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Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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1 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
2 # 2 #
3 # Use of this source code is governed by a BSD-style license 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 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 5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may 6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree. 7 # be found in the AUTHORS file in the root of the source tree.
8 { 8 {
9 'variables': {
10 'webrtc_all_dependencies': [
11 'base/base.gyp:*',
12 'sound/sound.gyp:*',
13 'common.gyp:*',
14 'common_audio/common_audio.gyp:*',
15 'common_video/common_video.gyp:*',
16 'media/media.gyp:*',
17 'modules/modules.gyp:*',
18 'p2p/p2p.gyp:*',
19 'system_wrappers/system_wrappers.gyp:*',
20 'tools/tools.gyp:*',
21 'voice_engine/voice_engine.gyp:*',
22 '<(webrtc_vp8_dir)/vp8.gyp:*',
23 '<(webrtc_vp9_dir)/vp9.gyp:*',
24 ],
25 },
9 'conditions': [ 26 'conditions': [
27 ['build_with_chromium==0', {
28 # TODO(kjellander): Move this to webrtc_all_dependencies once all of talk/
29 # has been moved to webrtc/. It can't be processed by Chromium since the
30 # reference to buid/java.gypi is using an absolute path (and includes
31 # entries cannot contain variables).
32 'variables': {
33 'webrtc_all_dependencies': [
34 'api/api.gyp:*',
35 ],
36 },
37 }],
10 ['include_tests==1', { 38 ['include_tests==1', {
11 'includes': [ 39 'includes': [
12 'libjingle/xmllite/xmllite_tests.gypi', 40 'libjingle/xmllite/xmllite_tests.gypi',
13 'libjingle/xmpp/xmpp_tests.gypi', 41 'libjingle/xmpp/xmpp_tests.gypi',
14 'media/media_tests.gypi', 42 'media/media_tests.gypi',
15 'p2p/p2p_tests.gypi', 43 'p2p/p2p_tests.gypi',
16 'sound/sound_tests.gypi', 44 'sound/sound_tests.gypi',
17 'webrtc_tests.gypi', 45 'webrtc_tests.gypi',
18 ], 46 ],
19 }], 47 }],
(...skipping 27 matching lines...) Expand all
47 }, 75 },
48 ], 76 ],
49 }], 77 }],
50 ], 78 ],
51 'includes': [ 79 'includes': [
52 'build/common.gypi', 80 'build/common.gypi',
53 'audio/webrtc_audio.gypi', 81 'audio/webrtc_audio.gypi',
54 'call/webrtc_call.gypi', 82 'call/webrtc_call.gypi',
55 'video/webrtc_video.gypi', 83 'video/webrtc_video.gypi',
56 ], 84 ],
57 'variables': {
58 'webrtc_all_dependencies': [
59 'base/base.gyp:*',
60 'sound/sound.gyp:*',
61 'common.gyp:*',
62 'common_audio/common_audio.gyp:*',
63 'common_video/common_video.gyp:*',
64 'media/media.gyp:*',
65 'modules/modules.gyp:*',
66 'p2p/p2p.gyp:*',
67 'system_wrappers/system_wrappers.gyp:*',
68 'tools/tools.gyp:*',
69 'voice_engine/voice_engine.gyp:*',
70 '<(webrtc_vp8_dir)/vp8.gyp:*',
71 '<(webrtc_vp9_dir)/vp9.gyp:*',
72 ],
73 },
74 'targets': [ 85 'targets': [
75 { 86 {
76 'target_name': 'webrtc_all', 87 'target_name': 'webrtc_all',
77 'type': 'none', 88 'type': 'none',
78 'dependencies': [ 89 'dependencies': [
79 '<@(webrtc_all_dependencies)', 90 '<@(webrtc_all_dependencies)',
80 'webrtc', 91 'webrtc',
81 ], 92 ],
82 'conditions': [ 93 'conditions': [
83 ['include_tests==1', { 94 ['include_tests==1', {
84 'dependencies': [ 95 'dependencies': [
96 'api/api_tests.gyp:*',
85 'common_video/common_video_unittests.gyp:*', 97 'common_video/common_video_unittests.gyp:*',
86 'rtc_unittests', 98 'rtc_unittests',
87 'system_wrappers/system_wrappers_tests.gyp:*', 99 'system_wrappers/system_wrappers_tests.gyp:*',
88 'test/metrics.gyp:*', 100 'test/metrics.gyp:*',
89 'test/test.gyp:*', 101 'test/test.gyp:*',
90 'test/webrtc_test_common.gyp:*', 102 'test/webrtc_test_common.gyp:*',
91 'webrtc_tests', 103 'webrtc_tests',
92 ], 104 ],
93 }], 105 }],
94 ['OS=="ios"', {
95 'dependencies': [
96 # TODO(tkchin): Move this target to webrtc_all_dependencies once it
97 # has more than iOS specific targets.
98 # TODO(tkchin): Figure out where to add this in BUILD.gn.
99 'api/api.gyp:*',
100 ],
101 }],
102 ], 106 ],
103 }, 107 },
104 { 108 {
105 'target_name': 'webrtc', 109 'target_name': 'webrtc',
106 'type': 'static_library', 110 'type': 'static_library',
107 'sources': [ 111 'sources': [
108 'audio_receive_stream.h', 112 'audio_receive_stream.h',
109 'audio_send_stream.h', 113 'audio_send_stream.h',
110 'audio_state.h', 114 'audio_state.h',
111 'call.h', 115 'call.h',
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155 ], 159 ],
156 'defines': [ 160 'defines': [
157 'ENABLE_RTC_EVENT_LOG', 161 'ENABLE_RTC_EVENT_LOG',
158 ], 162 ],
159 }], 163 }],
160 ], 164 ],
161 }, 165 },
162 166
163 ], 167 ],
164 } 168 }
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