Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(151)

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

Issue 2976293002: Remove remains of webrtc/base (Closed)
Patch Set: Add README.md Created 3 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/p2p/BUILD.gn ('k') | webrtc/rtc_base/BUILD.gn » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2016 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 if (is_android) { 10 if (is_android) {
(...skipping 42 matching lines...) Expand 10 before | Expand all | Expand 10 after
53 "srtpsession.cc", 53 "srtpsession.cc",
54 "srtpsession.h", 54 "srtpsession.h",
55 "voicechannel.h", 55 "voicechannel.h",
56 ] 56 ]
57 57
58 deps = [ 58 deps = [
59 "..:webrtc_common", 59 "..:webrtc_common",
60 "../api:call_api", 60 "../api:call_api",
61 "../api:libjingle_peerconnection_api", 61 "../api:libjingle_peerconnection_api",
62 "../api:ortc_api", 62 "../api:ortc_api",
63 "../base:rtc_base",
64 "../base:rtc_task_queue",
65 "../media:rtc_data", 63 "../media:rtc_data",
66 "../media:rtc_h264_profile_id", 64 "../media:rtc_h264_profile_id",
67 "../media:rtc_media_base", 65 "../media:rtc_media_base",
68 "../p2p:rtc_p2p", 66 "../p2p:rtc_p2p",
67 "../rtc_base:rtc_base",
68 "../rtc_base:rtc_task_queue",
69 ] 69 ]
70 70
71 if (rtc_build_libsrtp) { 71 if (rtc_build_libsrtp) {
72 deps += [ "//third_party/libsrtp" ] 72 deps += [ "//third_party/libsrtp" ]
73 } 73 }
74 74
75 public_configs = [ ":rtc_pc_config" ] 75 public_configs = [ ":rtc_pc_config" ]
76 76
77 if (!build_with_chromium && is_clang) { 77 if (!build_with_chromium && is_clang) {
78 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 78 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
(...skipping 79 matching lines...) Expand 10 before | Expand all | Expand 10 after
158 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 158 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
159 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 159 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
160 } 160 }
161 161
162 deps = [ 162 deps = [
163 ":rtc_pc_base", 163 ":rtc_pc_base",
164 "..:webrtc_common", 164 "..:webrtc_common",
165 "../api:call_api", 165 "../api:call_api",
166 "../api:rtc_stats_api", 166 "../api:rtc_stats_api",
167 "../api/video_codecs:video_codecs_api", 167 "../api/video_codecs:video_codecs_api",
168 "../base:rtc_base",
169 "../base:rtc_base_approved",
170 "../call:call_interfaces", 168 "../call:call_interfaces",
171 "../logging:rtc_event_log_api", 169 "../logging:rtc_event_log_api",
172 "../media:rtc_data", 170 "../media:rtc_data",
173 "../media:rtc_media_base", 171 "../media:rtc_media_base",
174 "../p2p:rtc_p2p", 172 "../p2p:rtc_p2p",
173 "../rtc_base:rtc_base",
174 "../rtc_base:rtc_base_approved",
175 "../stats", 175 "../stats",
176 "../system_wrappers:system_wrappers", 176 "../system_wrappers:system_wrappers",
177 ] 177 ]
178 178
179 public_deps = [ 179 public_deps = [
180 "../api:libjingle_peerconnection_api", 180 "../api:libjingle_peerconnection_api",
181 ] 181 ]
182 } 182 }
183 183
184 # This target implements CreatePeerConnectionFactory methods that will create a 184 # This target implements CreatePeerConnectionFactory methods that will create a
185 # PeerConnection will full functionality (audio, video and data). Applications 185 # PeerConnection will full functionality (audio, video and data). Applications
186 # that wish to reduce their binary size by ommitting functionality they don't 186 # that wish to reduce their binary size by ommitting functionality they don't
187 # need should use CreateModularCreatePeerConnectionFactory instead, using the 187 # need should use CreateModularCreatePeerConnectionFactory instead, using the
188 # "peerconnection" build target and other targets specific to their 188 # "peerconnection" build target and other targets specific to their
189 # requrements. See comment in peerconnectionfactoryinterface.h. 189 # requrements. See comment in peerconnectionfactoryinterface.h.
190 rtc_static_library("create_pc_factory") { 190 rtc_static_library("create_pc_factory") {
191 sources = [ 191 sources = [
192 "createpeerconnectionfactory.cc", 192 "createpeerconnectionfactory.cc",
193 ] 193 ]
194 194
195 deps = [ 195 deps = [
196 "../api:audio_mixer_api", 196 "../api:audio_mixer_api",
197 "../api:libjingle_peerconnection_api", 197 "../api:libjingle_peerconnection_api",
198 "../api/audio_codecs:audio_codecs_api", 198 "../api/audio_codecs:audio_codecs_api",
199 "../api/audio_codecs:builtin_audio_decoder_factory", 199 "../api/audio_codecs:builtin_audio_decoder_factory",
200 "../api/audio_codecs:builtin_audio_encoder_factory", 200 "../api/audio_codecs:builtin_audio_encoder_factory",
201 "../base:rtc_base",
202 "../base:rtc_base_approved",
203 "../call", 201 "../call",
204 "../call:call_interfaces", 202 "../call:call_interfaces",
205 "../logging:rtc_event_log_api", 203 "../logging:rtc_event_log_api",
206 "../media:rtc_audio_video", 204 "../media:rtc_audio_video",
207 "../modules/audio_device:audio_device", 205 "../modules/audio_device:audio_device",
208 "../modules/audio_processing:audio_processing", 206 "../modules/audio_processing:audio_processing",
207 "../rtc_base:rtc_base",
208 "../rtc_base:rtc_base_approved",
209 ] 209 ]
210 210
211 configs += [ ":libjingle_peerconnection_warnings_config" ] 211 configs += [ ":libjingle_peerconnection_warnings_config" ]
212 212
213 if (!build_with_chromium && is_clang) { 213 if (!build_with_chromium && is_clang) {
214 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 214 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
215 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 215 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
216 } 216 }
217 } 217 }
218 218
(...skipping 53 matching lines...) Expand 10 before | Expand all | Expand 10 after
272 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 272 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
273 } 273 }
274 274
275 if (is_win) { 275 if (is_win) {
276 libs = [ "strmiids.lib" ] 276 libs = [ "strmiids.lib" ]
277 } 277 }
278 278
279 deps = [ 279 deps = [
280 ":libjingle_peerconnection", 280 ":libjingle_peerconnection",
281 ":rtc_pc", 281 ":rtc_pc",
282 "../base:rtc_base",
283 "../base:rtc_base_approved",
284 "../base:rtc_base_tests_main",
285 "../base:rtc_base_tests_utils",
286 "../logging:rtc_event_log_api", 282 "../logging:rtc_event_log_api",
287 "../media:rtc_media_base", 283 "../media:rtc_media_base",
288 "../media:rtc_media_tests_utils", 284 "../media:rtc_media_tests_utils",
289 "../p2p:p2p_test_utils", 285 "../p2p:p2p_test_utils",
290 "../p2p:rtc_p2p", 286 "../p2p:rtc_p2p",
287 "../rtc_base:rtc_base",
288 "../rtc_base:rtc_base_approved",
289 "../rtc_base:rtc_base_tests_main",
290 "../rtc_base:rtc_base_tests_utils",
291 "../system_wrappers:metrics_default", 291 "../system_wrappers:metrics_default",
292 ] 292 ]
293 293
294 if (rtc_build_libsrtp) { 294 if (rtc_build_libsrtp) {
295 deps += [ "//third_party/libsrtp" ] 295 deps += [ "//third_party/libsrtp" ]
296 } 296 }
297 297
298 if (is_android) { 298 if (is_android) {
299 deps += [ "//testing/android/native_test:native_test_support" ] 299 deps += [ "//testing/android/native_test:native_test_support" ]
300 } 300 }
(...skipping 17 matching lines...) Expand all
318 "test/peerconnectiontestwrapper.h", 318 "test/peerconnectiontestwrapper.h",
319 "test/rtcstatsobtainer.h", 319 "test/rtcstatsobtainer.h",
320 "test/testsdpstrings.h", 320 "test/testsdpstrings.h",
321 ] 321 ]
322 322
323 deps = [ 323 deps = [
324 ":libjingle_peerconnection", 324 ":libjingle_peerconnection",
325 "..:webrtc_common", 325 "..:webrtc_common",
326 "../api:libjingle_peerconnection_test_api", 326 "../api:libjingle_peerconnection_test_api",
327 "../api:rtc_stats_api", 327 "../api:rtc_stats_api",
328 "../base:rtc_base",
329 "../base:rtc_base_approved",
330 "../base:rtc_base_tests_utils",
331 "../call:call_interfaces", 328 "../call:call_interfaces",
332 "../logging:rtc_event_log_api", 329 "../logging:rtc_event_log_api",
333 "../media:rtc_media", 330 "../media:rtc_media",
334 "../media:rtc_media_tests_utils", 331 "../media:rtc_media_tests_utils",
335 "../modules/audio_device:audio_device", 332 "../modules/audio_device:audio_device",
336 "../p2p:p2p_test_utils", 333 "../p2p:p2p_test_utils",
334 "../rtc_base:rtc_base",
335 "../rtc_base:rtc_base_approved",
336 "../rtc_base:rtc_base_tests_utils",
337 "../test:test_support", 337 "../test:test_support",
338 "//testing/gmock", 338 "//testing/gmock",
339 ] 339 ]
340 340
341 if (!build_with_chromium && is_clang) { 341 if (!build_with_chromium && is_clang) {
342 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 342 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
343 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 343 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
344 } 344 }
345 } 345 }
346 346
(...skipping 88 matching lines...) Expand 10 before | Expand all | Expand 10 after
435 "../sdk/android:libjingle_peerconnection_jni", 435 "../sdk/android:libjingle_peerconnection_jni",
436 "//testing/android/native_test:native_test_support", 436 "//testing/android/native_test:native_test_support",
437 ] 437 ]
438 } 438 }
439 439
440 deps += [ 440 deps += [
441 ":libjingle_peerconnection", 441 ":libjingle_peerconnection",
442 ":pc_test_utils", 442 ":pc_test_utils",
443 "..:webrtc_common", 443 "..:webrtc_common",
444 "../api:fakemetricsobserver", 444 "../api:fakemetricsobserver",
445 "../base:rtc_base_tests_main",
446 "../base:rtc_base_tests_utils",
447 "../media:rtc_media_tests_utils", 445 "../media:rtc_media_tests_utils",
448 "../pc:rtc_pc", 446 "../pc:rtc_pc",
447 "../rtc_base:rtc_base_tests_main",
448 "../rtc_base:rtc_base_tests_utils",
449 "../system_wrappers:metrics_default", 449 "../system_wrappers:metrics_default",
450 "../test:audio_codec_mocks", 450 "../test:audio_codec_mocks",
451 "//testing/gmock", 451 "//testing/gmock",
452 ] 452 ]
453 453
454 if (is_android) { 454 if (is_android) {
455 deps += [ "//testing/android/native_test:native_test_support" ] 455 deps += [ "//testing/android/native_test:native_test_support" ]
456 456
457 shard_timeout = 900 457 shard_timeout = 900
458 } 458 }
459 } 459 }
460 } 460 }
OLDNEW
« no previous file with comments | « webrtc/p2p/BUILD.gn ('k') | webrtc/rtc_base/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698