",
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"base/udptransport.h",
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"client/basicportallocator.cc",
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"client/basicportallocator.h",
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"client/relayportfactoryinterface.h",
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"client/socketmonitor.cc",
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"client/socketmonitor.h",
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"client/turnportfactory.cc",
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"client/turnportfactory.h",
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]
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defines = []
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deps = [
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"../api:libjingle_peerconnection_api",
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"../api:optional",
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"../api:ortc_api",
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"../rtc_base:checks",
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"../rtc_base:rtc_base",
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"../rtc_base:stringutils",
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"../system_wrappers:field_trial_api",
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]
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public_configs = [ ":rtc_p2p_inherited_config" ]
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if (build_with_chromium) {
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if (is_nacl) {
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deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
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}
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} else {
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sources += [
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"base/relayserver.cc",
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"base/relayserver.h",
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"base/stunserver.cc",
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"base/stunserver.h",
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"base/turnserver.cc",
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"base/turnserver.h",
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]
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defines += [
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"FEATURE_ENABLE_VOICEMAIL",
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"FEATURE_ENABLE_PSTN",
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]
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}
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}
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if (rtc_include_tests) {
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rtc_source_set("p2p_test_utils") {
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testonly = true
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sources = [
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"base/fakecandidatepair.h",
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"base/fakedtlstransport.h",
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"base/fakeicetransport.h",
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"base/fakepackettransport.h",
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"base/fakeportallocator.h",
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"base/mockicetransport.h",
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"base/testrelayserver.h",
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"base/teststunserver.cc",
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"base/teststunserver.h",
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"base/testturncustomizer.h",
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"base/testturnserver.h",
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]
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deps = [
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":rtc_p2p",
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"../api:libjingle_peerconnection_api",
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"../api:ortc_api",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../test:test_support",
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"//testing/gmock",
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]
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}
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rtc_source_set("rtc_p2p_unittests") {
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testonly = true
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sources = [
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"base/asyncstuntcpsocket_unittest.cc",
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"base/dtlstransport_unittest.cc",
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"base/p2ptransportchannel_unittest.cc",
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"base/packetlossestimator_unittest.cc",
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"base/port_unittest.cc",
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"base/portallocator_unittest.cc",
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"base/pseudotcp_unittest.cc",
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"base/relayport_unittest.cc",
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"base/relayserver_unittest.cc",
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"base/stun_unittest.cc",
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"base/stunport_unittest.cc",
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"base/stunrequest_unittest.cc",
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"base/stunserver_unittest.cc",
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"base/tcpport_unittest.cc",
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"base/transportdescriptionfactory_unittest.cc",
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"base/turnport_unittest.cc",
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"base/turnserver_unittest.cc",
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"base/udptransport_unittest.cc",
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"client/basicportallocator_unittest.cc",
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]
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deps = [
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":p2p_test_utils",
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":rtc_p2p",
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"../api:fakemetricsobserver",
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"../api:ortc_api",
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"../rtc_base:checks",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../rtc_base:stringutils",
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"../test:test_support",
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"//testing/gmock",
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"//testing/gtest",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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defines = [ "GTEST_RELATIVE_PATH" ]
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}
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}
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rtc_static_library("libstunprober") {
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visibility = [ "*" ]
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sources = [
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"stunprober/stunprober.cc",
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"stunprober/stunprober.h",
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]
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deps = [
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":rtc_p2p",
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"..:webrtc_common",
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"../:typedefs",
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"../rtc_base:checks",
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"../rtc_base:rtc_base",
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]
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}
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if (rtc_include_tests) {
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rtc_source_set("libstunprober_unittests") {
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testonly = true
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sources = [
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"stunprober/stunprober_unittest.cc",
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]
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deps = [
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":libstunprober",
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":p2p_test_utils",
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":rtc_p2p",
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"../rtc_base:checks",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_tests_utils",
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"//testing/gmock",
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"//testing/gtest",
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]
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defines = [ "GTEST_RELATIVE_PATH" ]
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}
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}
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