blob: 8a56fe765c44c196c025fc971560039653e67bec [file] [log] [blame]
henrike@webrtc.org00c3b1e2013-07-23 18:15:111# 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.org00c3b1e2013-07-23 18:15:118{
pbos@webrtc.org24e20892013-10-28 16:32:019 'includes': [
10 'build/common.gypi',
Peter Boströmbf9f73c2015-09-25 11:58:3011 'audio/webrtc_audio.gypi',
12 'call/webrtc_call.gypi',
pbos@webrtc.org24e20892013-10-28 16:32:0113 'video/webrtc_video.gypi',
14 ],
henrike@webrtc.org00c3b1e2013-07-23 18:15:1115 'targets': [
16 {
pbos@webrtc.org24e20892013-10-28 16:32:0117 'target_name': 'webrtc',
18 'type': 'static_library',
19 'sources': [
kjellander52cf08c2016-02-25 16:36:4220 'audio_receive_stream.h',
Jelena Marusic2fb88e42015-07-16 07:30:0921 'audio_send_stream.h',
solenbergb0f22c52015-11-06 23:34:4922 'audio_state.h',
pbos@webrtc.org24e20892013-10-28 16:32:0123 'call.h',
kjellander37cb0122016-05-18 12:00:5024 'config.h',
kjellander52cf08c2016-02-25 16:36:4225 'transport.h',
26 'video_receive_stream.h',
kjellander52cf08c2016-02-25 16:36:4227 'video_send_stream.h',
28
Peter Boströmbf9f73c2015-09-25 11:58:3029 '<@(webrtc_audio_sources)',
30 '<@(webrtc_call_sources)',
pbos@webrtc.org24e20892013-10-28 16:32:0131 '<@(webrtc_video_sources)',
32 ],
33 'dependencies': [
pbos@webrtc.org7e686932014-05-15 09:35:0634 'common.gyp:*',
Peter Boströmbf9f73c2015-09-25 11:58:3035 '<@(webrtc_audio_dependencies)',
36 '<@(webrtc_call_dependencies)',
pbos@webrtc.org24e20892013-10-28 16:32:0137 '<@(webrtc_video_dependencies)',
Bjorn Tereliusbd3fce52015-07-30 10:45:1838 'rtc_event_log',
pbos@webrtc.org24e20892013-10-28 16:32:0139 ],
andresp@webrtc.org0ab271b2014-09-18 08:58:1540 'conditions': [
Henrik Kjellanderd5c1d3d2015-10-08 12:40:5141 # TODO(andresp): Chromium should link directly with this and no if
42 # conditions should be needed on webrtc build files.
andresp@webrtc.org0ab271b2014-09-18 08:58:1543 ['build_with_chromium==1', {
pbos@webrtc.orge8dbbf42014-12-15 16:33:1644 'dependencies': [
kjellander@webrtc.org12971d42015-02-11 07:47:0045 '<(webrtc_root)/modules/modules.gyp:video_capture',
pbos@webrtc.orge8dbbf42014-12-15 16:33:1646 ],
47 }],
andresp@webrtc.org0ab271b2014-09-18 08:58:1548 ],
pbos@webrtc.org24e20892013-10-28 16:32:0149 },
Bjorn Tereliusbd3fce52015-07-30 10:45:1850 {
51 'target_name': 'rtc_event_log',
52 'type': 'static_library',
53 'sources': [
Peter Boströmbf9f73c2015-09-25 11:58:3054 'call/rtc_event_log.cc',
55 'call/rtc_event_log.h',
terelius450784a2016-04-22 19:40:3756 'call/rtc_event_log_helper_thread.cc',
57 'call/rtc_event_log_helper_thread.h',
Bjorn Tereliusbd3fce52015-07-30 10:45:1858 ],
59 'conditions': [
60 # If enable_protobuf is defined, we want to compile the protobuf
61 # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
62 ['enable_protobuf==1', {
63 'dependencies': [
64 'rtc_event_log_proto',
65 ],
66 'defines': [
67 'ENABLE_RTC_EVENT_LOG',
68 ],
69 }],
70 ],
71 },
kjellander827c5152016-06-14 09:09:1972 ], # targets
73 'conditions': [
74 ['include_tests==1', {
75 'includes': [
76 'webrtc_tests.gypi',
77 ],
78 }],
79 ['enable_protobuf==1', {
80 'targets': [
81 {
82 # This target should only be built if enable_protobuf is defined
83 'target_name': 'rtc_event_log_proto',
84 'type': 'static_library',
85 'sources': ['call/rtc_event_log.proto',],
86 'variables': {
87 'proto_in_dir': 'call',
88 'proto_out_dir': 'webrtc/call',
89 },
90 'includes': ['build/protoc.gypi'],
91 },
92 {
93 'target_name': 'rtc_event_log_parser',
94 'type': 'static_library',
95 'sources': [
96 'call/rtc_event_log_parser.cc',
97 'call/rtc_event_log_parser.h',
98 ],
99 'dependencies': [
100 'rtc_event_log_proto',
101 ],
102 'export_dependent_settings': [
103 'rtc_event_log_proto',
104 ],
105 },
106 ],
107 }],
108 ['include_tests==1 and enable_protobuf==1', {
109 'targets': [
110 {
111 'target_name': 'rtc_event_log2rtp_dump',
112 'type': 'executable',
113 'sources': ['call/rtc_event_log2rtp_dump.cc',],
114 'dependencies': [
115 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
116 'rtc_event_log_parser',
117 'rtc_event_log_proto',
118 'test/test.gyp:rtp_test_utils'
119 ],
120 },
121 ],
122 }],
123 ], # conditions
henrike@webrtc.org00c3b1e2013-07-23 18:15:11124}