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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 |
| 11 import("//build/config/crypto.gni") | 11 import("//build/config/crypto.gni") |
| 12 import("//build/config/linux/pkg_config.gni") | 12 import("//build/config/linux/pkg_config.gni") |
| 13 import("build/webrtc.gni") | 13 import("build/webrtc.gni") |
| 14 import("//third_party/protobuf/proto_library.gni") |
| 14 | 15 |
| 15 # Contains the defines and includes in common.gypi that are duplicated both as | 16 # Contains the defines and includes in common.gypi that are duplicated both as |
| 16 # target_defaults and direct_dependent_settings. | 17 # target_defaults and direct_dependent_settings. |
| 17 config("common_inherited_config") { | 18 config("common_inherited_config") { |
| 18 defines = [] | 19 defines = [] |
| 19 if (build_with_mozilla) { | 20 if (build_with_mozilla) { |
| 20 defines += [ "WEBRTC_MOZILLA_BUILD" ] | 21 defines += [ "WEBRTC_MOZILLA_BUILD" ] |
| 21 } | 22 } |
| 22 if (build_with_chromium) { | 23 if (build_with_chromium) { |
| 23 defines = [ "WEBRTC_CHROMIUM_BUILD" ] | 24 defines = [ "WEBRTC_CHROMIUM_BUILD" ] |
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| 169 | 170 |
| 170 source_set("webrtc") { | 171 source_set("webrtc") { |
| 171 sources = [ | 172 sources = [ |
| 172 "call.h", | 173 "call.h", |
| 173 "config.h", | 174 "config.h", |
| 174 "experiments.h", | 175 "experiments.h", |
| 175 "frame_callback.h", | 176 "frame_callback.h", |
| 176 "transport.h", | 177 "transport.h", |
| 177 ] | 178 ] |
| 178 | 179 |
| 180 defines = [] |
| 179 configs += [ ":common_config" ] | 181 configs += [ ":common_config" ] |
| 180 public_configs = [ ":common_inherited_config" ] | 182 public_configs = [ ":common_inherited_config" ] |
| 181 | 183 |
| 182 deps = [ | 184 deps = [ |
| 183 ":webrtc_common", | 185 ":webrtc_common", |
| 184 "base:rtc_base", | 186 "base:rtc_base", |
| 185 "common_audio", | 187 "common_audio", |
| 186 "common_video", | 188 "common_video", |
| 187 "modules/audio_coding", | 189 "modules/audio_coding", |
| 188 "modules/audio_conference_mixer", | 190 "modules/audio_conference_mixer", |
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| 200 "video", | 202 "video", |
| 201 "voice_engine", | 203 "voice_engine", |
| 202 ] | 204 ] |
| 203 | 205 |
| 204 if (build_with_chromium) { | 206 if (build_with_chromium) { |
| 205 deps += [ | 207 deps += [ |
| 206 "modules/video_capture", | 208 "modules/video_capture", |
| 207 "modules/video_render", | 209 "modules/video_render", |
| 208 ] | 210 ] |
| 209 } | 211 } |
| 212 |
| 213 if (rtc_enable_protobuf) { |
| 214 defines += [ "ENABLE_RTC_EVENT_LOG" ] |
| 215 deps += [ ":rtc_event_log_proto" ] |
| 216 } |
| 210 } | 217 } |
| 211 | 218 |
| 212 if (!build_with_chromium) { | 219 if (!build_with_chromium) { |
| 213 executable("webrtc_tests") { | 220 executable("webrtc_tests") { |
| 214 testonly = true | 221 testonly = true |
| 215 deps = [ | 222 deps = [ |
| 216 ":webrtc", | 223 ":webrtc", |
| 217 "modules/video_render:video_render_internal_impl", | 224 "modules/video_render:video_render_internal_impl", |
| 218 "modules/video_capture:video_capture_internal_impl", | 225 "modules/video_capture:video_capture_internal_impl", |
| 219 "test", | 226 "test", |
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| 233 | 240 |
| 234 configs += [ ":common_config" ] | 241 configs += [ ":common_config" ] |
| 235 public_configs = [ ":common_inherited_config" ] | 242 public_configs = [ ":common_inherited_config" ] |
| 236 } | 243 } |
| 237 | 244 |
| 238 source_set("gtest_prod") { | 245 source_set("gtest_prod") { |
| 239 sources = [ | 246 sources = [ |
| 240 "test/testsupport/gtest_prod_util.h", | 247 "test/testsupport/gtest_prod_util.h", |
| 241 ] | 248 ] |
| 242 } | 249 } |
| 250 |
| 251 if (rtc_enable_protobuf) { |
| 252 proto_library("rtc_event_log_proto") { |
| 253 sources = [ |
| 254 "video/rtc_event_log.proto", |
| 255 ] |
| 256 proto_out_dir = "webrtc/video" |
| 257 } |
| 258 } |
| 259 |
| 260 source_set("rtc_event_log") { |
| 261 sources = [ |
| 262 "video/rtc_event_log.cc", |
| 263 "video/rtc_event_log.h", |
| 264 ] |
| 265 |
| 266 defines = [] |
| 267 configs += [ ":common_config" ] |
| 268 public_configs = [ ":common_inherited_config" ] |
| 269 |
| 270 deps = [ |
| 271 ":webrtc_common", |
| 272 ] |
| 273 |
| 274 if (rtc_enable_protobuf) { |
| 275 defines += [ "ENABLE_RTC_EVENT_LOG" ] |
| 276 deps += [ ":rtc_event_log_proto" ] |
| 277 } |
| 278 if (is_clang) { |
| 279 # Suppress warnings from Chrome's Clang plugins. |
| 280 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. |
| 281 configs -= [ "//build/config/clang:find_bad_constructs" ] |
| 282 } |
| 283 } |
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