Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(235)

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

Issue 2855023003: Move RTCAudioSession* files modules/audio_device/ -> sdk/Framework. (Closed)
Patch Set: Order imports Created 3 years, 7 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/modules/audio_device/ios/voice_processing_audio_unit.mm ('k') | webrtc/sdk/objc/DEPS » ('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 2016 The WebRTC project authors. All Rights Reserved. 1 # Copyright 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_ios) { 10 if (is_ios) {
(...skipping 73 matching lines...) Expand 10 before | Expand all | Expand 10 after
84 } 84 }
85 } 85 }
86 if (!build_with_chromium) { 86 if (!build_with_chromium) {
87 sources += [ 87 sources += [
88 "objc/Framework/Classes/RTCFileLogger.mm", 88 "objc/Framework/Classes/RTCFileLogger.mm",
89 "objc/Framework/Headers/WebRTC/RTCFileLogger.h", 89 "objc/Framework/Headers/WebRTC/RTCFileLogger.h",
90 ] 90 ]
91 } 91 }
92 } 92 }
93 93
94 rtc_static_library("rtc_sdk_audio_session") {
tkchin_webrtc 2017/05/03 16:48:21 rtc_sdk_audio? There may be things in the future
95 sources = [
96 "objc/Framework/Classes/RTCAudioSession+Configuration.mm",
97 "objc/Framework/Classes/RTCAudioSession+Private.h",
98 "objc/Framework/Classes/RTCAudioSession.mm",
99 "objc/Framework/Classes/RTCAudioSessionConfiguration.m",
100 "objc/Framework/Headers/WebRTC/RTCAudioSession.h",
101 "objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h",
102 ]
103 configs += [ "..:common_objc" ]
104
105 public_configs = [ ":rtc_sdk_common_objc_config" ]
106 deps = [
107 "../base:rtc_base_approved",
108 ]
109
110 if (!build_with_chromium && is_clang) {
111 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
112 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
113 }
114 }
115
94 rtc_static_library("rtc_sdk_peerconnection_objc") { 116 rtc_static_library("rtc_sdk_peerconnection_objc") {
95 sources = [ 117 sources = [
96 "objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h", 118 "objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h",
97 "objc/Framework/Classes/RTCAVFoundationVideoCapturerInternal.h", 119 "objc/Framework/Classes/RTCAVFoundationVideoCapturerInternal.h",
98 "objc/Framework/Classes/RTCAVFoundationVideoCapturerInternal.mm", 120 "objc/Framework/Classes/RTCAVFoundationVideoCapturerInternal.mm",
99 "objc/Framework/Classes/RTCAVFoundationVideoSource+Private.h", 121 "objc/Framework/Classes/RTCAVFoundationVideoSource+Private.h",
100 "objc/Framework/Classes/RTCAVFoundationVideoSource.mm", 122 "objc/Framework/Classes/RTCAVFoundationVideoSource.mm",
101 "objc/Framework/Classes/RTCAudioSource+Private.h", 123 "objc/Framework/Classes/RTCAudioSource+Private.h",
102 "objc/Framework/Classes/RTCAudioSource.mm", 124 "objc/Framework/Classes/RTCAudioSource.mm",
103 "objc/Framework/Classes/RTCAudioTrack+Private.h", 125 "objc/Framework/Classes/RTCAudioTrack+Private.h",
(...skipping 163 matching lines...) Expand 10 before | Expand all | Expand 10 after
267 "objc/Framework/UnitTests/RTCConfigurationTest.mm", 289 "objc/Framework/UnitTests/RTCConfigurationTest.mm",
268 "objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm", 290 "objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm",
269 "objc/Framework/UnitTests/RTCIceCandidateTest.mm", 291 "objc/Framework/UnitTests/RTCIceCandidateTest.mm",
270 "objc/Framework/UnitTests/RTCIceServerTest.mm", 292 "objc/Framework/UnitTests/RTCIceServerTest.mm",
271 "objc/Framework/UnitTests/RTCMediaConstraintsTest.mm", 293 "objc/Framework/UnitTests/RTCMediaConstraintsTest.mm",
272 "objc/Framework/UnitTests/RTCPeerConnectionTest.mm", 294 "objc/Framework/UnitTests/RTCPeerConnectionTest.mm",
273 "objc/Framework/UnitTests/RTCSessionDescriptionTest.mm", 295 "objc/Framework/UnitTests/RTCSessionDescriptionTest.mm",
274 "objc/Framework/UnitTests/avformatmappertests.mm", 296 "objc/Framework/UnitTests/avformatmappertests.mm",
275 ] 297 ]
276 if (is_ios) { 298 if (is_ios) {
277 sources += [ "objc/Framework/UnitTests/RTCMTLVideoViewTests.mm" ] 299 sources += [
300 "objc/Framework/UnitTests/RTCAudioSessionTest.mm",
301 "objc/Framework/UnitTests/RTCMTLVideoViewTests.mm",
302 ]
278 if (current_cpu != "arm64") { 303 if (current_cpu != "arm64") {
279 sources += [ "objc/Framework/Classes/Metal/RTCMTLVideoView.m" ] 304 sources += [ "objc/Framework/Classes/Metal/RTCMTLVideoView.m" ]
280 } 305 }
281 } 306 }
282 307
283 # TODO(tkchin): Cleanup this warning. 308 # TODO(tkchin): Cleanup this warning.
284 cflags = [ "-Wno-objc-property-no-attribute" ] 309 cflags = [ "-Wno-objc-property-no-attribute" ]
285 310
286 # |-ObjC| flag needed to make sure category method implementations 311 # |-ObjC| flag needed to make sure category method implementations
287 # are included: 312 # are included:
(...skipping 14 matching lines...) Expand all
302 } 327 }
303 } 328 }
304 } 329 }
305 330
306 if (is_ios) { 331 if (is_ios) {
307 ios_framework_bundle("rtc_sdk_framework_objc") { 332 ios_framework_bundle("rtc_sdk_framework_objc") {
308 info_plist = "objc/Framework/Info.plist" 333 info_plist = "objc/Framework/Info.plist"
309 output_name = "WebRTC" 334 output_name = "WebRTC"
310 335
311 common_objc_headers = [ 336 common_objc_headers = [
337 "objc/Framework/Headers/WebRTC/RTCAudioSession.h",
338 "objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h",
312 "objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h", 339 "objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h",
313 "objc/Framework/Headers/WebRTC/RTCAudioSource.h", 340 "objc/Framework/Headers/WebRTC/RTCAudioSource.h",
314 "objc/Framework/Headers/WebRTC/RTCAudioTrack.h", 341 "objc/Framework/Headers/WebRTC/RTCAudioTrack.h",
315 "objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h", 342 "objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h",
316 "objc/Framework/Headers/WebRTC/RTCConfiguration.h", 343 "objc/Framework/Headers/WebRTC/RTCConfiguration.h",
317 "objc/Framework/Headers/WebRTC/RTCDataChannel.h", 344 "objc/Framework/Headers/WebRTC/RTCDataChannel.h",
318 "objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h", 345 "objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h",
319 "objc/Framework/Headers/WebRTC/RTCDispatcher.h", 346 "objc/Framework/Headers/WebRTC/RTCDispatcher.h",
320 "objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h", 347 "objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h",
321 "objc/Framework/Headers/WebRTC/RTCFieldTrials.h", 348 "objc/Framework/Headers/WebRTC/RTCFieldTrials.h",
(...skipping 92 matching lines...) Expand 10 before | Expand all | Expand 10 after
414 ":rtc_sdk_common_objc", 441 ":rtc_sdk_common_objc",
415 "../api:video_frame_api", 442 "../api:video_frame_api",
416 "../base:rtc_base_approved", 443 "../base:rtc_base_approved",
417 "../common_video", 444 "../common_video",
418 "../media:rtc_media_base", 445 "../media:rtc_media_base",
419 "../modules/video_coding:video_coding_utility", 446 "../modules/video_coding:video_coding_utility",
420 "../modules/video_coding:webrtc_h264", 447 "../modules/video_coding:webrtc_h264",
421 "../system_wrappers", 448 "../system_wrappers",
422 ] 449 ]
423 450
451 if (is_ios) {
452 deps += [ ":rtc_sdk_audio_session" ]
453 }
454
424 libs = [ 455 libs = [
425 "CoreFoundation.framework", 456 "CoreFoundation.framework",
426 "CoreMedia.framework", 457 "CoreMedia.framework",
427 "CoreVideo.framework", 458 "CoreVideo.framework",
428 "VideoToolbox.framework", 459 "VideoToolbox.framework",
429 ] 460 ]
430 461
431 if (!build_with_chromium && is_clang) { 462 if (!build_with_chromium && is_clang) {
432 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 463 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
433 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 464 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
434 } 465 }
435 466
436 if (rtc_build_libyuv) { 467 if (rtc_build_libyuv) {
437 deps += [ "$rtc_libyuv_dir" ] 468 deps += [ "$rtc_libyuv_dir" ]
438 public_deps = [ 469 public_deps = [
439 "$rtc_libyuv_dir", 470 "$rtc_libyuv_dir",
440 ] 471 ]
441 } else { 472 } else {
442 # Need to add a directory normally exported by libyuv. 473 # Need to add a directory normally exported by libyuv.
443 include_dirs = [ "$rtc_libyuv_dir/include" ] 474 include_dirs = [ "$rtc_libyuv_dir/include" ]
444 } 475 }
445 } 476 }
446 } 477 }
OLDNEW
« no previous file with comments | « webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm ('k') | webrtc/sdk/objc/DEPS » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698