1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
|
# Copyright (c) 2012 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.
{
'conditions': [
['include_tests==1', {
'includes': [
'libjingle/xmllite/xmllite_tests.gypi',
'libjingle/xmpp/xmpp_tests.gypi',
'p2p/p2p_tests.gypi',
'sound/sound_tests.gypi',
'webrtc_tests.gypi',
],
}],
],
'includes': [
'build/common.gypi',
'video/webrtc_video.gypi',
],
'variables': {
'webrtc_all_dependencies': [
'base/base.gyp:*',
'sound/sound.gyp:*',
'common.gyp:*',
'common_audio/common_audio.gyp:*',
'common_video/common_video.gyp:*',
'libjingle/xmllite/xmllite.gyp:*',
'libjingle/xmpp/xmpp.gyp:*',
'modules/modules.gyp:*',
'p2p/p2p.gyp:*',
'system_wrappers/source/system_wrappers.gyp:*',
'video_engine/video_engine.gyp:*',
'voice_engine/voice_engine.gyp:*',
'<(webrtc_vp8_dir)/vp8.gyp:*',
'<(webrtc_vp9_dir)/vp9.gyp:*',
],
},
'targets': [
{
'target_name': 'webrtc_all',
'type': 'none',
'dependencies': [
'<@(webrtc_all_dependencies)',
'webrtc',
],
'conditions': [
['include_tests==1', {
'dependencies': [
'common_video/common_video_unittests.gyp:*',
'system_wrappers/source/system_wrappers_tests.gyp:*',
'test/metrics.gyp:*',
'test/test.gyp:*',
'test/webrtc_test_common.gyp:webrtc_test_common_unittests',
'tools/tools.gyp:*',
'webrtc_tests',
'rtc_unittests',
],
}],
],
},
{
# TODO(pbos): This is intended to contain audio parts as well as soon as
# VoiceEngine moves to the same new API format.
'target_name': 'webrtc',
'type': 'static_library',
'sources': [
'call.h',
'config.h',
'experiments.h',
'frame_callback.h',
'transport.h',
'video_receive_stream.h',
'video_renderer.h',
'video_send_stream.h',
'<@(webrtc_video_sources)',
],
'dependencies': [
'common.gyp:*',
'<@(webrtc_video_dependencies)',
],
'conditions': [
# TODO(andresp): Chromium libpeerconnection should link directly with
# this and no if conditions should be needed on webrtc build files.
['build_with_chromium==1', {
'dependencies': [
'<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
'<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
],
}],
],
},
],
}
|
