Files
platform-external-webrtc/webrtc/pc/pc.gyp
kjellander c76dc95daf Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
The only thing that differs from the previous attempt in
https://codereview.webrtc.org/1979933002/ is that none of
the new targets are not hooked up to the webrtc target in
webrtc/BUILD.gn, which should make it not break the
chromium.webrtc.fyi bots.

Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.

Changes between previous attempt and the one before that
(https://codereview.webrtc.org/1973313002) are:
* Added libstunprober target
* Adjusted warnings for Chromium's clang plugins
* webrtc/pc/externalhmac.{h,cc} added for Chromium builds.

BUG=webrtc:4256
NOTRY=True
NOPRESUBMIT=True
TBR=perkj@webrtc.org, tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/2037983002
Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 10:09:40 +00:00

147 lines
4.1 KiB
Python
Executable File

# 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.
{
'includes': ['../build/common.gypi'],
'variables': {
'rtc_pc_defines': [
'SRTP_RELATIVE_PATH',
'HAVE_SCTP',
'HAVE_SRTP',
],
},
'targets': [
{
'target_name': 'rtc_pc',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/media/media.gyp:rtc_media',
],
'conditions': [
['build_with_chromium==1', {
'sources': [
'externalhmac.h',
'externalhmac.cc',
],
}],
['build_libsrtp==1', {
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
],
}],
],
'defines': [
'<@(rtc_pc_defines)',
],
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
],
'direct_dependent_settings': {
'defines': [
'<@(rtc_pc_defines)'
],
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
],
},
'sources': [
'audiomonitor.cc',
'audiomonitor.h',
'bundlefilter.cc',
'bundlefilter.h',
'channel.cc',
'channel.h',
'channelmanager.cc',
'channelmanager.h',
'currentspeakermonitor.cc',
'currentspeakermonitor.h',
'mediamonitor.cc',
'mediamonitor.h',
'mediasession.cc',
'mediasession.h',
'mediasink.h',
'rtcpmuxfilter.cc',
'rtcpmuxfilter.h',
'srtpfilter.cc',
'srtpfilter.h',
'voicechannel.h',
],
}, # target rtc_pc
], # targets
'conditions': [
['include_tests==1', {
'targets' : [
{
'target_name': 'rtc_pc_unittests',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/media/media.gyp:rtc_unittest_main',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
],
'include_dirs': [
'<(DEPTH)/third_party/libsrtp/srtp',
],
'sources': [
'bundlefilter_unittest.cc',
'channel_unittest.cc',
'channelmanager_unittest.cc',
'currentspeakermonitor_unittest.cc',
'mediasession_unittest.cc',
'rtcpmuxfilter_unittest.cc',
'srtpfilter_unittest.cc',
],
'conditions': [
['clang==0', {
'cflags': [
'-Wno-maybe-uninitialized', # Only exists for GCC.
],
}],
['build_libsrtp==1', {
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
],
}],
['OS=="win"', {
'msvs_settings': {
'VCLinkerTool': {
'AdditionalDependencies': [
'strmiids.lib',
],
},
},
}],
],
}, # target rtc_pc_unittests
], # targets
'conditions': [
['test_isolation_mode != "noop"', {
'targets': [
{
'target_name': 'rtc_pc_unittests_run',
'type': 'none',
'dependencies': [
'rtc_pc_unittests',
],
'includes': [
'../build/isolate.gypi',
],
'sources': [
'rtc_pc_unittests.isolate',
],
},
],
}],
], # conditions
}], # include_tests==1
], # conditions
}