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Issue 1227923005: Split webrtc/video into webrtc/{audio,call,video}. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 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 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330. 9 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
10 10
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171 "config.h", 171 "config.h",
172 "frame_callback.h", 172 "frame_callback.h",
173 "transport.h", 173 "transport.h",
174 ] 174 ]
175 175
176 defines = [] 176 defines = []
177 configs += [ ":common_config" ] 177 configs += [ ":common_config" ]
178 public_configs = [ ":common_inherited_config" ] 178 public_configs = [ ":common_inherited_config" ]
179 179
180 deps = [ 180 deps = [
181 "audio",
181 ":webrtc_common", 182 ":webrtc_common",
182 "base:rtc_base", 183 "base:rtc_base",
184 "call",
183 "common_audio", 185 "common_audio",
184 "common_video", 186 "common_video",
185 "modules/audio_coding", 187 "modules/audio_coding",
186 "modules/audio_conference_mixer", 188 "modules/audio_conference_mixer",
187 "modules/audio_device", 189 "modules/audio_device",
188 "modules/audio_processing", 190 "modules/audio_processing",
189 "modules/bitrate_controller", 191 "modules/bitrate_controller",
190 "modules/desktop_capture", 192 "modules/desktop_capture",
191 "modules/media_file", 193 "modules/media_file",
192 "modules/rtp_rtcp", 194 "modules/rtp_rtcp",
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240 242
241 source_set("gtest_prod") { 243 source_set("gtest_prod") {
242 sources = [ 244 sources = [
243 "test/testsupport/gtest_prod_util.h", 245 "test/testsupport/gtest_prod_util.h",
244 ] 246 ]
245 } 247 }
246 248
247 if (rtc_enable_protobuf) { 249 if (rtc_enable_protobuf) {
248 proto_library("rtc_event_log_proto") { 250 proto_library("rtc_event_log_proto") {
249 sources = [ 251 sources = [
250 "video/rtc_event_log.proto", 252 "call/rtc_event_log.proto",
251 ] 253 ]
252 proto_out_dir = "webrtc/video" 254 proto_out_dir = "webrtc/call"
253 } 255 }
254 } 256 }
255 257
256 source_set("rtc_event_log") { 258 source_set("rtc_event_log") {
257 sources = [ 259 sources = [
258 "video/rtc_event_log.cc", 260 "call/rtc_event_log.cc",
259 "video/rtc_event_log.h", 261 "call/rtc_event_log.h",
260 ] 262 ]
261 263
262 defines = [] 264 defines = []
263 configs += [ ":common_config" ] 265 configs += [ ":common_config" ]
264 public_configs = [ ":common_inherited_config" ] 266 public_configs = [ ":common_inherited_config" ]
265 267
266 deps = [ 268 deps = [
267 ":webrtc_common", 269 ":webrtc_common",
268 ] 270 ]
269 271
270 if (rtc_enable_protobuf) { 272 if (rtc_enable_protobuf) {
271 defines += [ "ENABLE_RTC_EVENT_LOG" ] 273 defines += [ "ENABLE_RTC_EVENT_LOG" ]
272 deps += [ ":rtc_event_log_proto" ] 274 deps += [ ":rtc_event_log_proto" ]
273 } 275 }
274 if (is_clang && !is_nacl) { 276 if (is_clang && !is_nacl) {
275 # Suppress warnings from Chrome's Clang plugins. 277 # Suppress warnings from Chrome's Clang plugins.
276 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. 278 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
277 configs -= [ "//build/config/clang:find_bad_constructs" ] 279 configs -= [ "//build/config/clang:find_bad_constructs" ]
278 } 280 }
279 } 281 }
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