blob: 283aa462087dce91eb78730d6082aec04ab090d6 [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',
25 'sources': ['video/rtc_event_log.proto',],
26 'variables': {
27 'proto_in_dir': 'video',
28 'proto_out_dir': 'webrtc/video',
29 },
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',
39 'sources': ['video/rtc_event_log2rtp_dump.cc',],
40 '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',
52 'video/webrtc_video.gypi',
53 ],
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000054 'variables': {
55 'webrtc_all_dependencies': [
henrike@webrtc.orgf0488722014-05-13 18:00:26 +000056 'base/base.gyp:*',
henrike@webrtc.org66a35822014-08-26 22:04:04 +000057 'sound/sound.gyp:*',
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +000058 'common.gyp:*',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000059 'common_audio/common_audio.gyp:*',
60 'common_video/common_video.gyp:*',
61 'modules/modules.gyp:*',
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000062 'p2p/p2p.gyp:*',
andresp@webrtc.org86e1e482015-01-14 09:30:52 +000063 'system_wrappers/system_wrappers.gyp:*',
kjellander@webrtc.orgd7e34e12015-01-26 19:17:26 +000064 'tools/tools.gyp:*',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000065 'voice_engine/voice_engine.gyp:*',
66 '<(webrtc_vp8_dir)/vp8.gyp:*',
marpan@webrtc.org5b883172014-11-01 06:10:48 +000067 '<(webrtc_vp9_dir)/vp9.gyp:*',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000068 ],
69 },
70 'targets': [
71 {
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000072 'target_name': 'webrtc_all',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000073 'type': 'none',
74 'dependencies': [
75 '<@(webrtc_all_dependencies)',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000076 'webrtc',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000077 ],
78 'conditions': [
79 ['include_tests==1', {
80 'dependencies': [
pbos@webrtc.org724947b2013-12-11 16:26:16 +000081 'common_video/common_video_unittests.gyp:*',
Peter Boström2ee24392015-06-22 07:57:16 +020082 'rtc_unittests',
andresp@webrtc.org86e1e482015-01-14 09:30:52 +000083 'system_wrappers/system_wrappers_tests.gyp:*',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000084 'test/metrics.gyp:*',
85 'test/test.gyp:*',
Henrik Kjellanderafb6b5e2015-09-16 14:07:33 +020086 'test/webrtc_test_common.gyp:*',
87 'video_engine/video_engine_core_unittests.gyp:*',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000088 'webrtc_tests',
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000089 ],
90 }],
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +000091 ],
92 },
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000093 {
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000094 'target_name': 'webrtc',
95 'type': 'static_library',
96 'sources': [
Jelena Marusiccd670222015-07-16 09:30:09 +020097 'audio_receive_stream.h',
98 'audio_send_stream.h',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000099 'call.h',
100 'config.h',
101 'frame_callback.h',
Jelena Marusiccd670222015-07-16 09:30:09 +0200102 'stream.h',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000103 'transport.h',
104 'video_receive_stream.h',
105 'video_renderer.h',
106 'video_send_stream.h',
107
108 '<@(webrtc_video_sources)',
109 ],
110 'dependencies': [
pbos@webrtc.org1e92b0a2014-05-15 09:35:06 +0000111 'common.gyp:*',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000112 '<@(webrtc_video_dependencies)',
Bjorn Terelius36411852015-07-30 12:45:18 +0200113 'rtc_event_log',
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000114 ],
andresp@webrtc.orgab071da2014-09-18 08:58:15 +0000115 'conditions': [
116 # TODO(andresp): Chromium libpeerconnection should link directly with
pbos@webrtc.orga7f77722014-12-15 16:33:16 +0000117 # this and no if conditions should be needed on webrtc build files.
andresp@webrtc.orgab071da2014-09-18 08:58:15 +0000118 ['build_with_chromium==1', {
pbos@webrtc.orga7f77722014-12-15 16:33:16 +0000119 'dependencies': [
kjellander@webrtc.orgf58fe0a2015-02-11 07:47:00 +0000120 '<(webrtc_root)/modules/modules.gyp:video_capture',
121 '<(webrtc_root)/modules/modules.gyp:video_render',
pbos@webrtc.orga7f77722014-12-15 16:33:16 +0000122 ],
123 }],
andresp@webrtc.orgab071da2014-09-18 08:58:15 +0000124 ],
pbos@webrtc.org16e03b72013-10-28 16:32:01 +0000125 },
Bjorn Terelius36411852015-07-30 12:45:18 +0200126 {
127 'target_name': 'rtc_event_log',
128 'type': 'static_library',
129 'sources': [
130 'video/rtc_event_log.cc',
131 'video/rtc_event_log.h',
132 ],
133 'conditions': [
134 # If enable_protobuf is defined, we want to compile the protobuf
135 # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
136 ['enable_protobuf==1', {
137 'dependencies': [
138 'rtc_event_log_proto',
139 ],
140 'defines': [
141 'ENABLE_RTC_EVENT_LOG',
142 ],
143 }],
144 ],
145 },
146
henrike@webrtc.org8d27a1c2013-07-23 18:15:11 +0000147 ],
148}