blob: 029962385069afb2c0f796c2d4699b20eacf9c48 [file] [log] [blame]
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +00001# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
2#
3# Use of this source code is governed by a BSD-style license
4# that can be found in the LICENSE file in the root of the source
5# tree. An additional intellectual property rights grant can be found
6# in the file PATENTS. All contributing project authors may
7# be found in the AUTHORS file in the root of the source tree.
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +00008{
pbos@webrtc.org16e03b72013-10-28 16:32:01 +00009 'conditions': [
10 ['include_tests==1', {
11 'includes': [
henrike@webrtc.org31b75ea2014-10-02 18:43:47 +000012 'libjingle/xmllite/xmllite_tests.gypi',
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000013 'libjingle/xmpp/xmpp_tests.gypi',
14 'p2p/p2p_tests.gypi',
henrike@webrtc.org593c3a02014-10-01 16:33:03 +000015 'sound/sound_tests.gypi',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000016 'webrtc_tests.gypi',
17 ],
18 }],
Bjorn Terelius36411852015-07-30 12:45:18 +020019 ['enable_protobuf==1', {
20 'targets': [
21 {
22 # This target should only be built if enable_protobuf is defined
23 'target_name': 'rtc_event_log_proto',
24 'type': 'static_library',
Peter Boström5c389d32015-09-25 13:58:30 +020025 'sources': ['call/rtc_event_log.proto',],
Bjorn Terelius36411852015-07-30 12:45:18 +020026 'variables': {
Peter Boström5c389d32015-09-25 13:58:30 +020027 'proto_in_dir': 'call',
28 'proto_out_dir': 'webrtc/call',
Bjorn Terelius36411852015-07-30 12:45:18 +020029 },
30 'includes': ['build/protoc.gypi'],
31 },
32 ],
33 }],
Ivo Creusene1aa5b52015-09-18 15:41:07 +020034 ['include_tests==1 and enable_protobuf==1', {
35 'targets': [
36 {
37 'target_name': 'rtc_event_log2rtp_dump',
38 'type': 'executable',
Peter Boström5c389d32015-09-25 13:58:30 +020039 'sources': ['call/rtc_event_log2rtp_dump.cc',],
Ivo Creusene1aa5b52015-09-18 15:41:07 +020040 'dependencies': [
41 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
42 'rtc_event_log',
43 'rtc_event_log_proto',
44 'test/test.gyp:rtp_test_utils'
45 ],
46 },
47 ],
48 }],
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000049 ],
50 'includes': [
51 'build/common.gypi',
Peter Boström5c389d32015-09-25 13:58:30 +020052 'audio/webrtc_audio.gypi',
53 'call/webrtc_call.gypi',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000054 'video/webrtc_video.gypi',
55 ],
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000056 'variables': {
57 'webrtc_all_dependencies': [
henrike@webrtc.orgf0488722014-05-13 18:00:26 +000058 'base/base.gyp:*',
henrike@webrtc.org66a35822014-08-26 22:04:04 +000059 'sound/sound.gyp:*',
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000060 'common.gyp:*',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000061 'common_audio/common_audio.gyp:*',
62 'common_video/common_video.gyp:*',
63 'modules/modules.gyp:*',
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000064 'p2p/p2p.gyp:*',
andresp@webrtc.org86e1e482015-01-14 09:30:52 +000065 'system_wrappers/system_wrappers.gyp:*',
kjellander@webrtc.orgd7e34e12015-01-26 19:17:26 +000066 'tools/tools.gyp:*',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000067 'voice_engine/voice_engine.gyp:*',
68 '<(webrtc_vp8_dir)/vp8.gyp:*',
marpan@webrtc.org5b883172014-11-01 06:10:48 +000069 '<(webrtc_vp9_dir)/vp9.gyp:*',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000070 ],
71 },
72 'targets': [
73 {
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000074 'target_name': 'webrtc_all',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000075 'type': 'none',
76 'dependencies': [
77 '<@(webrtc_all_dependencies)',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000078 'webrtc',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000079 ],
80 'conditions': [
81 ['include_tests==1', {
82 'dependencies': [
pbos@webrtc.org724947b2013-12-11 16:26:16 +000083 'common_video/common_video_unittests.gyp:*',
Peter Boström2ee24392015-06-22 07:57:16 +020084 'rtc_unittests',
andresp@webrtc.org86e1e482015-01-14 09:30:52 +000085 'system_wrappers/system_wrappers_tests.gyp:*',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000086 'test/metrics.gyp:*',
87 'test/test.gyp:*',
Henrik Kjellanderafb6b5e2015-09-16 14:07:33 +020088 'test/webrtc_test_common.gyp:*',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000089 'webrtc_tests',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000090 ],
91 }],
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000092 ],
93 },
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000094 {
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000095 'target_name': 'webrtc',
96 'type': 'static_library',
97 'sources': [
Jelena Marusiccd670222015-07-16 09:30:09 +020098 'audio_receive_stream.h',
99 'audio_send_stream.h',
solenberg566ef242015-11-06 15:34:49 -0800100 'audio_state.h',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000101 'call.h',
102 'config.h',
103 'frame_callback.h',
Jelena Marusiccd670222015-07-16 09:30:09 +0200104 'stream.h',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000105 'transport.h',
106 'video_receive_stream.h',
107 'video_renderer.h',
108 'video_send_stream.h',
109
Peter Boström5c389d32015-09-25 13:58:30 +0200110 '<@(webrtc_audio_sources)',
111 '<@(webrtc_call_sources)',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000112 '<@(webrtc_video_sources)',
113 ],
114 'dependencies': [
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +0000115 'common.gyp:*',
Peter Boström5c389d32015-09-25 13:58:30 +0200116 '<@(webrtc_audio_dependencies)',
117 '<@(webrtc_call_dependencies)',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000118 '<@(webrtc_video_dependencies)',
Bjorn Terelius36411852015-07-30 12:45:18 +0200119 'rtc_event_log',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000120 ],
andresp@webrtc.orgab071da2014-09-18 08:58:15 +0000121 'conditions': [
Henrik Kjellander6ffc3302015-10-08 14:40:51 +0200122 # TODO(andresp): Chromium should link directly with this and no if
123 # conditions should be needed on webrtc build files.
andresp@webrtc.orgab071da2014-09-18 08:58:15 +0000124 ['build_with_chromium==1', {
pbos@webrtc.orga7f77722014-12-15 16:33:16 +0000125 'dependencies': [
kjellander@webrtc.orgf58fe0a2015-02-11 07:47:00 +0000126 '<(webrtc_root)/modules/modules.gyp:video_capture',
127 '<(webrtc_root)/modules/modules.gyp:video_render',
pbos@webrtc.orga7f77722014-12-15 16:33:16 +0000128 ],
129 }],
andresp@webrtc.orgab071da2014-09-18 08:58:15 +0000130 ],
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000131 },
Bjorn Terelius36411852015-07-30 12:45:18 +0200132 {
133 'target_name': 'rtc_event_log',
134 'type': 'static_library',
135 'sources': [
Peter Boström5c389d32015-09-25 13:58:30 +0200136 'call/rtc_event_log.cc',
137 'call/rtc_event_log.h',
Bjorn Terelius36411852015-07-30 12:45:18 +0200138 ],
139 'conditions': [
140 # If enable_protobuf is defined, we want to compile the protobuf
141 # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
142 ['enable_protobuf==1', {
143 'dependencies': [
144 'rtc_event_log_proto',
145 ],
146 'defines': [
147 'ENABLE_RTC_EVENT_LOG',
148 ],
149 }],
150 ],
151 },
152
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +0000153 ],
154}