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Side by Side Diff: webrtc/video/BUILD.gn

Issue 2974863003: Use relative paths in GN files. (Closed)
Patch Set: Created 3 years, 5 months ago
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1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2014 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 import("../webrtc.gni") 9 import("../webrtc.gni")
10 10
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100 "../logging:rtc_event_log_api", 100 "../logging:rtc_event_log_api",
101 "../media:rtc_media", 101 "../media:rtc_media",
102 "../media:rtc_media_base", 102 "../media:rtc_media_base",
103 "../modules/audio_mixer:audio_mixer_impl", 103 "../modules/audio_mixer:audio_mixer_impl",
104 "../modules/rtp_rtcp", 104 "../modules/rtp_rtcp",
105 "../modules/video_coding:webrtc_h264", 105 "../modules/video_coding:webrtc_h264",
106 "../modules/video_coding:webrtc_vp8", 106 "../modules/video_coding:webrtc_vp8",
107 "../modules/video_coding:webrtc_vp9", 107 "../modules/video_coding:webrtc_vp9",
108 "../system_wrappers", 108 "../system_wrappers",
109 "../test:test_common", 109 "../test:test_common",
110 "../test:test_renderer",
110 "../test:test_support", 111 "../test:test_support",
112 "../test:video_test_common",
111 "../voice_engine", 113 "../voice_engine",
112 "//testing/gtest", 114 "//testing/gtest",
113 "//webrtc/test:test_renderer",
114 "//webrtc/test:video_test_common",
115 ] 115 ]
116 if (!build_with_chromium && is_clang) { 116 if (!build_with_chromium && is_clang) {
117 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 117 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
118 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 118 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
119 } 119 }
120 } 120 }
121 121
122 rtc_source_set("video_full_stack_tests") { 122 rtc_source_set("video_full_stack_tests") {
123 testonly = true 123 testonly = true
124 124
125 # Skip restricting visibility on mobile platforms since the tests on those 125 # Skip restricting visibility on mobile platforms since the tests on those
126 # gets additional generated targets which would require many lines here to 126 # gets additional generated targets which would require many lines here to
127 # cover (which would be confusing to read and hard to maintain). 127 # cover (which would be confusing to read and hard to maintain).
128 if (!is_android && !is_ios) { 128 if (!is_android && !is_ios) {
129 visibility = [ "//webrtc:webrtc_perf_tests" ] 129 visibility = [ "..:webrtc_perf_tests" ]
130 } 130 }
131 sources = [ 131 sources = [
132 "full_stack_tests.cc", 132 "full_stack_tests.cc",
133 ] 133 ]
134 deps = [ 134 deps = [
135 ":video_quality_test", 135 ":video_quality_test",
136 "../modules/pacing:pacing", 136 "../modules/pacing:pacing",
137 "../test:field_trial", 137 "../test:field_trial",
138 "../test:test_common",
138 "../test:test_support", 139 "../test:test_support",
139 "//testing/gtest", 140 "//testing/gtest",
140 "//webrtc/test:test_common",
141 ] 141 ]
142 if (!build_with_chromium && is_clang) { 142 if (!build_with_chromium && is_clang) {
143 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 143 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
144 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 144 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
145 } 145 }
146 if (rtc_use_h264) { 146 if (rtc_use_h264) {
147 defines = [ "WEBRTC_USE_H264" ] 147 defines = [ "WEBRTC_USE_H264" ]
148 } 148 }
149 } 149 }
150 150
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226 } 226 }
227 227
228 # TODO(pbos): Rename test suite. 228 # TODO(pbos): Rename test suite.
229 rtc_source_set("video_tests") { 229 rtc_source_set("video_tests") {
230 testonly = true 230 testonly = true
231 231
232 # Skip restricting visibility on mobile platforms since the tests on those 232 # Skip restricting visibility on mobile platforms since the tests on those
233 # gets additional generated targets which would require many lines here to 233 # gets additional generated targets which would require many lines here to
234 # cover (which would be confusing to read and hard to maintain). 234 # cover (which would be confusing to read and hard to maintain).
235 if (!is_android && !is_ios) { 235 if (!is_android && !is_ios) {
236 visibility = [ "//webrtc:video_engine_tests" ] 236 visibility = [ "..:video_engine_tests" ]
237 } 237 }
238 defines = [] 238 defines = []
239 sources = [ 239 sources = [
240 "call_stats_unittest.cc", 240 "call_stats_unittest.cc",
241 "encoder_rtcp_feedback_unittest.cc", 241 "encoder_rtcp_feedback_unittest.cc",
242 "end_to_end_tests.cc", 242 "end_to_end_tests.cc",
243 "overuse_frame_detector_unittest.cc", 243 "overuse_frame_detector_unittest.cc",
244 "payload_router_unittest.cc", 244 "payload_router_unittest.cc",
245 "picture_id_tests.cc", 245 "picture_id_tests.cc",
246 "quality_threshold_unittest.cc", 246 "quality_threshold_unittest.cc",
(...skipping 47 matching lines...) Expand 10 before | Expand all | Expand 10 after
294 ] 294 ]
295 if (!build_with_chromium && is_clang) { 295 if (!build_with_chromium && is_clang) {
296 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 296 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
297 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 297 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
298 } 298 }
299 if (rtc_use_h264) { 299 if (rtc_use_h264) {
300 defines += [ "WEBRTC_USE_H264" ] 300 defines += [ "WEBRTC_USE_H264" ]
301 } 301 }
302 } 302 }
303 } 303 }
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