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Side by Side Diff: webrtc/webrtc.gyp

Issue 1230973005: Adds logging of configuration information for VideoReceiveStream (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed Ivo's latest comments Created 5 years, 5 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 '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
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
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