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Issue 2633673002: Refactor OveruseFrameDetector to use timing in us units (Closed)
Patch Set: Rebased. Created 3 years, 11 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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278 278
279 # Only the root target should depend on this. 279 # Only the root target should depend on this.
280 visibility = [ "//:default" ] 280 visibility = [ "//:default" ]
281 281
282 deps = [ 282 deps = [
283 ":rtc_unittests", 283 ":rtc_unittests",
284 ":video_engine_tests", 284 ":video_engine_tests",
285 ":webrtc_nonparallel_tests", 285 ":webrtc_nonparallel_tests",
286 ":webrtc_perf_tests", 286 ":webrtc_perf_tests",
287 "api:peerconnection_unittests", 287 "api:peerconnection_unittests",
288 "base:rtc_base_tests_utils",
288 "common_audio:common_audio_unittests", 289 "common_audio:common_audio_unittests",
289 "common_video:common_video_unittests", 290 "common_video:common_video_unittests",
290 "media:rtc_media_unittests", 291 "media:rtc_media_unittests",
291 "modules:modules_tests", 292 "modules:modules_tests",
292 "modules:modules_unittests", 293 "modules:modules_unittests",
293 "modules/audio_coding:audio_coding_tests", 294 "modules/audio_coding:audio_coding_tests",
294 "modules/audio_processing:audio_processing_tests", 295 "modules/audio_processing:audio_processing_tests",
295 "modules/rtp_rtcp:test_packet_masks_metrics", 296 "modules/rtp_rtcp:test_packet_masks_metrics",
296 "modules/video_capture:video_capture_internal_impl", 297 "modules/video_capture:video_capture_internal_impl",
297 "pc:rtc_pc_unittests", 298 "pc:rtc_pc_unittests",
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399 outputs = [ 400 outputs = [
400 "{{bundle_resources_dir}}/{{source_file_part}}", 401 "{{bundle_resources_dir}}/{{source_file_part}}",
401 ] 402 ]
402 } 403 }
403 } 404 }
404 405
405 rtc_test("video_engine_tests") { 406 rtc_test("video_engine_tests") {
406 testonly = true 407 testonly = true
407 deps = [ 408 deps = [
408 "audio:audio_tests", 409 "audio:audio_tests",
410 "base:rtc_base_tests_utils",
409 "call:call_tests", 411 "call:call_tests",
410 "modules/video_capture", 412 "modules/video_capture",
411 "test:test_common", 413 "test:test_common",
412 "test:test_main", 414 "test:test_main",
413 "video:video_tests", 415 "video:video_tests",
414 ] 416 ]
415 data = video_engine_tests_resources 417 data = video_engine_tests_resources
416 if (!build_with_chromium && is_clang) { 418 if (!build_with_chromium && is_clang) {
417 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 419 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
418 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] 420 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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497 ] 499 ]
498 500
499 deps = [ 501 deps = [
500 "//base:base_java_test_support", 502 "//base:base_java_test_support",
501 "//webrtc/examples:AppRTCMobile_javalib", 503 "//webrtc/examples:AppRTCMobile_javalib",
502 "//webrtc/sdk/android:libjingle_peerconnection_java", 504 "//webrtc/sdk/android:libjingle_peerconnection_java",
503 ] 505 ]
504 } 506 }
505 } 507 }
506 } 508 }
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