blob: 9fe514cb01b64847e55c52aebe2398f8746b1d88 [file] [log] [blame]
henrike@webrtc.org00c3b1e2013-07-23 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.org00c3b1e2013-07-23 18:15:11 +00008{
pbos@webrtc.org24e20892013-10-28 16:32:01 +00009 'conditions': [
10 ['include_tests==1', {
11 'includes': [
12 'webrtc_tests.gypi',
13 ],
14 }],
15 ],
16 'includes': [
17 'build/common.gypi',
18 'video/webrtc_video.gypi',
19 ],
henrike@webrtc.org00c3b1e2013-07-23 18:15:11 +000020 'variables': {
21 'webrtc_all_dependencies': [
22 'common_audio/common_audio.gyp:*',
23 'common_video/common_video.gyp:*',
24 'modules/modules.gyp:*',
25 'system_wrappers/source/system_wrappers.gyp:*',
26 'video_engine/video_engine.gyp:*',
27 'voice_engine/voice_engine.gyp:*',
28 '<(webrtc_vp8_dir)/vp8.gyp:*',
29 ],
30 },
31 'targets': [
32 {
pbos@webrtc.org24e20892013-10-28 16:32:01 +000033 'target_name': 'webrtc_all',
henrike@webrtc.org00c3b1e2013-07-23 18:15:11 +000034 'type': 'none',
35 'dependencies': [
36 '<@(webrtc_all_dependencies)',
pbos@webrtc.org24e20892013-10-28 16:32:01 +000037 'webrtc',
henrike@webrtc.org00c3b1e2013-07-23 18:15:11 +000038 ],
39 'conditions': [
40 ['include_tests==1', {
41 'dependencies': [
42 'system_wrappers/source/system_wrappers_tests.gyp:*',
43 'test/metrics.gyp:*',
44 'test/test.gyp:*',
45 'tools/tools.gyp:*',
pbos@webrtc.org24e20892013-10-28 16:32:01 +000046 'webrtc_tests',
henrike@webrtc.org00c3b1e2013-07-23 18:15:11 +000047 ],
48 }],
kjellander@webrtc.org8e701082013-10-10 08:48:16 +000049 ['build_with_chromium==0 and OS=="android"', {
henrike@webrtc.org00c3b1e2013-07-23 18:15:11 +000050 'dependencies': [
51 '../tools/android/android_tools_precompiled.gyp:*',
henrike@webrtc.org00c3b1e2013-07-23 18:15:11 +000052 ],
53 }],
54 ],
55 },
pbos@webrtc.org24e20892013-10-28 16:32:01 +000056 {
57 # TODO(pbos): This is intended to contain audio parts as well as soon as
58 # VoiceEngine moves to the same new API format.
59 'target_name': 'webrtc',
60 'type': 'static_library',
61 'sources': [
62 'call.cc',
63 'call.h',
64 'config.h',
65 'frame_callback.h',
66 'transport.h',
67 'video_receive_stream.h',
68 'video_renderer.h',
69 'video_send_stream.h',
70
71 '<@(webrtc_video_sources)',
72 ],
73 'dependencies': [
74 '<@(webrtc_video_dependencies)',
75 ],
76 },
henrike@webrtc.org00c3b1e2013-07-23 18:15:11 +000077 ],
78}