OLD | NEW |
---|---|
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 'conditions': [ | 9 'conditions': [ |
10 ['include_tests==1', { | 10 ['include_tests==1', { |
11 'includes': [ | 11 'includes': [ |
12 'libjingle/xmllite/xmllite_tests.gypi', | 12 'libjingle/xmllite/xmllite_tests.gypi', |
13 'libjingle/xmpp/xmpp_tests.gypi', | 13 'libjingle/xmpp/xmpp_tests.gypi', |
14 'p2p/p2p_tests.gypi', | 14 'p2p/p2p_tests.gypi', |
15 'sound/sound_tests.gypi', | 15 'sound/sound_tests.gypi', |
16 'webrtc_tests.gypi', | 16 'webrtc_tests.gypi', |
17 ], | 17 ], |
18 }], | 18 }], |
19 ['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 }], | |
19 ], | 34 ], |
20 'includes': [ | 35 'includes': [ |
21 'build/common.gypi', | 36 'build/common.gypi', |
22 'video/webrtc_video.gypi', | 37 'video/webrtc_video.gypi', |
23 ], | 38 ], |
24 'variables': { | 39 'variables': { |
25 'webrtc_all_dependencies': [ | 40 'webrtc_all_dependencies': [ |
26 'base/base.gyp:*', | 41 'base/base.gyp:*', |
27 'sound/sound.gyp:*', | 42 'sound/sound.gyp:*', |
28 'common.gyp:*', | 43 'common.gyp:*', |
(...skipping 45 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
74 'transport.h', | 89 'transport.h', |
75 'video_receive_stream.h', | 90 'video_receive_stream.h', |
76 'video_renderer.h', | 91 'video_renderer.h', |
77 'video_send_stream.h', | 92 'video_send_stream.h', |
78 | 93 |
79 '<@(webrtc_video_sources)', | 94 '<@(webrtc_video_sources)', |
80 ], | 95 ], |
81 'dependencies': [ | 96 'dependencies': [ |
82 'common.gyp:*', | 97 'common.gyp:*', |
83 '<@(webrtc_video_dependencies)', | 98 '<@(webrtc_video_dependencies)', |
99 'rtc_event_log', | |
84 ], | 100 ], |
85 'conditions': [ | 101 'conditions': [ |
86 # TODO(andresp): Chromium libpeerconnection should link directly with | 102 # TODO(andresp): Chromium libpeerconnection should link directly with |
87 # this and no if conditions should be needed on webrtc build files. | 103 # this and no if conditions should be needed on webrtc build files. |
88 ['build_with_chromium==1', { | 104 ['build_with_chromium==1', { |
89 'dependencies': [ | 105 'dependencies': [ |
90 '<(webrtc_root)/modules/modules.gyp:video_capture', | 106 '<(webrtc_root)/modules/modules.gyp:video_capture', |
91 '<(webrtc_root)/modules/modules.gyp:video_render', | 107 '<(webrtc_root)/modules/modules.gyp:video_render', |
92 ], | 108 ], |
93 }], | 109 }], |
94 ], | 110 ], |
95 }, | 111 }, |
112 { | |
113 'target_name': 'rtc_event_log', | |
114 'type': 'static_library', | |
115 'sources': [ | |
116 'video/rtc_event_log.cc', | |
117 'video/rtc_event_log.h', | |
118 ], | |
119 'conditions': [ | |
120 # If enable_protobuf is defined, we want to compile the protobuf | |
121 # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources. | |
122 ['enable_protobuf==1', { | |
123 'dependencies': [ | |
124 'rtc_event_log_proto', | |
125 ], | |
126 'defines': [ | |
127 'ENABLE_RTC_EVENT_LOG', | |
128 ], | |
129 }], | |
130 ], | |
131 }, | |
132 | |
96 ], | 133 ], |
97 } | 134 } |
135 | |
pbos-webrtc
2015/07/22 11:05:27
Remove empty lines.
terelius
2015/07/23 15:54:01
Done.
| |
136 | |
137 | |
OLD | NEW |