diff --git a/PRESUBMIT.py b/PRESUBMIT.py index 2a22d73f5d..3cc90cc0f9 100755 --- a/PRESUBMIT.py +++ b/PRESUBMIT.py @@ -18,6 +18,7 @@ CPPLINT_BLACKLIST = [ 'tools_webrtc', 'webrtc/api/video_codecs/video_decoder.h', 'webrtc/api/video_codecs/video_encoder.h', + 'webrtc/base', 'webrtc/examples/objc', 'webrtc/media', 'webrtc/modules/audio_coding', @@ -73,6 +74,7 @@ NATIVE_API_DIRS = ( # These directories should not be used but are maintained only to avoid breaking # some legacy downstream code. LEGACY_API_DIRS = ( + 'webrtc/base', 'webrtc/common_audio/include', 'webrtc/modules/audio_coding/include', 'webrtc/modules/audio_conference_mixer/include', @@ -89,7 +91,6 @@ LEGACY_API_DIRS = ( 'webrtc/modules/video_coding/codecs/vp8/include', 'webrtc/modules/video_coding/codecs/vp9/include', 'webrtc/modules/video_coding/include', - 'webrtc/rtc_base', 'webrtc/system_wrappers/include', 'webrtc/voice_engine/include', ) diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index 12ee93c872..9280d540bc 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -232,8 +232,8 @@ rtc_source_set("video_stream_api") { deps = [ ":webrtc_common", "api:transport_api", + "base:rtc_base_approved", "common_video:common_video", - "rtc_base:rtc_base_approved", ] } @@ -252,6 +252,7 @@ if (!build_with_chromium) { "api", "api:transport_api", "audio", + "base", "call", "common_audio", "common_video", @@ -290,6 +291,7 @@ if (!build_with_chromium) { ":video_engine_tests", ":webrtc_nonparallel_tests", ":webrtc_perf_tests", + "base:rtc_base_tests_utils", "common_audio:common_audio_unittests", "common_video:common_video_unittests", "media:rtc_media_unittests", @@ -304,7 +306,6 @@ if (!build_with_chromium) { "ortc:ortc_unittests", "pc:peerconnection_unittests", "pc:rtc_pc_unittests", - "rtc_base:rtc_base_tests_utils", "stats:rtc_stats_unittests", "system_wrappers:system_wrappers_unittests", "test", @@ -392,16 +393,16 @@ if (rtc_include_tests) { ":webrtc_common", "api:rtc_api_unittests", "api/audio_codecs/test:audio_codecs_api_unittests", + "base:rtc_base_approved_unittests", + "base:rtc_base_tests_main", + "base:rtc_base_tests_utils", + "base:rtc_base_unittests", + "base:rtc_numerics_unittests", + "base:rtc_task_queue_unittests", + "base:sequenced_task_checker_unittests", + "base:weak_ptr_unittests", "p2p:libstunprober_unittests", "p2p:rtc_p2p_unittests", - "rtc_base:rtc_base_approved_unittests", - "rtc_base:rtc_base_tests_main", - "rtc_base:rtc_base_tests_utils", - "rtc_base:rtc_base_unittests", - "rtc_base:rtc_numerics_unittests", - "rtc_base:rtc_task_queue_unittests", - "rtc_base:sequenced_task_checker_unittests", - "rtc_base:weak_ptr_unittests", "system_wrappers:metrics_default", ] @@ -439,12 +440,12 @@ if (rtc_include_tests) { testonly = true deps = [ "audio:audio_tests", + "base:rtc_base_tests_utils", # TODO(eladalon): call_tests aren't actually video-specific, so we # should move them to a more appropriate test suite. "call:call_tests", "modules/video_capture", - "rtc_base:rtc_base_tests_utils", "test:test_common", "test:test_main", "test:video_test_common", @@ -516,7 +517,7 @@ if (rtc_include_tests) { rtc_test("webrtc_nonparallel_tests") { testonly = true deps = [ - "rtc_base:rtc_base_nonparallel_tests", + "base:rtc_base_nonparallel_tests", ] if (is_android) { deps += [ "//testing/android/native_test:native_test_support" ] diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn index c413f2874a..249411b1c2 100644 --- a/webrtc/api/BUILD.gn +++ b/webrtc/api/BUILD.gn @@ -28,7 +28,7 @@ rtc_source_set("call_api") { ":audio_mixer_api", ":transport_api", "..:webrtc_common", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "audio_codecs:audio_codecs_api", ] } @@ -83,8 +83,8 @@ rtc_static_library("libjingle_peerconnection_api") { deps = [ ":rtc_stats_api", "..:webrtc_common", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", + "../base:rtc_base", + "../base:rtc_base_approved", "audio_codecs:audio_codecs_api", ] @@ -143,7 +143,7 @@ rtc_source_set("rtc_stats_api") { ] deps = [ - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } @@ -153,8 +153,8 @@ rtc_source_set("audio_mixer_api") { ] deps = [ + "../base:rtc_base_approved", "../modules:module_api", - "../rtc_base:rtc_base_approved", ] } @@ -178,7 +178,7 @@ rtc_source_set("video_frame_api") { ] deps = [ - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "../system_wrappers", ] @@ -206,7 +206,7 @@ rtc_source_set("libjingle_peerconnection_test_api") { ] deps = [ - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } @@ -235,7 +235,7 @@ if (rtc_include_tests) { ] deps = [ ":libjingle_peerconnection_api", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] if (!build_with_chromium && is_clang) { # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). diff --git a/webrtc/api/audio_codecs/BUILD.gn b/webrtc/api/audio_codecs/BUILD.gn index 2174fb1063..416ccbbf3f 100644 --- a/webrtc/api/audio_codecs/BUILD.gn +++ b/webrtc/api/audio_codecs/BUILD.gn @@ -27,7 +27,7 @@ rtc_source_set("audio_codecs_api") { ] deps = [ "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } @@ -38,8 +38,8 @@ rtc_static_library("builtin_audio_decoder_factory") { ] deps = [ ":audio_codecs_api", + "../../base:rtc_base_approved", "../../modules/audio_coding:builtin_audio_decoder_factory_internal", - "../../rtc_base:rtc_base_approved", ] } @@ -50,7 +50,7 @@ rtc_static_library("builtin_audio_encoder_factory") { ] deps = [ ":audio_codecs_api", + "../../base:rtc_base_approved", "../../modules/audio_coding:builtin_audio_encoder_factory_internal", - "../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/api/audio_codecs/g722/BUILD.gn b/webrtc/api/audio_codecs/g722/BUILD.gn index 2c1349a7c5..d2470a26e4 100644 --- a/webrtc/api/audio_codecs/g722/BUILD.gn +++ b/webrtc/api/audio_codecs/g722/BUILD.gn @@ -26,8 +26,8 @@ rtc_static_library("audio_encoder_g722") { deps = [ ":audio_encoder_g722_config", "..:audio_codecs_api", + "../../../base:rtc_base_approved", "../../../modules/audio_coding:g722", - "../../../rtc_base:rtc_base_approved", ] } @@ -39,7 +39,7 @@ rtc_static_library("audio_decoder_g722") { deps = [ "..:audio_codecs_api", "../../..:webrtc_common", + "../../../base:rtc_base_approved", "../../../modules/audio_coding:g722", - "../../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/api/audio_codecs/ilbc/BUILD.gn b/webrtc/api/audio_codecs/ilbc/BUILD.gn index 6ef8856639..bba2662ad6 100644 --- a/webrtc/api/audio_codecs/ilbc/BUILD.gn +++ b/webrtc/api/audio_codecs/ilbc/BUILD.gn @@ -26,8 +26,8 @@ rtc_static_library("audio_encoder_ilbc") { deps = [ ":audio_encoder_ilbc_config", "..:audio_codecs_api", + "../../../base:rtc_base_approved", "../../../modules/audio_coding:ilbc", - "../../../rtc_base:rtc_base_approved", ] } @@ -39,7 +39,7 @@ rtc_static_library("audio_decoder_ilbc") { deps = [ "..:audio_codecs_api", "../../..:webrtc_common", + "../../../base:rtc_base_approved", "../../../modules/audio_coding:ilbc", - "../../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/api/audio_codecs/opus/BUILD.gn b/webrtc/api/audio_codecs/opus/BUILD.gn index 29a68ff74e..c7f7ac8201 100644 --- a/webrtc/api/audio_codecs/opus/BUILD.gn +++ b/webrtc/api/audio_codecs/opus/BUILD.gn @@ -18,7 +18,7 @@ rtc_static_library("audio_encoder_opus_config") { "audio_encoder_opus_config.h", ] deps = [ - "../../../rtc_base:rtc_base_approved", + "../../../base:rtc_base_approved", ] defines = [] if (rtc_opus_variable_complexity) { @@ -35,9 +35,9 @@ rtc_source_set("audio_encoder_opus") { deps = [ ":audio_encoder_opus_config", "..:audio_codecs_api", + "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed? + "../../../base:rtc_base_approved", "../../../modules/audio_coding:webrtc_opus", - "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed? - "../../../rtc_base:rtc_base_approved", ] } @@ -49,7 +49,7 @@ rtc_static_library("audio_decoder_opus") { deps = [ "..:audio_codecs_api", "../../..:webrtc_common", + "../../../base:rtc_base_approved", "../../../modules/audio_coding:webrtc_opus", - "../../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/api/audio_codecs/test/BUILD.gn b/webrtc/api/audio_codecs/test/BUILD.gn index 4a0c878920..32cef2d5cb 100644 --- a/webrtc/api/audio_codecs/test/BUILD.gn +++ b/webrtc/api/audio_codecs/test/BUILD.gn @@ -21,8 +21,8 @@ if (rtc_include_tests) { ] deps = [ "..:audio_codecs_api", - "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed? - "../../../rtc_base:rtc_base_approved", + "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed? + "../../../base:rtc_base_approved", "../../../test:audio_codec_mocks", "../../../test:test_support", "../g722:audio_decoder_g722", diff --git a/webrtc/api/video_codecs/BUILD.gn b/webrtc/api/video_codecs/BUILD.gn index 5e27c78059..d435534d83 100644 --- a/webrtc/api/video_codecs/BUILD.gn +++ b/webrtc/api/video_codecs/BUILD.gn @@ -21,7 +21,7 @@ rtc_source_set("video_codecs_api") { deps = [ "..:video_frame_api", "../..:webrtc_common", + "../../base:rtc_base_approved", "../../common_video", - "../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn index 2b7d06fcfa..1577316cd4 100644 --- a/webrtc/audio/BUILD.gn +++ b/webrtc/audio/BUILD.gn @@ -37,6 +37,8 @@ rtc_static_library("audio") { "../api:call_api", "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_encoder_factory", + "../base:rtc_base_approved", + "../base:rtc_task_queue", "../call:call_interfaces", "../call:rtp_interfaces", "../common_audio", @@ -48,8 +50,6 @@ rtc_static_library("audio") { "../modules/pacing:pacing", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp:rtp_rtcp", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", "../system_wrappers", "../voice_engine", ] @@ -77,14 +77,14 @@ if (rtc_include_tests) { deps = [ ":audio", "../api:mock_audio_mixer", + "../base:rtc_base_approved", + "../base:rtc_task_queue", "../call:rtp_receiver", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", "../modules/congestion_controller:congestion_controller", "../modules/congestion_controller:mock_congestion_controller", "../modules/pacing:pacing", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", "../test:test_common", "../test:test_support", "utility:utility_tests", diff --git a/webrtc/audio/utility/BUILD.gn b/webrtc/audio/utility/BUILD.gn index 65f9cb0da6..ac477e4f25 100644 --- a/webrtc/audio/utility/BUILD.gn +++ b/webrtc/audio/utility/BUILD.gn @@ -21,9 +21,9 @@ rtc_static_library("audio_frame_operations") { deps = [ "../..:webrtc_common", + "../../base:rtc_base_approved", "../../modules:module_api", "../../modules/audio_coding:audio_format_conversion", - "../../rtc_base:rtc_base_approved", ] } @@ -35,8 +35,8 @@ if (rtc_include_tests) { ] deps = [ ":audio_frame_operations", + "../../base:rtc_base_approved", "../../modules:module_api", - "../../rtc_base:rtc_base_approved", "../../test:test_support", "//testing/gtest", ] diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn new file mode 100644 index 0000000000..c786f15916 --- /dev/null +++ b/webrtc/base/BUILD.gn @@ -0,0 +1,135 @@ +# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +import("//build/config/crypto.gni") +import("//build/config/ui.gni") +import("../webrtc.gni") + +if (is_android) { + import("//build/config/android/config.gni") + import("//build/config/android/rules.gni") +} +if (is_win) { + import("//build/config/clang/clang.gni") +} + +group("base") { + public_deps = [ + ":rtc_base", + ":rtc_base_approved", + ":rtc_task_queue", + ":sequenced_task_checker", + ":weak_ptr", + ] +} + +if (!rtc_build_ssl) { + config("external_ssl_library") { + assert(rtc_ssl_root != "", + "You must specify rtc_ssl_root when rtc_build_ssl==0.") + include_dirs = [ rtc_ssl_root ] + } +} + +# The targets below are deprecated and only exist here temporarily during +# refactoring. See https://bugs.webrtc.org/7634 for more details. + +group("protobuf_utils") { + public_deps = [ "../rtc_base:protobuf_utils" ] +} + +group("compile_assert_c") { + public_deps = [ "../rtc_base:compile_assert_c" ] +} + +group("rtc_base_approved") { + public_deps = [ "../rtc_base:rtc_base_approved" ] +} + +group("rtc_task_queue") { + public_deps = [ "../rtc_base:rtc_task_queue" ] +} + +group("sequenced_task_checker") { + public_deps = [ "../rtc_base:sequenced_task_checker" ] +} + +group("weak_ptr") { + public_deps = [ "../rtc_base:weak_ptr" ] +} + +group("rtc_numerics") { + public_deps = [ "../rtc_base:rtc_numerics" ] +} + +group("rtc_json") { + public_deps = [ "../rtc_base:rtc_json" ] +} + +group("rtc_base") { + public_deps = [ "../rtc_base:rtc_base" ] +} + +group("gtest_prod") { + public_deps = [ "../rtc_base:gtest_prod" ] +} + +group("rtc_base_tests_utils") { + testonly = true + public_deps = [ "../rtc_base:rtc_base_tests_utils" ] +} + +if (rtc_include_tests) { + group("rtc_base_tests_main") { + testonly = true + public_deps = [ "../rtc_base:rtc_base_tests_main" ] + } + + group("rtc_base_nonparallel_tests") { + testonly = true + public_deps = [ "../rtc_base:rtc_base_nonparallel_tests" ] + } + + group("rtc_base_approved_unittests") { + testonly = true + public_deps = [ "../rtc_base:rtc_base_approved_unittests" ] + } + + group("sequenced_task_checker_unittests") { + testonly = true + public_deps = [ "../rtc_base:sequenced_task_checker_unittests" ] + } + + group("weak_ptr_unittests") { + testonly = true + public_deps = [ "../rtc_base:weak_ptr_unittests" ] + } + + group("rtc_task_queue_unittests") { + testonly = true + public_deps = [ "../rtc_base:rtc_task_queue_unittests" ] + } + + + group("rtc_numerics_unittests") { + testonly = true + public_deps = [ "../rtc_base:rtc_numerics_unittests" ] + } + + group("rtc_base_unittests") { + testonly = true + public_deps = [ "../rtc_base:rtc_base_unittests" ] + } +} + +if (is_android) { + android_library("base_java") { + java_files = [ "Dummy.java" ] # Need one file to avoid hitting an assert. + deps = [ "../rtc_base:base_java" ] + } +} diff --git a/webrtc/base/Dummy.java b/webrtc/base/Dummy.java new file mode 100644 index 0000000000..60cd440fd4 --- /dev/null +++ b/webrtc/base/Dummy.java @@ -0,0 +1,9 @@ +/** + * This class only exists as glue in a transition. + * TODO(kjellander): Remove. + * See https://bugs.webrtc.org/7634 for more details. + */ +class Dummy { + Dummy() { + } +} diff --git a/webrtc/base/array_view.h b/webrtc/base/array_view.h new file mode 100644 index 0000000000..a451b59e2d --- /dev/null +++ b/webrtc/base/array_view.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ARRAY_VIEW_H_ +#define WEBRTC_BASE_ARRAY_VIEW_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/array_view.h" + +#endif // WEBRTC_BASE_ARRAY_VIEW_H_ diff --git a/webrtc/base/arraysize.h b/webrtc/base/arraysize.h new file mode 100644 index 0000000000..8b37efa04b --- /dev/null +++ b/webrtc/base/arraysize.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ARRAYSIZE_H_ +#define WEBRTC_BASE_ARRAYSIZE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/arraysize.h" + +#endif // WEBRTC_BASE_ARRAYSIZE_H_ diff --git a/webrtc/base/asyncinvoker-inl.h b/webrtc/base/asyncinvoker-inl.h new file mode 100644 index 0000000000..cce42264ab --- /dev/null +++ b/webrtc/base/asyncinvoker-inl.h @@ -0,0 +1,19 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ASYNCINVOKER_INL_H_ +#define WEBRTC_BASE_ASYNCINVOKER_INL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/asyncinvoker-inl.h" + +#endif // WEBRTC_BASE_ASYNCINVOKER_INL_H_ diff --git a/webrtc/base/asyncinvoker.h b/webrtc/base/asyncinvoker.h new file mode 100644 index 0000000000..0fcfc04947 --- /dev/null +++ b/webrtc/base/asyncinvoker.h @@ -0,0 +1,19 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ASYNCINVOKER_H_ +#define WEBRTC_BASE_ASYNCINVOKER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/asyncinvoker.h" + +#endif // WEBRTC_BASE_ASYNCINVOKER_H_ diff --git a/webrtc/base/asyncpacketsocket.h b/webrtc/base/asyncpacketsocket.h new file mode 100644 index 0000000000..809f1789af --- /dev/null +++ b/webrtc/base/asyncpacketsocket.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_ +#define WEBRTC_BASE_ASYNCPACKETSOCKET_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/asyncpacketsocket.h" + +#endif // WEBRTC_BASE_ASYNCPACKETSOCKET_H_ diff --git a/webrtc/base/asyncresolverinterface.h b/webrtc/base/asyncresolverinterface.h new file mode 100644 index 0000000000..b2a172fb17 --- /dev/null +++ b/webrtc/base/asyncresolverinterface.h @@ -0,0 +1,19 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ASYNCRESOLVERINTERFACE_H_ +#define WEBRTC_BASE_ASYNCRESOLVERINTERFACE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/asyncresolverinterface.h" + +#endif diff --git a/webrtc/base/asyncsocket.h b/webrtc/base/asyncsocket.h new file mode 100644 index 0000000000..9c971394d3 --- /dev/null +++ b/webrtc/base/asyncsocket.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ASYNCSOCKET_H_ +#define WEBRTC_BASE_ASYNCSOCKET_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/asyncsocket.h" + +#endif // WEBRTC_BASE_ASYNCSOCKET_H_ diff --git a/webrtc/base/asynctcpsocket.h b/webrtc/base/asynctcpsocket.h new file mode 100644 index 0000000000..d64927bcd5 --- /dev/null +++ b/webrtc/base/asynctcpsocket.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ASYNCTCPSOCKET_H_ +#define WEBRTC_BASE_ASYNCTCPSOCKET_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/asynctcpsocket.h" + +#endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_ diff --git a/webrtc/base/asyncudpsocket.h b/webrtc/base/asyncudpsocket.h new file mode 100644 index 0000000000..c3212c0cc6 --- /dev/null +++ b/webrtc/base/asyncudpsocket.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ASYNCUDPSOCKET_H_ +#define WEBRTC_BASE_ASYNCUDPSOCKET_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/asyncudpsocket.h" + +#endif // WEBRTC_BASE_ASYNCUDPSOCKET_H_ diff --git a/webrtc/base/atomicops.h b/webrtc/base/atomicops.h new file mode 100644 index 0000000000..3c3684814a --- /dev/null +++ b/webrtc/base/atomicops.h @@ -0,0 +1,19 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ATOMICOPS_H_ +#define WEBRTC_BASE_ATOMICOPS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/atomicops.h" + +#endif // WEBRTC_BASE_ATOMICOPS_H_ diff --git a/webrtc/base/base64.h b/webrtc/base/base64.h new file mode 100644 index 0000000000..1e28357a67 --- /dev/null +++ b/webrtc/base/base64.h @@ -0,0 +1,20 @@ + +//********************************************************************* +//* C_Base64 - a simple base64 encoder and decoder. +//* +//* Copyright (c) 1999, Bob Withers - bwit@pobox.com +//* +//* This code may be freely used for any purpose, either personal +//* or commercial, provided the authors copyright notice remains +//* intact. +//********************************************************************* + +#ifndef WEBRTC_BASE_BASE64_H_ +#define WEBRTC_BASE_BASE64_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/base64.h" + +#endif // WEBRTC_BASE_BASE64_H_ diff --git a/webrtc/base/basictypes.h b/webrtc/base/basictypes.h new file mode 100644 index 0000000000..42ffa5a62e --- /dev/null +++ b/webrtc/base/basictypes.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_BASICTYPES_H_ +#define WEBRTC_BASE_BASICTYPES_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/basictypes.h" + +#endif // WEBRTC_BASE_BASICTYPES_H_ diff --git a/webrtc/base/bind.h b/webrtc/base/bind.h new file mode 100644 index 0000000000..39d441f008 --- /dev/null +++ b/webrtc/base/bind.h @@ -0,0 +1,69 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Bind() is an overloaded function that converts method calls into function +// objects (aka functors). The method object is captured as a scoped_refptr<> if +// possible, and as a raw pointer otherwise. Any arguments to the method are +// captured by value. The return value of Bind is a stateful, nullary function +// object. Care should be taken about the lifetime of objects captured by +// Bind(); the returned functor knows nothing about the lifetime of a non +// ref-counted method object or any arguments passed by pointer, and calling the +// functor with a destroyed object will surely do bad things. +// +// To prevent the method object from being captured as a scoped_refptr<>, you +// can use Unretained. But this should only be done when absolutely necessary, +// and when the caller knows the extra reference isn't needed. +// +// Example usage: +// struct Foo { +// int Test1() { return 42; } +// int Test2() const { return 52; } +// int Test3(int x) { return x*x; } +// float Test4(int x, float y) { return x + y; } +// }; +// +// int main() { +// Foo foo; +// cout << rtc::Bind(&Foo::Test1, &foo)() << endl; +// cout << rtc::Bind(&Foo::Test2, &foo)() << endl; +// cout << rtc::Bind(&Foo::Test3, &foo, 3)() << endl; +// cout << rtc::Bind(&Foo::Test4, &foo, 7, 8.5f)() << endl; +// } +// +// Example usage of ref counted objects: +// struct Bar { +// int AddRef(); +// int Release(); +// +// void Test() {} +// void BindThis() { +// // The functor passed to AsyncInvoke() will keep this object alive. +// invoker.AsyncInvoke(RTC_FROM_HERE,rtc::Bind(&Bar::Test, this)); +// } +// }; +// +// int main() { +// rtc::scoped_refptr bar = new rtc::RefCountedObject(); +// auto functor = rtc::Bind(&Bar::Test, bar); +// bar = nullptr; +// // The functor stores an internal scoped_refptr, so this is safe. +// functor(); +// } +// + +#ifndef WEBRTC_BASE_BIND_H_ +#define WEBRTC_BASE_BIND_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/bind.h" + +#endif // WEBRTC_BASE_BIND_H_ diff --git a/webrtc/base/bitbuffer.h b/webrtc/base/bitbuffer.h new file mode 100644 index 0000000000..09cba3c10b --- /dev/null +++ b/webrtc/base/bitbuffer.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_BITBUFFER_H_ +#define WEBRTC_BASE_BITBUFFER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/bitbuffer.h" + +#endif // WEBRTC_BASE_BITBUFFER_H_ diff --git a/webrtc/base/buffer.h b/webrtc/base/buffer.h new file mode 100644 index 0000000000..92c85d9591 --- /dev/null +++ b/webrtc/base/buffer.h @@ -0,0 +1,18 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_BUFFER_H_ +#define WEBRTC_BASE_BUFFER_H_ + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/buffer.h" + +#endif // WEBRTC_BASE_BUFFER_H_ diff --git a/webrtc/base/bufferqueue.h b/webrtc/base/bufferqueue.h new file mode 100644 index 0000000000..3142ae3703 --- /dev/null +++ b/webrtc/base/bufferqueue.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_BUFFERQUEUE_H_ +#define WEBRTC_BASE_BUFFERQUEUE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/bufferqueue.h" + +#endif // WEBRTC_BASE_BUFFERQUEUE_H_ diff --git a/webrtc/base/bytebuffer.h b/webrtc/base/bytebuffer.h new file mode 100644 index 0000000000..0cc9a12d1c --- /dev/null +++ b/webrtc/base/bytebuffer.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_BYTEBUFFER_H_ +#define WEBRTC_BASE_BYTEBUFFER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/bytebuffer.h" + +#endif // WEBRTC_BASE_BYTEBUFFER_H_ diff --git a/webrtc/base/byteorder.h b/webrtc/base/byteorder.h new file mode 100644 index 0000000000..28cbaa577b --- /dev/null +++ b/webrtc/base/byteorder.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_BYTEORDER_H_ +#define WEBRTC_BASE_BYTEORDER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/byteorder.h" + +#endif // WEBRTC_BASE_BYTEORDER_H_ diff --git a/webrtc/base/callback.h b/webrtc/base/callback.h new file mode 100644 index 0000000000..4da1e6dfab --- /dev/null +++ b/webrtc/base/callback.h @@ -0,0 +1,70 @@ +// This file was GENERATED by command: +// pump.py callback.h.pump +// DO NOT EDIT BY HAND!!! + +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// To generate callback.h from callback.h.pump, execute: +// /home/build/google3/third_party/gtest/scripts/pump.py callback.h.pump + +// Callbacks are callable object containers. They can hold a function pointer +// or a function object and behave like a value type. Internally, data is +// reference-counted, making copies and pass-by-value inexpensive. +// +// Callbacks are typed using template arguments. The format is: +// CallbackN +// where N is the number of arguments supplied to the callable object. +// Callbacks are invoked using operator(), just like a function or a function +// object. Default-constructed callbacks are "empty," and executing an empty +// callback does nothing. A callback can be made empty by assigning it from +// a default-constructed callback. +// +// Callbacks are similar in purpose to std::function (which isn't available on +// all platforms we support) and a lightweight alternative to sigslots. Since +// they effectively hide the type of the object they call, they're useful in +// breaking dependencies between objects that need to interact with one another. +// Notably, they can hold the results of Bind(), std::bind*, etc, without +// needing +// to know the resulting object type of those calls. +// +// Sigslots, on the other hand, provide a fuller feature set, such as multiple +// subscriptions to a signal, optional thread-safety, and lifetime tracking of +// slots. When these features are needed, choose sigslots. +// +// Example: +// int sqr(int x) { return x * x; } +// struct AddK { +// int k; +// int operator()(int x) const { return x + k; } +// } add_k = {5}; +// +// Callback1 my_callback; +// cout << my_callback.empty() << endl; // true +// +// my_callback = Callback1(&sqr); +// cout << my_callback.empty() << endl; // false +// cout << my_callback(3) << endl; // 9 +// +// my_callback = Callback1(add_k); +// cout << my_callback(10) << endl; // 15 +// +// my_callback = Callback1(); +// cout << my_callback.empty() << endl; // true + +#ifndef WEBRTC_BASE_CALLBACK_H_ +#define WEBRTC_BASE_CALLBACK_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/callback.h" + +#endif // WEBRTC_BASE_CALLBACK_H_ diff --git a/webrtc/base/checks.h b/webrtc/base/checks.h new file mode 100644 index 0000000000..f56f157224 --- /dev/null +++ b/webrtc/base/checks.h @@ -0,0 +1,19 @@ +/* + * Copyright 2006 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_CHECKS_H_ +#define WEBRTC_BASE_CHECKS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/checks.h" + +#endif // WEBRTC_BASE_CHECKS_H_ diff --git a/webrtc/base/compile_assert_c.h b/webrtc/base/compile_assert_c.h new file mode 100644 index 0000000000..934cc9be7c --- /dev/null +++ b/webrtc/base/compile_assert_c.h @@ -0,0 +1,18 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_COMPILE_ASSERT_C_H_ +#define WEBRTC_BASE_COMPILE_ASSERT_C_H_ + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/compile_assert_c.h" + +#endif // WEBRTC_BASE_COMPILE_ASSERT_C_H_ diff --git a/webrtc/base/constructormagic.h b/webrtc/base/constructormagic.h new file mode 100644 index 0000000000..21652c2d3d --- /dev/null +++ b/webrtc/base/constructormagic.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_CONSTRUCTORMAGIC_H_ +#define WEBRTC_BASE_CONSTRUCTORMAGIC_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/constructormagic.h" + +#endif // WEBRTC_BASE_CONSTRUCTORMAGIC_H_ diff --git a/webrtc/base/copyonwritebuffer.h b/webrtc/base/copyonwritebuffer.h new file mode 100644 index 0000000000..6a95b31ced --- /dev/null +++ b/webrtc/base/copyonwritebuffer.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_COPYONWRITEBUFFER_H_ +#define WEBRTC_BASE_COPYONWRITEBUFFER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/copyonwritebuffer.h" + +#endif // WEBRTC_BASE_COPYONWRITEBUFFER_H_ diff --git a/webrtc/base/cpu_time.h b/webrtc/base/cpu_time.h new file mode 100644 index 0000000000..f627790822 --- /dev/null +++ b/webrtc/base/cpu_time.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_CPU_TIME_H_ +#define WEBRTC_BASE_CPU_TIME_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/cpu_time.h" + +#endif // WEBRTC_BASE_CPU_TIME_H_ diff --git a/webrtc/base/crc32.h b/webrtc/base/crc32.h new file mode 100644 index 0000000000..6854567cc6 --- /dev/null +++ b/webrtc/base/crc32.h @@ -0,0 +1,19 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_CRC32_H_ +#define WEBRTC_BASE_CRC32_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/crc32.h" + +#endif // WEBRTC_BASE_CRC32_H_ diff --git a/webrtc/base/criticalsection.h b/webrtc/base/criticalsection.h new file mode 100644 index 0000000000..ab3f542244 --- /dev/null +++ b/webrtc/base/criticalsection.h @@ -0,0 +1,18 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_CRITICALSECTION_H_ +#define WEBRTC_BASE_CRITICALSECTION_H_ + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/criticalsection.h" + +#endif // WEBRTC_BASE_CRITICALSECTION_H_ diff --git a/webrtc/base/cryptstring.h b/webrtc/base/cryptstring.h new file mode 100644 index 0000000000..1a474b43f6 --- /dev/null +++ b/webrtc/base/cryptstring.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_CRYPTSTRING_H_ +#define WEBRTC_BASE_CRYPTSTRING_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/cryptstring.h" + +#endif // WEBRTC_BASE_CRYPTSTRING_H_ diff --git a/webrtc/base/deprecation.h b/webrtc/base/deprecation.h new file mode 100644 index 0000000000..d6c5124c39 --- /dev/null +++ b/webrtc/base/deprecation.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_DEPRECATION_H_ +#define WEBRTC_BASE_DEPRECATION_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/deprecation.h" + +#endif // WEBRTC_BASE_DEPRECATION_H_ diff --git a/webrtc/base/dscp.h b/webrtc/base/dscp.h new file mode 100644 index 0000000000..1cf2756cdc --- /dev/null +++ b/webrtc/base/dscp.h @@ -0,0 +1,19 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_DSCP_H_ +#define WEBRTC_BASE_DSCP_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/dscp.h" + +#endif // WEBRTC_BASE_DSCP_H_ diff --git a/webrtc/base/event.h b/webrtc/base/event.h new file mode 100644 index 0000000000..28ff7315e4 --- /dev/null +++ b/webrtc/base/event.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_EVENT_H_ +#define WEBRTC_BASE_EVENT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/event.h" + +#endif // WEBRTC_BASE_EVENT_H_ diff --git a/webrtc/base/event_tracer.h b/webrtc/base/event_tracer.h new file mode 100644 index 0000000000..b6da14a47b --- /dev/null +++ b/webrtc/base/event_tracer.h @@ -0,0 +1,34 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// This file defines the interface for event tracing in WebRTC. +// +// Event log handlers are set through SetupEventTracer(). User of this API will +// provide two function pointers to handle event tracing calls. +// +// * GetCategoryEnabledPtr +// Event tracing system calls this function to determine if a particular +// event category is enabled. +// +// * AddTraceEventPtr +// Adds a tracing event. It is the user's responsibility to log the data +// provided. +// +// Parameters for the above two functions are described in trace_event.h. + +#ifndef WEBRTC_BASE_EVENT_TRACER_H_ +#define WEBRTC_BASE_EVENT_TRACER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/event_tracer.h" + +#endif // WEBRTC_BASE_EVENT_TRACER_H_ diff --git a/webrtc/base/fakeclock.h b/webrtc/base/fakeclock.h new file mode 100644 index 0000000000..22d640dbe5 --- /dev/null +++ b/webrtc/base/fakeclock.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FAKECLOCK_H_ +#define WEBRTC_BASE_FAKECLOCK_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/fakeclock.h" + +#endif // WEBRTC_BASE_FAKECLOCK_H_ diff --git a/webrtc/base/fakenetwork.h b/webrtc/base/fakenetwork.h new file mode 100644 index 0000000000..c2c8e6dc40 --- /dev/null +++ b/webrtc/base/fakenetwork.h @@ -0,0 +1,19 @@ +/* + * Copyright 2009 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FAKENETWORK_H_ +#define WEBRTC_BASE_FAKENETWORK_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/fakenetwork.h" + +#endif // WEBRTC_BASE_FAKENETWORK_H_ diff --git a/webrtc/base/fakesslidentity.h b/webrtc/base/fakesslidentity.h new file mode 100644 index 0000000000..da204b2ae6 --- /dev/null +++ b/webrtc/base/fakesslidentity.h @@ -0,0 +1,19 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FAKESSLIDENTITY_H_ +#define WEBRTC_BASE_FAKESSLIDENTITY_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/fakesslidentity.h" + +#endif // WEBRTC_BASE_FAKESSLIDENTITY_H_ diff --git a/webrtc/base/file.h b/webrtc/base/file.h new file mode 100644 index 0000000000..5a4465f6ac --- /dev/null +++ b/webrtc/base/file.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FILE_H_ +#define WEBRTC_BASE_FILE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/file.h" + +#endif // WEBRTC_BASE_FILE_H_ diff --git a/webrtc/base/filerotatingstream.h b/webrtc/base/filerotatingstream.h new file mode 100644 index 0000000000..26306db6e0 --- /dev/null +++ b/webrtc/base/filerotatingstream.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FILEROTATINGSTREAM_H_ +#define WEBRTC_BASE_FILEROTATINGSTREAM_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/filerotatingstream.h" + +#endif // WEBRTC_BASE_FILEROTATINGSTREAM_H_ diff --git a/webrtc/base/fileutils.h b/webrtc/base/fileutils.h new file mode 100644 index 0000000000..18de30cf4d --- /dev/null +++ b/webrtc/base/fileutils.h @@ -0,0 +1,20 @@ + +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FILEUTILS_H_ +#define WEBRTC_BASE_FILEUTILS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/fileutils.h" + +#endif // WEBRTC_BASE_FILEUTILS_H_ diff --git a/webrtc/base/firewallsocketserver.h b/webrtc/base/firewallsocketserver.h new file mode 100644 index 0000000000..18ad9bcdf3 --- /dev/null +++ b/webrtc/base/firewallsocketserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FIREWALLSOCKETSERVER_H_ +#define WEBRTC_BASE_FIREWALLSOCKETSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/firewallsocketserver.h" + +#endif // WEBRTC_BASE_FIREWALLSOCKETSERVER_H_ diff --git a/webrtc/base/flags.h b/webrtc/base/flags.h new file mode 100644 index 0000000000..9094466403 --- /dev/null +++ b/webrtc/base/flags.h @@ -0,0 +1,31 @@ +/* + * Copyright 2006 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + + +// Originally comes from shared/commandlineflags/flags.h + +// Flags are defined and declared using DEFINE_xxx and DECLARE_xxx macros, +// where xxx is the flag type. Flags are referred to via FLAG_yyy, +// where yyy is the flag name. For intialization and iteration of flags, +// see the FlagList class. For full programmatic access to any +// flag, see the Flag class. +// +// The implementation only relies and basic C++ functionality +// and needs no special library or STL support. + +#ifndef WEBRTC_BASE_FLAGS_H_ +#define WEBRTC_BASE_FLAGS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/flags.h" + +#endif // SHARED_COMMANDLINEFLAGS_FLAGS_H_ diff --git a/webrtc/base/format_macros.h b/webrtc/base/format_macros.h new file mode 100644 index 0000000000..844e71ebbb --- /dev/null +++ b/webrtc/base/format_macros.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FORMAT_MACROS_H_ +#define WEBRTC_BASE_FORMAT_MACROS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/format_macros.h" + +#endif // WEBRTC_BASE_FORMAT_MACROS_H_ diff --git a/webrtc/base/function_view.h b/webrtc/base/function_view.h new file mode 100644 index 0000000000..12300268ef --- /dev/null +++ b/webrtc/base/function_view.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_FUNCTION_VIEW_H_ +#define WEBRTC_BASE_FUNCTION_VIEW_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/function_view.h" + +#endif // WEBRTC_BASE_FUNCTION_VIEW_H_ diff --git a/webrtc/base/gtest_prod_util.h b/webrtc/base/gtest_prod_util.h new file mode 100644 index 0000000000..0c25943f2c --- /dev/null +++ b/webrtc/base/gtest_prod_util.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_GTEST_PROD_UTIL_H_ +#define WEBRTC_BASE_GTEST_PROD_UTIL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/gtest_prod_util.h" + +#endif // WEBRTC_BASE_GTEST_PROD_UTIL_H_ diff --git a/webrtc/base/gunit.h b/webrtc/base/gunit.h new file mode 100644 index 0000000000..d6c092e029 --- /dev/null +++ b/webrtc/base/gunit.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_GUNIT_H_ +#define WEBRTC_BASE_GUNIT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/gunit.h" + +#endif // WEBRTC_BASE_GUNIT_H_ diff --git a/webrtc/base/gunit_prod.h b/webrtc/base/gunit_prod.h new file mode 100644 index 0000000000..436abee92a --- /dev/null +++ b/webrtc/base/gunit_prod.h @@ -0,0 +1,18 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_GUNIT_PROD_H_ +#define WEBRTC_BASE_GUNIT_PROD_H_ + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/gunit_prod.h" + +#endif // WEBRTC_BASE_GUNIT_PROD_H_ diff --git a/webrtc/base/helpers.h b/webrtc/base/helpers.h new file mode 100644 index 0000000000..86a388e8b0 --- /dev/null +++ b/webrtc/base/helpers.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_HELPERS_H_ +#define WEBRTC_BASE_HELPERS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/helpers.h" + +#endif // WEBRTC_BASE_HELPERS_H_ diff --git a/webrtc/base/httpbase.h b/webrtc/base/httpbase.h new file mode 100644 index 0000000000..a66ce15a7f --- /dev/null +++ b/webrtc/base/httpbase.h @@ -0,0 +1,20 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + + +#ifndef WEBRTC_BASE_HTTPBASE_H_ +#define WEBRTC_BASE_HTTPBASE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/httpbase.h" + +#endif // WEBRTC_BASE_HTTPBASE_H_ diff --git a/webrtc/base/httpcommon-inl.h b/webrtc/base/httpcommon-inl.h new file mode 100644 index 0000000000..7dfe18242d --- /dev/null +++ b/webrtc/base/httpcommon-inl.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_HTTPCOMMON_INL_H_ +#define WEBRTC_BASE_HTTPCOMMON_INL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/httpcommon-inl.h" + +#endif // WEBRTC_BASE_HTTPCOMMON_INL_H_ diff --git a/webrtc/base/httpcommon.h b/webrtc/base/httpcommon.h new file mode 100644 index 0000000000..3946dfcd77 --- /dev/null +++ b/webrtc/base/httpcommon.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_HTTPCOMMON_H_ +#define WEBRTC_BASE_HTTPCOMMON_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/httpcommon.h" + +#endif // WEBRTC_BASE_HTTPCOMMON_H_ diff --git a/webrtc/base/httpserver.h b/webrtc/base/httpserver.h new file mode 100644 index 0000000000..4fd75a2a05 --- /dev/null +++ b/webrtc/base/httpserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_HTTPSERVER_H_ +#define WEBRTC_BASE_HTTPSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/httpserver.h" + +#endif // WEBRTC_BASE_HTTPSERVER_H_ diff --git a/webrtc/base/ifaddrs-android.h b/webrtc/base/ifaddrs-android.h new file mode 100644 index 0000000000..9c49c9ffb0 --- /dev/null +++ b/webrtc/base/ifaddrs-android.h @@ -0,0 +1,19 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_IFADDRS_ANDROID_H_ +#define WEBRTC_BASE_IFADDRS_ANDROID_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/ifaddrs-android.h" + +#endif // WEBRTC_BASE_IFADDRS_ANDROID_H_ diff --git a/webrtc/base/ifaddrs_converter.h b/webrtc/base/ifaddrs_converter.h new file mode 100644 index 0000000000..de7ad87eee --- /dev/null +++ b/webrtc/base/ifaddrs_converter.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_IFADDRS_CONVERTER_H_ +#define WEBRTC_BASE_IFADDRS_CONVERTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/ifaddrs_converter.h" + +#endif // WEBRTC_BASE_IFADDRS_CONVERTER_H_ diff --git a/webrtc/base/ignore_wundef.h b/webrtc/base/ignore_wundef.h new file mode 100644 index 0000000000..fdfba9b84a --- /dev/null +++ b/webrtc/base/ignore_wundef.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_IGNORE_WUNDEF_H_ +#define WEBRTC_BASE_IGNORE_WUNDEF_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/ignore_wundef.h" + +#endif // WEBRTC_BASE_IGNORE_WUNDEF_H_ diff --git a/webrtc/base/ipaddress.h b/webrtc/base/ipaddress.h new file mode 100644 index 0000000000..44e432d2c8 --- /dev/null +++ b/webrtc/base/ipaddress.h @@ -0,0 +1,19 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_IPADDRESS_H_ +#define WEBRTC_BASE_IPADDRESS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/ipaddress.h" + +#endif // WEBRTC_BASE_IPADDRESS_H_ diff --git a/webrtc/base/json.h b/webrtc/base/json.h new file mode 100644 index 0000000000..175028f607 --- /dev/null +++ b/webrtc/base/json.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_JSON_H_ +#define WEBRTC_BASE_JSON_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/json.h" + +#endif // WEBRTC_BASE_JSON_H_ diff --git a/webrtc/base/keep_ref_until_done.h b/webrtc/base/keep_ref_until_done.h new file mode 100644 index 0000000000..171e04886d --- /dev/null +++ b/webrtc/base/keep_ref_until_done.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_ +#define WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/keep_ref_until_done.h" + +#endif // WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_ diff --git a/webrtc/base/location.h b/webrtc/base/location.h new file mode 100644 index 0000000000..432471c013 --- /dev/null +++ b/webrtc/base/location.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_LOCATION_H_ +#define WEBRTC_BASE_LOCATION_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/location.h" + +#endif // WEBRTC_BASE_LOCATION_H_ diff --git a/webrtc/base/logging.h b/webrtc/base/logging.h new file mode 100644 index 0000000000..594d9c992a --- /dev/null +++ b/webrtc/base/logging.h @@ -0,0 +1,54 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// LOG(...) an ostream target that can be used to send formatted +// output to a variety of logging targets, such as debugger console, stderr, +// or any LogSink. +// The severity level passed as the first argument to the LOGging +// functions is used as a filter, to limit the verbosity of the logging. +// Static members of LogMessage documented below are used to control the +// verbosity and target of the output. +// There are several variations on the LOG macro which facilitate logging +// of common error conditions, detailed below. + +// LOG(sev) logs the given stream at severity "sev", which must be a +// compile-time constant of the LoggingSeverity type, without the namespace +// prefix. +// LOG_V(sev) Like LOG(), but sev is a run-time variable of the LoggingSeverity +// type (basically, it just doesn't prepend the namespace). +// LOG_F(sev) Like LOG(), but includes the name of the current function. +// LOG_T(sev) Like LOG(), but includes the this pointer. +// LOG_T_F(sev) Like LOG_F(), but includes the this pointer. +// LOG_GLE(M)(sev [, mod]) attempt to add a string description of the +// HRESULT returned by GetLastError. The "M" variant allows searching of a +// DLL's string table for the error description. +// LOG_ERRNO(sev) attempts to add a string description of an errno-derived +// error. errno and associated facilities exist on both Windows and POSIX, +// but on Windows they only apply to the C/C++ runtime. +// LOG_ERR(sev) is an alias for the platform's normal error system, i.e. _GLE on +// Windows and _ERRNO on POSIX. +// (The above three also all have _EX versions that let you specify the error +// code, rather than using the last one.) +// LOG_E(sev, ctx, err, ...) logs a detailed error interpreted using the +// specified context. +// LOG_CHECK_LEVEL(sev) (and LOG_CHECK_LEVEL_V(sev)) can be used as a test +// before performing expensive or sensitive operations whose sole purpose is +// to output logging data at the desired level. +// Lastly, PLOG(sev, err) is an alias for LOG_ERR_EX. + +#ifndef WEBRTC_BASE_LOGGING_H_ +#define WEBRTC_BASE_LOGGING_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/logging.h" + +#endif // WEBRTC_BASE_LOGGING_H_ diff --git a/webrtc/base/logsinks.h b/webrtc/base/logsinks.h new file mode 100644 index 0000000000..95e6dc6154 --- /dev/null +++ b/webrtc/base/logsinks.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_LOGSINKS_H_ +#define WEBRTC_BASE_LOGSINKS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/logsinks.h" + +#endif // WEBRTC_BASE_LOGSINKS_H_ diff --git a/webrtc/base/macutils.h b/webrtc/base/macutils.h new file mode 100644 index 0000000000..ed0c4f5473 --- /dev/null +++ b/webrtc/base/macutils.h @@ -0,0 +1,19 @@ +/* + * Copyright 2007 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_MACUTILS_H_ +#define WEBRTC_BASE_MACUTILS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/macutils.h" + +#endif // WEBRTC_BASE_MACUTILS_H_ diff --git a/webrtc/base/mathutils.h b/webrtc/base/mathutils.h new file mode 100644 index 0000000000..9e5c3cab78 --- /dev/null +++ b/webrtc/base/mathutils.h @@ -0,0 +1,19 @@ +/* + * Copyright 2005 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_MATHUTILS_H_ +#define WEBRTC_BASE_MATHUTILS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/mathutils.h" + +#endif // WEBRTC_BASE_MATHUTILS_H_ diff --git a/webrtc/base/md5.h b/webrtc/base/md5.h new file mode 100644 index 0000000000..fd17541960 --- /dev/null +++ b/webrtc/base/md5.h @@ -0,0 +1,31 @@ +/* + * This is the header file for the MD5 message-digest algorithm. + * The algorithm is due to Ron Rivest. This code was + * written by Colin Plumb in 1993, no copyright is claimed. + * This code is in the public domain; do with it what you wish. + * + * Equivalent code is available from RSA Data Security, Inc. + * This code has been tested against that, and is equivalent, + * except that you don't need to include two pages of legalese + * with every copy. + * To compute the message digest of a chunk of bytes, declare an + * MD5Context structure, pass it to MD5Init, call MD5Update as + * needed on buffers full of bytes, and then call MD5Final, which + * will fill a supplied 16-byte array with the digest. + * + */ + +// Changes(fbarchard): Ported to C++ and Google style guide. +// Made context first parameter in MD5Final for consistency with Sha1. +// Changes(hellner): added rtc namespace +// Changes(pbos): Reverted types back to uint32(8)_t with _t suffix. + +#ifndef WEBRTC_BASE_MD5_H_ +#define WEBRTC_BASE_MD5_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/md5.h" + +#endif // WEBRTC_BASE_MD5_H_ diff --git a/webrtc/base/md5digest.h b/webrtc/base/md5digest.h new file mode 100644 index 0000000000..66d6ee1820 --- /dev/null +++ b/webrtc/base/md5digest.h @@ -0,0 +1,19 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_MD5DIGEST_H_ +#define WEBRTC_BASE_MD5DIGEST_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/md5digest.h" + +#endif // WEBRTC_BASE_MD5DIGEST_H_ diff --git a/webrtc/base/memory_usage.h b/webrtc/base/memory_usage.h new file mode 100644 index 0000000000..5c225597a7 --- /dev/null +++ b/webrtc/base/memory_usage.h @@ -0,0 +1,18 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef WEBRTC_BASE_MEMORY_USAGE_H_ +#define WEBRTC_BASE_MEMORY_USAGE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/memory_usage.h" + +#endif // WEBRTC_BASE_MEMORY_USAGE_H_ diff --git a/webrtc/base/messagedigest.h b/webrtc/base/messagedigest.h new file mode 100644 index 0000000000..b73f9079c8 --- /dev/null +++ b/webrtc/base/messagedigest.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_MESSAGEDIGEST_H_ +#define WEBRTC_BASE_MESSAGEDIGEST_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/messagedigest.h" + +#endif // WEBRTC_BASE_MESSAGEDIGEST_H_ diff --git a/webrtc/base/messagehandler.h b/webrtc/base/messagehandler.h new file mode 100644 index 0000000000..943d0d7d9b --- /dev/null +++ b/webrtc/base/messagehandler.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_MESSAGEHANDLER_H_ +#define WEBRTC_BASE_MESSAGEHANDLER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/messagehandler.h" + +#endif // WEBRTC_BASE_MESSAGEHANDLER_H_ diff --git a/webrtc/base/messagequeue.h b/webrtc/base/messagequeue.h new file mode 100644 index 0000000000..353a4b7725 --- /dev/null +++ b/webrtc/base/messagequeue.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_MESSAGEQUEUE_H_ +#define WEBRTC_BASE_MESSAGEQUEUE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/messagequeue.h" + +#endif // WEBRTC_BASE_MESSAGEQUEUE_H_ diff --git a/webrtc/base/mod_ops.h b/webrtc/base/mod_ops.h new file mode 100644 index 0000000000..d61bd055e7 --- /dev/null +++ b/webrtc/base/mod_ops.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_MOD_OPS_H_ +#define WEBRTC_BASE_MOD_OPS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/mod_ops.h" + +#endif // WEBRTC_BASE_MOD_OPS_H_ diff --git a/webrtc/base/natserver.h b/webrtc/base/natserver.h new file mode 100644 index 0000000000..b803ad8587 --- /dev/null +++ b/webrtc/base/natserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NATSERVER_H_ +#define WEBRTC_BASE_NATSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/natserver.h" + +#endif // WEBRTC_BASE_NATSERVER_H_ diff --git a/webrtc/base/natsocketfactory.h b/webrtc/base/natsocketfactory.h new file mode 100644 index 0000000000..31c29ab277 --- /dev/null +++ b/webrtc/base/natsocketfactory.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NATSOCKETFACTORY_H_ +#define WEBRTC_BASE_NATSOCKETFACTORY_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/natsocketfactory.h" + +#endif // WEBRTC_BASE_NATSOCKETFACTORY_H_ diff --git a/webrtc/base/nattypes.h b/webrtc/base/nattypes.h new file mode 100644 index 0000000000..001f57fe7d --- /dev/null +++ b/webrtc/base/nattypes.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NATTYPES_H_ +#define WEBRTC_BASE_NATTYPES_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/nattypes.h" + +#endif // WEBRTC_BASE_NATTYPES_H_ diff --git a/webrtc/base/nethelpers.h b/webrtc/base/nethelpers.h new file mode 100644 index 0000000000..9a8e6073dd --- /dev/null +++ b/webrtc/base/nethelpers.h @@ -0,0 +1,19 @@ +/* + * Copyright 2008 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NETHELPERS_H_ +#define WEBRTC_BASE_NETHELPERS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/nethelpers.h" + +#endif // WEBRTC_BASE_NETHELPERS_H_ diff --git a/webrtc/base/network.h b/webrtc/base/network.h new file mode 100644 index 0000000000..29530987c1 --- /dev/null +++ b/webrtc/base/network.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NETWORK_H_ +#define WEBRTC_BASE_NETWORK_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/network.h" + +#endif // WEBRTC_BASE_NETWORK_H_ diff --git a/webrtc/base/networkmonitor.h b/webrtc/base/networkmonitor.h new file mode 100644 index 0000000000..290da4f48e --- /dev/null +++ b/webrtc/base/networkmonitor.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NETWORKMONITOR_H_ +#define WEBRTC_BASE_NETWORKMONITOR_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/networkmonitor.h" + +#endif // WEBRTC_BASE_NETWORKMONITOR_H_ diff --git a/webrtc/base/networkroute.h b/webrtc/base/networkroute.h new file mode 100644 index 0000000000..b5e8c13842 --- /dev/null +++ b/webrtc/base/networkroute.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NETWORKROUTE_H_ +#define WEBRTC_BASE_NETWORKROUTE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/networkroute.h" + +#endif // WEBRTC_BASE_NETWORKROUTE_H_ diff --git a/webrtc/base/nullsocketserver.h b/webrtc/base/nullsocketserver.h new file mode 100644 index 0000000000..214c542b5f --- /dev/null +++ b/webrtc/base/nullsocketserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NULLSOCKETSERVER_H_ +#define WEBRTC_BASE_NULLSOCKETSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/nullsocketserver.h" + +#endif // WEBRTC_BASE_NULLSOCKETSERVER_H_ diff --git a/webrtc/base/numerics/exp_filter.h b/webrtc/base/numerics/exp_filter.h new file mode 100644 index 0000000000..a4eaea2c91 --- /dev/null +++ b/webrtc/base/numerics/exp_filter.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NUMERICS_EXP_FILTER_H_ +#define WEBRTC_BASE_NUMERICS_EXP_FILTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/numerics/exp_filter.h" + +#endif // WEBRTC_BASE_NUMERICS_EXP_FILTER_H_ diff --git a/webrtc/base/numerics/percentile_filter.h b/webrtc/base/numerics/percentile_filter.h new file mode 100644 index 0000000000..a9058a2b4c --- /dev/null +++ b/webrtc/base/numerics/percentile_filter.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_ +#define WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/numerics/percentile_filter.h" + +#endif // WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_ diff --git a/webrtc/base/onetimeevent.h b/webrtc/base/onetimeevent.h new file mode 100644 index 0000000000..6849bac581 --- /dev/null +++ b/webrtc/base/onetimeevent.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ONETIMEEVENT_H_ +#define WEBRTC_BASE_ONETIMEEVENT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/onetimeevent.h" + +#endif // WEBRTC_BASE_ONETIMEEVENT_H_ diff --git a/webrtc/base/openssl.h b/webrtc/base/openssl.h new file mode 100644 index 0000000000..795af70321 --- /dev/null +++ b/webrtc/base/openssl.h @@ -0,0 +1,19 @@ +/* + * Copyright 2013 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_OPENSSL_H_ +#define WEBRTC_BASE_OPENSSL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/openssl.h" + +#endif // WEBRTC_BASE_OPENSSL_H_ diff --git a/webrtc/base/openssladapter.h b/webrtc/base/openssladapter.h new file mode 100644 index 0000000000..6444215098 --- /dev/null +++ b/webrtc/base/openssladapter.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_OPENSSLADAPTER_H_ +#define WEBRTC_BASE_OPENSSLADAPTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/openssladapter.h" + +#endif // WEBRTC_BASE_OPENSSLADAPTER_H_ diff --git a/webrtc/base/openssldigest.h b/webrtc/base/openssldigest.h new file mode 100644 index 0000000000..031c0b1cb0 --- /dev/null +++ b/webrtc/base/openssldigest.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_OPENSSLDIGEST_H_ +#define WEBRTC_BASE_OPENSSLDIGEST_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/openssldigest.h" + +#endif // WEBRTC_BASE_OPENSSLDIGEST_H_ diff --git a/webrtc/base/opensslidentity.h b/webrtc/base/opensslidentity.h new file mode 100644 index 0000000000..59fa571ce5 --- /dev/null +++ b/webrtc/base/opensslidentity.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_OPENSSLIDENTITY_H_ +#define WEBRTC_BASE_OPENSSLIDENTITY_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/opensslidentity.h" + +#endif // WEBRTC_BASE_OPENSSLIDENTITY_H_ diff --git a/webrtc/base/opensslstreamadapter.h b/webrtc/base/opensslstreamadapter.h new file mode 100644 index 0000000000..e17e029ffe --- /dev/null +++ b/webrtc/base/opensslstreamadapter.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_ +#define WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/opensslstreamadapter.h" + +#endif // WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_ diff --git a/webrtc/base/optional.h b/webrtc/base/optional.h new file mode 100644 index 0000000000..7657ec3366 --- /dev/null +++ b/webrtc/base/optional.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_OPTIONAL_H_ +#define WEBRTC_BASE_OPTIONAL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/optional.h" + +#endif // WEBRTC_BASE_OPTIONAL_H_ diff --git a/webrtc/base/optionsfile.h b/webrtc/base/optionsfile.h new file mode 100644 index 0000000000..e77fd8adfc --- /dev/null +++ b/webrtc/base/optionsfile.h @@ -0,0 +1,19 @@ +/* + * Copyright 2008 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_OPTIONSFILE_H_ +#define WEBRTC_BASE_OPTIONSFILE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/optionsfile.h" + +#endif // WEBRTC_BASE_OPTIONSFILE_H_ diff --git a/webrtc/base/pathutils.h b/webrtc/base/pathutils.h new file mode 100644 index 0000000000..b45ca04f7c --- /dev/null +++ b/webrtc/base/pathutils.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_PATHUTILS_H_ +#define WEBRTC_BASE_PATHUTILS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/pathutils.h" + +#endif // WEBRTC_BASE_PATHUTILS_H_ diff --git a/webrtc/base/physicalsocketserver.h b/webrtc/base/physicalsocketserver.h new file mode 100644 index 0000000000..63e6dfa5b9 --- /dev/null +++ b/webrtc/base/physicalsocketserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_PHYSICALSOCKETSERVER_H_ +#define WEBRTC_BASE_PHYSICALSOCKETSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/physicalsocketserver.h" + +#endif // WEBRTC_BASE_PHYSICALSOCKETSERVER_H_ diff --git a/webrtc/base/platform_file.h b/webrtc/base/platform_file.h new file mode 100644 index 0000000000..c7396ec4c7 --- /dev/null +++ b/webrtc/base/platform_file.h @@ -0,0 +1,19 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_PLATFORM_FILE_H_ +#define WEBRTC_BASE_PLATFORM_FILE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/platform_file.h" + +#endif // WEBRTC_BASE_PLATFORM_FILE_H_ diff --git a/webrtc/base/platform_thread.h b/webrtc/base/platform_thread.h new file mode 100644 index 0000000000..626d66fc07 --- /dev/null +++ b/webrtc/base/platform_thread.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_PLATFORM_THREAD_H_ +#define WEBRTC_BASE_PLATFORM_THREAD_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/platform_thread.h" + +#endif // WEBRTC_BASE_PLATFORM_THREAD_H_ diff --git a/webrtc/base/platform_thread_types.h b/webrtc/base/platform_thread_types.h new file mode 100644 index 0000000000..f2dbd58363 --- /dev/null +++ b/webrtc/base/platform_thread_types.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_ +#define WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/platform_thread_types.h" + +#endif // WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_ diff --git a/webrtc/base/protobuf_utils.h b/webrtc/base/protobuf_utils.h new file mode 100644 index 0000000000..3d2dd862ff --- /dev/null +++ b/webrtc/base/protobuf_utils.h @@ -0,0 +1,21 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include + +#ifndef WEBRTC_BASE_PROTOBUF_UTILS_H_ +#define WEBRTC_BASE_PROTOBUF_UTILS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/protobuf_utils.h" + +#endif // WEBRTC_BASE_PROTOBUF_UTILS_H_ diff --git a/webrtc/base/proxyinfo.h b/webrtc/base/proxyinfo.h new file mode 100644 index 0000000000..f0ae1825e4 --- /dev/null +++ b/webrtc/base/proxyinfo.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_PROXYINFO_H_ +#define WEBRTC_BASE_PROXYINFO_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/proxyinfo.h" + +#endif // WEBRTC_BASE_PROXYINFO_H_ diff --git a/webrtc/base/proxyserver.h b/webrtc/base/proxyserver.h new file mode 100644 index 0000000000..1bf580ad70 --- /dev/null +++ b/webrtc/base/proxyserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_PROXYSERVER_H_ +#define WEBRTC_BASE_PROXYSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/proxyserver.h" + +#endif // WEBRTC_BASE_PROXYSERVER_H_ diff --git a/webrtc/base/ptr_util.h b/webrtc/base/ptr_util.h new file mode 100644 index 0000000000..aa6f3b4016 --- /dev/null +++ b/webrtc/base/ptr_util.h @@ -0,0 +1,21 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// This implementation is borrowed from chromium. + +#ifndef WEBRTC_BASE_PTR_UTIL_H_ +#define WEBRTC_BASE_PTR_UTIL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/ptr_util.h" + +#endif // WEBRTC_BASE_PTR_UTIL_H_ diff --git a/webrtc/base/race_checker.h b/webrtc/base/race_checker.h new file mode 100644 index 0000000000..474fdb59ba --- /dev/null +++ b/webrtc/base/race_checker.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_RACE_CHECKER_H_ +#define WEBRTC_BASE_RACE_CHECKER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/race_checker.h" + +#endif // WEBRTC_BASE_RACE_CHECKER_H_ diff --git a/webrtc/base/random.h b/webrtc/base/random.h new file mode 100644 index 0000000000..12a490202b --- /dev/null +++ b/webrtc/base/random.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_RANDOM_H_ +#define WEBRTC_BASE_RANDOM_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/random.h" + +#endif // WEBRTC_BASE_RANDOM_H_ diff --git a/webrtc/base/rate_limiter.h b/webrtc/base/rate_limiter.h new file mode 100644 index 0000000000..0cba5fb9a9 --- /dev/null +++ b/webrtc/base/rate_limiter.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_RATE_LIMITER_H_ +#define WEBRTC_BASE_RATE_LIMITER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/rate_limiter.h" + +#endif // WEBRTC_BASE_RATE_LIMITER_H_ diff --git a/webrtc/base/rate_statistics.h b/webrtc/base/rate_statistics.h new file mode 100644 index 0000000000..1a17500727 --- /dev/null +++ b/webrtc/base/rate_statistics.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_RATE_STATISTICS_H_ +#define WEBRTC_BASE_RATE_STATISTICS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/rate_statistics.h" + +#endif // WEBRTC_BASE_RATE_STATISTICS_H_ diff --git a/webrtc/base/ratelimiter.h b/webrtc/base/ratelimiter.h new file mode 100644 index 0000000000..0e372db691 --- /dev/null +++ b/webrtc/base/ratelimiter.h @@ -0,0 +1,19 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_RATELIMITER_H_ +#define WEBRTC_BASE_RATELIMITER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/ratelimiter.h" + +#endif // WEBRTC_BASE_RATELIMITER_H_ diff --git a/webrtc/base/ratetracker.h b/webrtc/base/ratetracker.h new file mode 100644 index 0000000000..d1fd75d0ee --- /dev/null +++ b/webrtc/base/ratetracker.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_RATETRACKER_H_ +#define WEBRTC_BASE_RATETRACKER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/ratetracker.h" + +#endif // WEBRTC_BASE_RATETRACKER_H_ diff --git a/webrtc/base/refcount.h b/webrtc/base/refcount.h new file mode 100644 index 0000000000..4a7cea313f --- /dev/null +++ b/webrtc/base/refcount.h @@ -0,0 +1,18 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef WEBRTC_BASE_REFCOUNT_H_ +#define WEBRTC_BASE_REFCOUNT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/refcount.h" + +#endif // WEBRTC_BASE_REFCOUNT_H_ diff --git a/webrtc/base/refcountedobject.h b/webrtc/base/refcountedobject.h new file mode 100644 index 0000000000..78304fa5f5 --- /dev/null +++ b/webrtc/base/refcountedobject.h @@ -0,0 +1,18 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef WEBRTC_BASE_REFCOUNTEDOBJECT_H_ +#define WEBRTC_BASE_REFCOUNTEDOBJECT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/refcountedobject.h" + +#endif // WEBRTC_BASE_REFCOUNTEDOBJECT_H_ diff --git a/webrtc/base/rollingaccumulator.h b/webrtc/base/rollingaccumulator.h new file mode 100644 index 0000000000..a7de4f19dd --- /dev/null +++ b/webrtc/base/rollingaccumulator.h @@ -0,0 +1,19 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_ROLLINGACCUMULATOR_H_ +#define WEBRTC_BASE_ROLLINGACCUMULATOR_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/rollingaccumulator.h" + +#endif // WEBRTC_BASE_ROLLINGACCUMULATOR_H_ diff --git a/webrtc/base/rtccertificate.h b/webrtc/base/rtccertificate.h new file mode 100644 index 0000000000..22d8fe754b --- /dev/null +++ b/webrtc/base/rtccertificate.h @@ -0,0 +1,19 @@ +/* + * Copyright 2015 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_RTCCERTIFICATE_H_ +#define WEBRTC_BASE_RTCCERTIFICATE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/rtccertificate.h" + +#endif // WEBRTC_BASE_RTCCERTIFICATE_H_ diff --git a/webrtc/base/rtccertificategenerator.h b/webrtc/base/rtccertificategenerator.h new file mode 100644 index 0000000000..fac1cec9ef --- /dev/null +++ b/webrtc/base/rtccertificategenerator.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_ +#define WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/rtccertificategenerator.h" + +#endif // WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_ diff --git a/webrtc/base/safe_compare.h b/webrtc/base/safe_compare.h new file mode 100644 index 0000000000..acdd9cebd7 --- /dev/null +++ b/webrtc/base/safe_compare.h @@ -0,0 +1,39 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// This file defines six constexpr functions: +// +// rtc::SafeEq // == +// rtc::SafeNe // != +// rtc::SafeLt // < +// rtc::SafeLe // <= +// rtc::SafeGt // > +// rtc::SafeGe // >= +// +// They each accept two arguments of arbitrary types, and in almost all cases, +// they simply call the appropriate comparison operator. However, if both +// arguments are integers, they don't compare them using C++'s quirky rules, +// but instead adhere to the true mathematical definitions. It is as if the +// arguments were first converted to infinite-range signed integers, and then +// compared, although of course nothing expensive like that actually takes +// place. In practice, for signed/signed and unsigned/unsigned comparisons and +// some mixed-signed comparisons with a compile-time constant, the overhead is +// zero; in the remaining cases, it is just a few machine instructions (no +// branches). + +#ifndef WEBRTC_BASE_SAFE_COMPARE_H_ +#define WEBRTC_BASE_SAFE_COMPARE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/safe_compare.h" + +#endif // WEBRTC_BASE_SAFE_COMPARE_H_ diff --git a/webrtc/base/safe_conversions.h b/webrtc/base/safe_conversions.h new file mode 100644 index 0000000000..ac0bb651f3 --- /dev/null +++ b/webrtc/base/safe_conversions.h @@ -0,0 +1,21 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Borrowed from Chromium's src/base/numerics/safe_conversions.h. + +#ifndef WEBRTC_BASE_SAFE_CONVERSIONS_H_ +#define WEBRTC_BASE_SAFE_CONVERSIONS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/safe_conversions.h" + +#endif // WEBRTC_BASE_SAFE_CONVERSIONS_H_ diff --git a/webrtc/base/safe_conversions_impl.h b/webrtc/base/safe_conversions_impl.h new file mode 100644 index 0000000000..497e156dbb --- /dev/null +++ b/webrtc/base/safe_conversions_impl.h @@ -0,0 +1,21 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Borrowed from Chromium's src/base/numerics/safe_conversions_impl.h. + +#ifndef WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_ +#define WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/safe_conversions_impl.h" + +#endif // WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_ diff --git a/webrtc/base/safe_minmax.h b/webrtc/base/safe_minmax.h new file mode 100644 index 0000000000..54d99b720b --- /dev/null +++ b/webrtc/base/safe_minmax.h @@ -0,0 +1,18 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SAFE_MINMAX_H_ +#define WEBRTC_BASE_SAFE_MINMAX_H_ + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/safe_minmax.h" + +#endif // WEBRTC_BASE_SAFE_MINMAX_H_ diff --git a/webrtc/base/sanitizer.h b/webrtc/base/sanitizer.h new file mode 100644 index 0000000000..56a5e103f7 --- /dev/null +++ b/webrtc/base/sanitizer.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SANITIZER_H_ +#define WEBRTC_BASE_SANITIZER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sanitizer.h" + +#endif // WEBRTC_BASE_SANITIZER_H_ diff --git a/webrtc/base/scoped_ref_ptr.h b/webrtc/base/scoped_ref_ptr.h new file mode 100644 index 0000000000..259956292f --- /dev/null +++ b/webrtc/base/scoped_ref_ptr.h @@ -0,0 +1,71 @@ +/* + * Copyright 2011 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Originally these classes are from Chromium. +// http://src.chromium.org/viewvc/chrome/trunk/src/base/memory/ref_counted.h?view=markup + +// +// A smart pointer class for reference counted objects. Use this class instead +// of calling AddRef and Release manually on a reference counted object to +// avoid common memory leaks caused by forgetting to Release an object +// reference. Sample usage: +// +// class MyFoo : public RefCounted { +// ... +// }; +// +// void some_function() { +// scoped_refptr foo = new MyFoo(); +// foo->Method(param); +// // |foo| is released when this function returns +// } +// +// void some_other_function() { +// scoped_refptr foo = new MyFoo(); +// ... +// foo = nullptr; // explicitly releases |foo| +// ... +// if (foo) +// foo->Method(param); +// } +// +// The above examples show how scoped_refptr acts like a pointer to T. +// Given two scoped_refptr classes, it is also possible to exchange +// references between the two objects, like so: +// +// { +// scoped_refptr a = new MyFoo(); +// scoped_refptr b; +// +// b.swap(a); +// // now, |b| references the MyFoo object, and |a| references null. +// } +// +// To make both |a| and |b| in the above example reference the same MyFoo +// object, simply use the assignment operator: +// +// { +// scoped_refptr a = new MyFoo(); +// scoped_refptr b; +// +// b = a; +// // now, |a| and |b| each own a reference to the same MyFoo object. +// } +// + +#ifndef WEBRTC_BASE_SCOPED_REF_PTR_H_ +#define WEBRTC_BASE_SCOPED_REF_PTR_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/scoped_ref_ptr.h" + +#endif // WEBRTC_BASE_SCOPED_REF_PTR_H_ diff --git a/webrtc/base/sequenced_task_checker.h b/webrtc/base/sequenced_task_checker.h new file mode 100644 index 0000000000..e586b8d6da --- /dev/null +++ b/webrtc/base/sequenced_task_checker.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_ +#define WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sequenced_task_checker.h" + +#endif // WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_ diff --git a/webrtc/base/sequenced_task_checker_impl.h b/webrtc/base/sequenced_task_checker_impl.h new file mode 100644 index 0000000000..4972539e66 --- /dev/null +++ b/webrtc/base/sequenced_task_checker_impl.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_ +#define WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sequenced_task_checker_impl.h" + +#endif // WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_ diff --git a/webrtc/base/sha1.h b/webrtc/base/sha1.h new file mode 100644 index 0000000000..fde3e598c3 --- /dev/null +++ b/webrtc/base/sha1.h @@ -0,0 +1,18 @@ +/* + * SHA-1 in C + * By Steve Reid + * 100% Public Domain + * +*/ + +// Ported to C++, Google style, under namespace rtc. + +#ifndef WEBRTC_BASE_SHA1_H_ +#define WEBRTC_BASE_SHA1_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sha1.h" + +#endif // WEBRTC_BASE_SHA1_H_ diff --git a/webrtc/base/sha1digest.h b/webrtc/base/sha1digest.h new file mode 100644 index 0000000000..e3b4ef840b --- /dev/null +++ b/webrtc/base/sha1digest.h @@ -0,0 +1,19 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SHA1DIGEST_H_ +#define WEBRTC_BASE_SHA1DIGEST_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sha1digest.h" + +#endif // WEBRTC_BASE_SHA1DIGEST_H_ diff --git a/webrtc/base/sigslot.h b/webrtc/base/sigslot.h new file mode 100644 index 0000000000..9d31441a49 --- /dev/null +++ b/webrtc/base/sigslot.h @@ -0,0 +1,104 @@ +// sigslot.h: Signal/Slot classes +// +// Written by Sarah Thompson (sarah@telergy.com) 2002. +// +// License: Public domain. You are free to use this code however you like, with +// the proviso that the author takes on no responsibility or liability for any +// use. +// +// QUICK DOCUMENTATION +// +// (see also the full documentation at http://sigslot.sourceforge.net/) +// +// #define switches +// SIGSLOT_PURE_ISO: +// Define this to force ISO C++ compliance. This also disables all of +// the thread safety support on platforms where it is available. +// +// SIGSLOT_USE_POSIX_THREADS: +// Force use of Posix threads when using a C++ compiler other than gcc +// on a platform that supports Posix threads. (When using gcc, this is +// the default - use SIGSLOT_PURE_ISO to disable this if necessary) +// +// SIGSLOT_DEFAULT_MT_POLICY: +// Where thread support is enabled, this defaults to +// multi_threaded_global. Otherwise, the default is single_threaded. +// #define this yourself to override the default. In pure ISO mode, +// anything other than single_threaded will cause a compiler error. +// +// PLATFORM NOTES +// +// Win32: +// On Win32, the WEBRTC_WIN symbol must be #defined. Most mainstream +// compilers do this by default, but you may need to define it yourself +// if your build environment is less standard. This causes the Win32 +// thread support to be compiled in and used automatically. +// +// Unix/Linux/BSD, etc.: +// If you're using gcc, it is assumed that you have Posix threads +// available, so they are used automatically. You can override this (as +// under Windows) with the SIGSLOT_PURE_ISO switch. If you're using +// something other than gcc but still want to use Posix threads, you +// need to #define SIGSLOT_USE_POSIX_THREADS. +// +// ISO C++: +// If none of the supported platforms are detected, or if +// SIGSLOT_PURE_ISO is defined, all multithreading support is turned +// off, along with any code that might cause a pure ISO C++ environment +// to complain. Before you ask, gcc -ansi -pedantic won't compile this +// library, but gcc -ansi is fine. Pedantic mode seems to throw a lot of +// errors that aren't really there. If you feel like investigating this, +// please contact the author. +// +// +// THREADING MODES +// +// single_threaded: +// Your program is assumed to be single threaded from the point of view +// of signal/slot usage (i.e. all objects using signals and slots are +// created and destroyed from a single thread). Behaviour if objects are +// destroyed concurrently is undefined (i.e. you'll get the occasional +// segmentation fault/memory exception). +// +// multi_threaded_global: +// Your program is assumed to be multi threaded. Objects using signals +// and slots can be safely created and destroyed from any thread, even +// when connections exist. In multi_threaded_global mode, this is +// achieved by a single global mutex (actually a critical section on +// Windows because they are faster). This option uses less OS resources, +// but results in more opportunities for contention, possibly resulting +// in more context switches than are strictly necessary. +// +// multi_threaded_local: +// Behaviour in this mode is essentially the same as +// multi_threaded_global, except that each signal, and each object that +// inherits has_slots, all have their own mutex/critical section. In +// practice, this means that mutex collisions (and hence context +// switches) only happen if they are absolutely essential. However, on +// some platforms, creating a lot of mutexes can slow down the whole OS, +// so use this option with care. +// +// USING THE LIBRARY +// +// See the full documentation at http://sigslot.sourceforge.net/ +// +// Libjingle specific: +// +// This file has been modified such that has_slots and signalx do not have to be +// using the same threading requirements. E.g. it is possible to connect a +// has_slots and signal0 or +// has_slots and signal0. +// If has_slots is single threaded the user must ensure that it is not trying +// to connect or disconnect to signalx concurrently or data race may occur. +// If signalx is single threaded the user must ensure that disconnect, connect +// or signal is not happening concurrently or data race may occur. + +#ifndef WEBRTC_BASE_SIGSLOT_H_ +#define WEBRTC_BASE_SIGSLOT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sigslot.h" + +#endif // WEBRTC_BASE_SIGSLOT_H_ diff --git a/webrtc/base/sigslottester.h b/webrtc/base/sigslottester.h new file mode 100644 index 0000000000..545bf9e235 --- /dev/null +++ b/webrtc/base/sigslottester.h @@ -0,0 +1,23 @@ +// This file was GENERATED by command: +// pump.py sigslottester.h.pump +// DO NOT EDIT BY HAND!!! + +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SIGSLOTTESTER_H_ +#define WEBRTC_BASE_SIGSLOTTESTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sigslottester.h" + +#endif // WEBRTC_BASE_SIGSLOTTESTER_H_ diff --git a/webrtc/base/socket.h b/webrtc/base/socket.h new file mode 100644 index 0000000000..19ea7a032f --- /dev/null +++ b/webrtc/base/socket.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SOCKET_H_ +#define WEBRTC_BASE_SOCKET_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/socket.h" + +#endif // WEBRTC_BASE_SOCKET_H_ diff --git a/webrtc/base/socket_unittest.h b/webrtc/base/socket_unittest.h new file mode 100644 index 0000000000..f6769f9470 --- /dev/null +++ b/webrtc/base/socket_unittest.h @@ -0,0 +1,19 @@ +/* + * Copyright 2009 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SOCKET_UNITTEST_H_ +#define WEBRTC_BASE_SOCKET_UNITTEST_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/socket_unittest.h" + +#endif // WEBRTC_BASE_SOCKET_UNITTEST_H_ diff --git a/webrtc/base/socketadapters.h b/webrtc/base/socketadapters.h new file mode 100644 index 0000000000..7df0f3ae2f --- /dev/null +++ b/webrtc/base/socketadapters.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SOCKETADAPTERS_H_ +#define WEBRTC_BASE_SOCKETADAPTERS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/socketadapters.h" + +#endif // WEBRTC_BASE_SOCKETADAPTERS_H_ diff --git a/webrtc/base/socketaddress.h b/webrtc/base/socketaddress.h new file mode 100644 index 0000000000..20199ad96b --- /dev/null +++ b/webrtc/base/socketaddress.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SOCKETADDRESS_H_ +#define WEBRTC_BASE_SOCKETADDRESS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/socketaddress.h" + +#endif // WEBRTC_BASE_SOCKETADDRESS_H_ diff --git a/webrtc/base/socketaddresspair.h b/webrtc/base/socketaddresspair.h new file mode 100644 index 0000000000..3f53f10fee --- /dev/null +++ b/webrtc/base/socketaddresspair.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SOCKETADDRESSPAIR_H_ +#define WEBRTC_BASE_SOCKETADDRESSPAIR_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/socketaddresspair.h" + +#endif // WEBRTC_BASE_SOCKETADDRESSPAIR_H_ diff --git a/webrtc/base/socketfactory.h b/webrtc/base/socketfactory.h new file mode 100644 index 0000000000..3a829ac10d --- /dev/null +++ b/webrtc/base/socketfactory.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SOCKETFACTORY_H_ +#define WEBRTC_BASE_SOCKETFACTORY_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/socketfactory.h" + +#endif // WEBRTC_BASE_SOCKETFACTORY_H_ diff --git a/webrtc/base/socketserver.h b/webrtc/base/socketserver.h new file mode 100644 index 0000000000..55b427da7e --- /dev/null +++ b/webrtc/base/socketserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SOCKETSERVER_H_ +#define WEBRTC_BASE_SOCKETSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/socketserver.h" + +#endif // WEBRTC_BASE_SOCKETSERVER_H_ diff --git a/webrtc/base/socketstream.h b/webrtc/base/socketstream.h new file mode 100644 index 0000000000..a76ffb3814 --- /dev/null +++ b/webrtc/base/socketstream.h @@ -0,0 +1,19 @@ +/* + * Copyright 2005 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SOCKETSTREAM_H_ +#define WEBRTC_BASE_SOCKETSTREAM_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/socketstream.h" + +#endif // WEBRTC_BASE_SOCKETSTREAM_H_ diff --git a/webrtc/base/ssladapter.h b/webrtc/base/ssladapter.h new file mode 100644 index 0000000000..3d432ecd0c --- /dev/null +++ b/webrtc/base/ssladapter.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SSLADAPTER_H_ +#define WEBRTC_BASE_SSLADAPTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/ssladapter.h" + +#endif // WEBRTC_BASE_SSLADAPTER_H_ diff --git a/webrtc/base/sslfingerprint.h b/webrtc/base/sslfingerprint.h new file mode 100644 index 0000000000..6be82fd1b2 --- /dev/null +++ b/webrtc/base/sslfingerprint.h @@ -0,0 +1,19 @@ +/* + * Copyright 2012 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SSLFINGERPRINT_H_ +#define WEBRTC_BASE_SSLFINGERPRINT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sslfingerprint.h" + +#endif // WEBRTC_BASE_SSLFINGERPRINT_H_ diff --git a/webrtc/base/sslidentity.h b/webrtc/base/sslidentity.h new file mode 100644 index 0000000000..1cedfa09c1 --- /dev/null +++ b/webrtc/base/sslidentity.h @@ -0,0 +1,21 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Handling of certificates and keypairs for SSLStreamAdapter's peer mode. + +#ifndef WEBRTC_BASE_SSLIDENTITY_H_ +#define WEBRTC_BASE_SSLIDENTITY_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sslidentity.h" + +#endif // WEBRTC_BASE_SSLIDENTITY_H_ diff --git a/webrtc/base/sslroots.h b/webrtc/base/sslroots.h new file mode 100644 index 0000000000..9fa706b415 --- /dev/null +++ b/webrtc/base/sslroots.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SSLROOTS_H_ +#define WEBRTC_BASE_SSLROOTS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sslroots.h" + +#endif // WEBRTC_BASE_SSLROOTS_H_ diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h new file mode 100644 index 0000000000..d7c062e4b8 --- /dev/null +++ b/webrtc/base/sslstreamadapter.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SSLSTREAMADAPTER_H_ +#define WEBRTC_BASE_SSLSTREAMADAPTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/sslstreamadapter.h" + +#endif // WEBRTC_BASE_SSLSTREAMADAPTER_H_ diff --git a/webrtc/base/stream.h b/webrtc/base/stream.h new file mode 100644 index 0000000000..18dd865414 --- /dev/null +++ b/webrtc/base/stream.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_STREAM_H_ +#define WEBRTC_BASE_STREAM_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/stream.h" + +#endif // WEBRTC_BASE_STREAM_H_ diff --git a/webrtc/base/string_to_number.h b/webrtc/base/string_to_number.h new file mode 100644 index 0000000000..fa88ba4da3 --- /dev/null +++ b/webrtc/base/string_to_number.h @@ -0,0 +1,19 @@ +/* + * Copyright 2017 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_STRING_TO_NUMBER_H_ +#define WEBRTC_BASE_STRING_TO_NUMBER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/string_to_number.h" + +#endif // WEBRTC_BASE_STRING_TO_NUMBER_H_ diff --git a/webrtc/base/stringencode.h b/webrtc/base/stringencode.h new file mode 100644 index 0000000000..27b810ea3c --- /dev/null +++ b/webrtc/base/stringencode.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_STRINGENCODE_H_ +#define WEBRTC_BASE_STRINGENCODE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/stringencode.h" + +#endif // WEBRTC_BASE_STRINGENCODE_H__ diff --git a/webrtc/base/stringize_macros.h b/webrtc/base/stringize_macros.h new file mode 100644 index 0000000000..5f8a5b1b86 --- /dev/null +++ b/webrtc/base/stringize_macros.h @@ -0,0 +1,26 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Modified from the Chromium original: +// src/base/strings/stringize_macros.h + +// This file defines preprocessor macros for stringizing preprocessor +// symbols (or their output) and manipulating preprocessor symbols +// that define strings. + +#ifndef WEBRTC_BASE_STRINGIZE_MACROS_H_ +#define WEBRTC_BASE_STRINGIZE_MACROS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/stringize_macros.h" + +#endif // WEBRTC_BASE_STRINGIZE_MACROS_H_ diff --git a/webrtc/base/stringutils.h b/webrtc/base/stringutils.h new file mode 100644 index 0000000000..e3b5e07822 --- /dev/null +++ b/webrtc/base/stringutils.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_STRINGUTILS_H_ +#define WEBRTC_BASE_STRINGUTILS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/stringutils.h" + +#endif // WEBRTC_BASE_STRINGUTILS_H_ diff --git a/webrtc/base/swap_queue.h b/webrtc/base/swap_queue.h new file mode 100644 index 0000000000..711114748f --- /dev/null +++ b/webrtc/base/swap_queue.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_SWAP_QUEUE_H_ +#define WEBRTC_BASE_SWAP_QUEUE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/swap_queue.h" + +#endif // WEBRTC_BASE_SWAP_QUEUE_H_ diff --git a/webrtc/base/task_queue.h b/webrtc/base/task_queue.h new file mode 100644 index 0000000000..12f5cbbf9f --- /dev/null +++ b/webrtc/base/task_queue.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TASK_QUEUE_H_ +#define WEBRTC_BASE_TASK_QUEUE_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/task_queue.h" + +#endif // WEBRTC_BASE_TASK_QUEUE_H_ diff --git a/webrtc/base/task_queue_posix.h b/webrtc/base/task_queue_posix.h new file mode 100644 index 0000000000..6cb51a03c6 --- /dev/null +++ b/webrtc/base/task_queue_posix.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TASK_QUEUE_POSIX_H_ +#define WEBRTC_BASE_TASK_QUEUE_POSIX_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/task_queue_posix.h" + +#endif // WEBRTC_BASE_TASK_QUEUE_POSIX_H_ diff --git a/webrtc/base/template_util.h b/webrtc/base/template_util.h new file mode 100644 index 0000000000..9a05643ddc --- /dev/null +++ b/webrtc/base/template_util.h @@ -0,0 +1,21 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Borrowed from Chromium's src/base/template_util.h. + +#ifndef WEBRTC_BASE_TEMPLATE_UTIL_H_ +#define WEBRTC_BASE_TEMPLATE_UTIL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/template_util.h" + +#endif // WEBRTC_BASE_TEMPLATE_UTIL_H_ diff --git a/webrtc/base/testbase64.h b/webrtc/base/testbase64.h new file mode 100644 index 0000000000..fc9846f1d8 --- /dev/null +++ b/webrtc/base/testbase64.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TESTBASE64_H_ +#define WEBRTC_BASE_TESTBASE64_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/testbase64.h" + +#endif // WEBRTC_BASE_TESTBASE64_H_ diff --git a/webrtc/base/testclient.h b/webrtc/base/testclient.h new file mode 100644 index 0000000000..378e2b81d4 --- /dev/null +++ b/webrtc/base/testclient.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TESTCLIENT_H_ +#define WEBRTC_BASE_TESTCLIENT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/testclient.h" + +#endif // WEBRTC_BASE_TESTCLIENT_H_ diff --git a/webrtc/base/testechoserver.h b/webrtc/base/testechoserver.h new file mode 100644 index 0000000000..21365e2a82 --- /dev/null +++ b/webrtc/base/testechoserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TESTECHOSERVER_H_ +#define WEBRTC_BASE_TESTECHOSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/testechoserver.h" + +#endif // WEBRTC_BASE_TESTECHOSERVER_H_ diff --git a/webrtc/base/testutils.h b/webrtc/base/testutils.h new file mode 100644 index 0000000000..74f216066e --- /dev/null +++ b/webrtc/base/testutils.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TESTUTILS_H_ +#define WEBRTC_BASE_TESTUTILS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/testutils.h" + +#endif // WEBRTC_BASE_TESTUTILS_H_ diff --git a/webrtc/base/thread.h b/webrtc/base/thread.h new file mode 100644 index 0000000000..6a6887aa0c --- /dev/null +++ b/webrtc/base/thread.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_THREAD_H_ +#define WEBRTC_BASE_THREAD_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/thread.h" + +#endif // WEBRTC_BASE_THREAD_H_ diff --git a/webrtc/base/thread_annotations.h b/webrtc/base/thread_annotations.h new file mode 100644 index 0000000000..5b94ffed85 --- /dev/null +++ b/webrtc/base/thread_annotations.h @@ -0,0 +1,27 @@ +// +// Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. +// +// Use of this source code is governed by a BSD-style license +// that can be found in the LICENSE file in the root of the source +// tree. An additional intellectual property rights grant can be found +// in the file PATENTS. All contributing project authors may +// be found in the AUTHORS file in the root of the source tree. +// +// Borrowed from +// https://code.google.com/p/gperftools/source/browse/src/base/thread_annotations.h +// but adapted for clang attributes instead of the gcc. +// +// This header file contains the macro definitions for thread safety +// annotations that allow the developers to document the locking policies +// of their multi-threaded code. The annotations can also help program +// analysis tools to identify potential thread safety issues. + +#ifndef WEBRTC_BASE_THREAD_ANNOTATIONS_H_ +#define WEBRTC_BASE_THREAD_ANNOTATIONS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/thread_annotations.h" + +#endif // WEBRTC_BASE_THREAD_ANNOTATIONS_H_ diff --git a/webrtc/base/thread_checker.h b/webrtc/base/thread_checker.h new file mode 100644 index 0000000000..ade52564ec --- /dev/null +++ b/webrtc/base/thread_checker.h @@ -0,0 +1,21 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Borrowed from Chromium's src/base/threading/thread_checker.h. + +#ifndef WEBRTC_BASE_THREAD_CHECKER_H_ +#define WEBRTC_BASE_THREAD_CHECKER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/thread_checker.h" + +#endif // WEBRTC_BASE_THREAD_CHECKER_H_ diff --git a/webrtc/base/thread_checker_impl.h b/webrtc/base/thread_checker_impl.h new file mode 100644 index 0000000000..3a0a6c7315 --- /dev/null +++ b/webrtc/base/thread_checker_impl.h @@ -0,0 +1,21 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +// Borrowed from Chromium's src/base/threading/thread_checker_impl.h. + +#ifndef WEBRTC_BASE_THREAD_CHECKER_IMPL_H_ +#define WEBRTC_BASE_THREAD_CHECKER_IMPL_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/thread_checker_impl.h" + +#endif // WEBRTC_BASE_THREAD_CHECKER_IMPL_H_ diff --git a/webrtc/base/timedelta.h b/webrtc/base/timedelta.h new file mode 100644 index 0000000000..f2e98a8cc2 --- /dev/null +++ b/webrtc/base/timedelta.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TIMEDELTA_H_ +#define WEBRTC_BASE_TIMEDELTA_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/timedelta.h" + +#endif // WEBRTC_BASE_TIMEDELTA_H_ diff --git a/webrtc/base/timestampaligner.h b/webrtc/base/timestampaligner.h new file mode 100644 index 0000000000..60c36311df --- /dev/null +++ b/webrtc/base/timestampaligner.h @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TIMESTAMPALIGNER_H_ +#define WEBRTC_BASE_TIMESTAMPALIGNER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/timestampaligner.h" + +#endif // WEBRTC_BASE_TIMESTAMPALIGNER_H_ diff --git a/webrtc/base/timeutils.h b/webrtc/base/timeutils.h new file mode 100644 index 0000000000..1569b58f48 --- /dev/null +++ b/webrtc/base/timeutils.h @@ -0,0 +1,19 @@ +/* + * Copyright 2005 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TIMEUTILS_H_ +#define WEBRTC_BASE_TIMEUTILS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/timeutils.h" + +#endif // WEBRTC_BASE_TIMEUTILS_H_ diff --git a/webrtc/base/trace_event.h b/webrtc/base/trace_event.h new file mode 100644 index 0000000000..1bea5f4db8 --- /dev/null +++ b/webrtc/base/trace_event.h @@ -0,0 +1,14 @@ +// Copyright (c) 2012 The Chromium Authors. All rights reserved. +// Use of this source code is governed by a BSD-style license that can be +// found in the LICENSE file under third_party_mods/chromium or at: +// http://src.chromium.org/svn/trunk/src/LICENSE + +#ifndef WEBRTC_BASE_TRACE_EVENT_H_ +#define WEBRTC_BASE_TRACE_EVENT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/trace_event.h" + +#endif // WEBRTC_BASE_TRACE_EVENT_H_ diff --git a/webrtc/base/transformadapter.h b/webrtc/base/transformadapter.h new file mode 100644 index 0000000000..3d9c86bb26 --- /dev/null +++ b/webrtc/base/transformadapter.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TRANSFORMADAPTER_H_ +#define WEBRTC_BASE_TRANSFORMADAPTER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/transformadapter.h" + +#endif // WEBRTC_BASE_TRANSFORMADAPTER_H_ diff --git a/webrtc/base/type_traits.h b/webrtc/base/type_traits.h new file mode 100644 index 0000000000..6a4ac8d24e --- /dev/null +++ b/webrtc/base/type_traits.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_TYPE_TRAITS_H_ +#define WEBRTC_BASE_TYPE_TRAITS_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/type_traits.h" + +#endif // WEBRTC_BASE_TYPE_TRAITS_H_ diff --git a/webrtc/base/unixfilesystem.h b/webrtc/base/unixfilesystem.h new file mode 100644 index 0000000000..7a182055af --- /dev/null +++ b/webrtc/base/unixfilesystem.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_UNIXFILESYSTEM_H_ +#define WEBRTC_BASE_UNIXFILESYSTEM_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/unixfilesystem.h" + +#endif // WEBRTC_BASE_UNIXFILESYSTEM_H_ diff --git a/webrtc/base/virtualsocketserver.h b/webrtc/base/virtualsocketserver.h new file mode 100644 index 0000000000..31ce96d2e0 --- /dev/null +++ b/webrtc/base/virtualsocketserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_VIRTUALSOCKETSERVER_H_ +#define WEBRTC_BASE_VIRTUALSOCKETSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/virtualsocketserver.h" + +#endif // WEBRTC_BASE_VIRTUALSOCKETSERVER_H_ diff --git a/webrtc/base/weak_ptr.h b/webrtc/base/weak_ptr.h new file mode 100644 index 0000000000..282a551628 --- /dev/null +++ b/webrtc/base/weak_ptr.h @@ -0,0 +1,19 @@ +/* + * Copyright 2016 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_WEAK_PTR_H_ +#define WEBRTC_BASE_WEAK_PTR_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/weak_ptr.h" + +#endif // WEBRTC_BASE_WEAK_PTR_H_ diff --git a/webrtc/base/win32.h b/webrtc/base/win32.h new file mode 100644 index 0000000000..413bd11cab --- /dev/null +++ b/webrtc/base/win32.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_WIN32_H_ +#define WEBRTC_BASE_WIN32_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/win32.h" + +#endif // WEBRTC_BASE_WIN32_H_ diff --git a/webrtc/base/win32filesystem.h b/webrtc/base/win32filesystem.h new file mode 100644 index 0000000000..d647c440f0 --- /dev/null +++ b/webrtc/base/win32filesystem.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_WIN32FILESYSTEM_H_ +#define WEBRTC_BASE_WIN32FILESYSTEM_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/win32filesystem.h" + +#endif // WEBRTC_BASE_WIN32FILESYSTEM_H_ diff --git a/webrtc/base/win32socketinit.h b/webrtc/base/win32socketinit.h new file mode 100644 index 0000000000..d7017e1387 --- /dev/null +++ b/webrtc/base/win32socketinit.h @@ -0,0 +1,19 @@ +/* + * Copyright 2009 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_WIN32SOCKETINIT_H_ +#define WEBRTC_BASE_WIN32SOCKETINIT_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/win32socketinit.h" + +#endif // WEBRTC_BASE_WIN32SOCKETINIT_H_ diff --git a/webrtc/base/win32socketserver.h b/webrtc/base/win32socketserver.h new file mode 100644 index 0000000000..c14369295b --- /dev/null +++ b/webrtc/base/win32socketserver.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_WIN32SOCKETSERVER_H_ +#define WEBRTC_BASE_WIN32SOCKETSERVER_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/win32socketserver.h" + +#endif // WEBRTC_BASE_WIN32SOCKETSERVER_H_ diff --git a/webrtc/base/win32window.h b/webrtc/base/win32window.h new file mode 100644 index 0000000000..ffffdf9aa7 --- /dev/null +++ b/webrtc/base/win32window.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_WIN32WINDOW_H_ +#define WEBRTC_BASE_WIN32WINDOW_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/win32window.h" + +#endif // WEBRTC_BASE_WIN32WINDOW_H_ diff --git a/webrtc/base/window.h b/webrtc/base/window.h new file mode 100644 index 0000000000..d515f7c829 --- /dev/null +++ b/webrtc/base/window.h @@ -0,0 +1,19 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_BASE_WINDOW_H_ +#define WEBRTC_BASE_WINDOW_H_ + + +// This header is deprecated and is just left here temporarily during +// refactoring. See https://bugs.webrtc.org/7634 for more details. +#include "webrtc/rtc_base/window.h" + +#endif // WEBRTC_BASE_WINDOW_H_ diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn index 2d1853063a..6a50ea09a2 100644 --- a/webrtc/call/BUILD.gn +++ b/webrtc/call/BUILD.gn @@ -28,8 +28,8 @@ rtc_source_set("call_interfaces") { "../api:libjingle_peerconnection_api", "../api:transport_api", "../api/audio_codecs:audio_codecs_api", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", + "../base:rtc_base", + "../base:rtc_base_approved", ] } @@ -43,7 +43,7 @@ rtc_source_set("rtp_interfaces") { "rtp_transport_controller_send_interface.h", ] deps = [ - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } @@ -64,8 +64,8 @@ rtc_source_set("rtp_receiver") { deps = [ ":rtp_interfaces", "..:webrtc_common", + "../base:rtc_base_approved", "../modules/rtp_rtcp", - "../rtc_base:rtc_base_approved", ] } @@ -76,8 +76,8 @@ rtc_source_set("rtp_sender") { ] deps = [ ":rtp_interfaces", + "../base:rtc_base_approved", "../modules/congestion_controller", - "../rtc_base:rtc_base_approved", ] } @@ -109,6 +109,7 @@ rtc_static_library("call") { "..:webrtc_common", "../api:transport_api", "../audio", + "../base:rtc_task_queue", "../logging:rtc_event_log_api", "../logging:rtc_event_log_impl", "../modules/bitrate_controller", @@ -116,7 +117,6 @@ rtc_static_library("call") { "../modules/pacing", "../modules/rtp_rtcp", "../modules/utility", - "../rtc_base:rtc_task_queue", "../system_wrappers", "../video", ] @@ -149,6 +149,7 @@ if (rtc_include_tests) { ":rtp_sender", "..:webrtc_common", "../api:mock_audio_mixer", + "../base:rtc_base_approved", "../logging:rtc_event_log_api", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer", @@ -157,7 +158,6 @@ if (rtc_include_tests) { "../modules/pacing", "../modules/rtp_rtcp", "../modules/rtp_rtcp:mock_rtp_rtcp", - "../rtc_base:rtc_base_approved", "../system_wrappers", "../test:audio_codec_mocks", "../test:direct_transport", @@ -191,11 +191,11 @@ if (rtc_include_tests) { ":call_interfaces", "..:webrtc_common", "../api/audio_codecs:builtin_audio_encoder_factory", + "../base:rtc_base_approved", "../logging:rtc_event_log_api", "../modules/audio_coding", "../modules/audio_mixer:audio_mixer_impl", "../modules/rtp_rtcp", - "../rtc_base:rtc_base_approved", "../system_wrappers", "../system_wrappers:metrics_default", "../test:direct_transport", diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn index f7f3efbf36..ff0aa267fd 100644 --- a/webrtc/common_audio/BUILD.gn +++ b/webrtc/common_audio/BUILD.gn @@ -63,8 +63,8 @@ rtc_static_library("common_audio") { deps = [ ":sinc_resampler", "..:webrtc_common", - "../rtc_base:gtest_prod", - "../rtc_base:rtc_base_approved", + "../base:gtest_prod", + "../base:rtc_base_approved", "../system_wrappers", ] public_deps = [ @@ -209,8 +209,8 @@ rtc_source_set("common_audio_c") { ":common_audio_c_arm_asm", ":common_audio_cc", "..:webrtc_common", - "../rtc_base:compile_assert_c", - "../rtc_base:rtc_base_approved", + "../base:compile_assert_c", + "../base:rtc_base_approved", "../system_wrappers:system_wrappers", ] } @@ -225,7 +225,7 @@ rtc_source_set("common_audio_cc") { public_configs = [ ":common_audio_config" ] deps = [ "..:webrtc_common", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "../system_wrappers:system_wrappers", ] } @@ -236,8 +236,8 @@ rtc_source_set("sinc_resampler") { ] deps = [ "..:webrtc_common", - "../rtc_base:gtest_prod", - "../rtc_base:rtc_base_approved", + "../base:gtest_prod", + "../base:rtc_base_approved", "../system_wrappers", ] } @@ -344,7 +344,7 @@ if (rtc_build_with_neon) { } deps = [ ":common_audio_c", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } } @@ -401,8 +401,8 @@ if (rtc_include_tests) { ":common_audio", ":sinc_resampler", "..:webrtc_common", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", "../system_wrappers", "../test:test_main", "//testing/gmock", diff --git a/webrtc/common_video/BUILD.gn b/webrtc/common_video/BUILD.gn index 20953c802c..68b4934317 100644 --- a/webrtc/common_video/BUILD.gn +++ b/webrtc/common_video/BUILD.gn @@ -57,10 +57,10 @@ rtc_static_library("common_video") { deps = [ "..:webrtc_common", + "../base:rtc_base", + "../base:rtc_task_queue", "../media:rtc_h264_profile_id", "../modules:module_api", - "../rtc_base:rtc_base", - "../rtc_base:rtc_task_queue", "../system_wrappers", ] public_deps = [ @@ -114,9 +114,9 @@ if (rtc_include_tests) { deps = [ ":common_video", + "../base:rtc_base", + "../base:rtc_base_approved", "../modules/video_capture:video_capture", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", "../system_wrappers:system_wrappers", "../test:test_main", "../test:video_test_common", diff --git a/webrtc/examples/BUILD.gn b/webrtc/examples/BUILD.gn index 55b72ddf16..85813fa222 100644 --- a/webrtc/examples/BUILD.gn +++ b/webrtc/examples/BUILD.gn @@ -422,7 +422,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) { "objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm", ] deps = [ - "//webrtc/rtc_base:rtc_base", + "//webrtc/base:rtc_base", ] public_deps = [ ":AppRTCMobile_ios_frameworks", @@ -524,12 +524,12 @@ if (is_linux || is_win) { "//third_party/libyuv", "//webrtc/api:libjingle_peerconnection_test_api", "//webrtc/api:video_frame_api", + "//webrtc/base:rtc_base", + "//webrtc/base:rtc_base_approved", + "//webrtc/base:rtc_json", "//webrtc/media:rtc_media", "//webrtc/modules/video_capture:video_capture_module", "//webrtc/pc:libjingle_peerconnection", - "//webrtc/rtc_base:rtc_base", - "//webrtc/rtc_base:rtc_base_approved", - "//webrtc/rtc_base:rtc_json", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -548,7 +548,7 @@ if (is_linux || is_win) { ] deps = [ "//webrtc:webrtc_common", - "//webrtc/rtc_base:rtc_base_approved", + "//webrtc/base:rtc_base_approved", "//webrtc/rtc_tools:command_line_parser", ] if (!build_with_chromium && is_clang) { @@ -562,10 +562,10 @@ if (is_linux || is_win) { "relayserver/relayserver_main.cc", ] deps = [ - "../rtc_base:rtc_base", + "../base:rtc_base", + "//webrtc/base:rtc_base_approved", "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", - "//webrtc/rtc_base:rtc_base_approved", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -580,10 +580,10 @@ if (is_linux || is_win) { "turnserver/turnserver_main.cc", ] deps = [ - "../rtc_base:rtc_base", + "../base:rtc_base", + "//webrtc/base:rtc_base_approved", "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", - "//webrtc/rtc_base:rtc_base_approved", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -598,10 +598,10 @@ if (is_linux || is_win) { "stunserver/stunserver_main.cc", ] deps = [ - "../rtc_base:rtc_base", + "../base:rtc_base", + "//webrtc/base:rtc_base_approved", "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", - "//webrtc/rtc_base:rtc_base_approved", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -633,13 +633,13 @@ if (is_win) { deps = [ "//webrtc/api:libjingle_peerconnection_test_api", "//webrtc/api:video_frame_api", + "//webrtc/base:rtc_base", + "//webrtc/base:rtc_base_approved", + "//webrtc/base:rtc_json", "//webrtc/media:rtc_media", "//webrtc/media:rtc_media_base", "//webrtc/modules/video_capture:video_capture_module", "//webrtc/pc:libjingle_peerconnection", - "//webrtc/rtc_base:rtc_base", - "//webrtc/rtc_base:rtc_base_approved", - "//webrtc/rtc_base:rtc_json", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -661,10 +661,10 @@ if (!build_with_chromium) { } deps = [ + "../base:rtc_base", + "../base:rtc_base_approved", "../p2p:libstunprober", "../p2p:rtc_p2p", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", "../system_wrappers:field_trial_default", ] } diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn index 6a70324e5b..f3c3469807 100644 --- a/webrtc/logging/BUILD.gn +++ b/webrtc/logging/BUILD.gn @@ -30,7 +30,7 @@ rtc_source_set("rtc_event_log_api") { deps = [ "..:video_stream_api", "..:webrtc_common", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } @@ -48,11 +48,11 @@ rtc_static_library("rtc_event_log_impl") { deps = [ ":rtc_event_log_api", "..:webrtc_common", + "../base:protobuf_utils", + "../base:rtc_base_approved", "../modules/audio_coding:audio_network_adaptor", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp", - "../rtc_base:protobuf_utils", - "../rtc_base:rtc_base_approved", "../system_wrappers", ] @@ -96,8 +96,8 @@ if (rtc_enable_protobuf) { } deps = [ "..:video_stream_api", - "../rtc_base:protobuf_utils", - "../rtc_base:rtc_base_approved", + "../base:protobuf_utils", + "../base:rtc_base_approved", ] } @@ -111,12 +111,12 @@ if (rtc_enable_protobuf) { deps = [ ":rtc_event_log_impl", ":rtc_event_log_parser", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", "../call", "../modules/audio_coding:audio_network_adaptor", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", "../test:test_support", "//testing/gmock", @@ -136,8 +136,8 @@ if (rtc_enable_protobuf) { ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_parser", + "../base:rtc_base_approved", "../modules/rtp_rtcp:rtp_rtcp", - "../rtc_base:rtc_base_approved", "../system_wrappers:field_trial_default", "../system_wrappers:metrics_default", "../test:rtp_test_utils", @@ -159,7 +159,7 @@ if (rtc_enable_protobuf) { ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_parser", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", # TODO(kwiberg): Remove this dependency. "../api/audio_codecs:audio_codecs_api", @@ -182,7 +182,7 @@ if (rtc_enable_protobuf) { ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_proto", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "//third_party/gflags", ] if (!build_with_chromium && is_clang) { diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn index ef9b79c7e2..fad410b2dd 100644 --- a/webrtc/media/BUILD.gn +++ b/webrtc/media/BUILD.gn @@ -45,8 +45,8 @@ rtc_source_set("rtc_h264_profile_id") { deps = [ "..:webrtc_common", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", + "../base:rtc_base", + "../base:rtc_base_approved", ] } @@ -115,9 +115,9 @@ rtc_static_library("rtc_media_base") { ":rtc_h264_profile_id", "..:webrtc_common", "../api:libjingle_peerconnection_api", + "../base:rtc_base", + "../base:rtc_base_approved", "../p2p", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", ] if (is_nacl) { @@ -227,6 +227,10 @@ rtc_static_library("rtc_audio_video") { "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", "../api/video_codecs:video_codecs_api", + "../base:rtc_base", + "../base:rtc_base_approved", + "../base:rtc_task_queue", + "../base:sequenced_task_checker", "../call", "../common_video:common_video", "../modules/audio_coding:rent_a_codec", @@ -241,10 +245,6 @@ rtc_static_library("rtc_audio_video") { "../modules/video_coding:webrtc_vp9", "../p2p:rtc_p2p", "../pc:rtc_pc_base", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", - "../rtc_base:sequenced_task_checker", "../system_wrappers", "../video", "../voice_engine", @@ -292,9 +292,9 @@ rtc_static_library("rtc_data") { "..:webrtc_common", "../api:call_api", "../api:transport_api", + "../base:rtc_base", + "../base:rtc_base_approved", "../p2p:rtc_p2p", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", "../system_wrappers", ] } @@ -368,10 +368,10 @@ if (rtc_include_tests) { "../api:call_api", "../api:video_frame_api", "../api/video_codecs:video_codecs_api", + "../base:rtc_base", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", "../call:call_interfaces", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", "../test:test_support", "//testing/gtest", ] @@ -508,6 +508,10 @@ if (rtc_include_tests) { "../api/audio_codecs:builtin_audio_encoder_factory", "../api/video_codecs:video_codecs_api", "../audio", + "../base:rtc_base", + "../base:rtc_base_approved", + "../base:rtc_base_tests_main", + "../base:rtc_base_tests_utils", "../call:call_interfaces", "../common_video:common_video", "../logging:rtc_event_log_api", @@ -517,10 +521,6 @@ if (rtc_include_tests) { "../modules/video_coding:video_coding_utility", "../modules/video_coding:webrtc_vp8", "../p2p:p2p_test_utils", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_main", - "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", "../test:audio_codec_mocks", "../test:test_support", diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn index 0ae50414fc..258683165d 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn @@ -37,7 +37,7 @@ rtc_source_set("module_api") { deps = [ "..:webrtc_common", "../api:video_frame_api", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index f751963ca4..d5b669c0eb 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -47,7 +47,7 @@ rtc_static_library("audio_format_conversion") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } @@ -58,8 +58,8 @@ rtc_static_library("builtin_audio_decoder_factory_internal") { ] deps = [ "../..:webrtc_common", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", "../../api/audio_codecs:audio_codecs_api", ] + audio_codec_deps defines = audio_codec_defines @@ -72,8 +72,8 @@ rtc_static_library("builtin_audio_encoder_factory_internal") { ] deps = [ "../..:webrtc_common", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", "../../api/audio_codecs:audio_codecs_api", ] + audio_codec_deps defines = audio_codec_defines @@ -89,8 +89,8 @@ rtc_static_library("rent_a_codec") { deps = [ "../../api/audio_codecs:audio_codecs_api", "../..:webrtc_common", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", "../../system_wrappers", ":audio_coding_module_typedefs", ":isac_common", @@ -156,7 +156,7 @@ rtc_static_library("audio_coding") { ":audio_coding_module_typedefs", ":neteq", ":rent_a_codec", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../logging:rtc_event_log_api", ] defines = audio_coding_defines @@ -169,7 +169,7 @@ rtc_static_library("legacy_encoded_audio_frame") { ] deps = [ "../../api/audio_codecs:audio_codecs_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } @@ -193,8 +193,8 @@ rtc_static_library("cng") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", ] } @@ -212,8 +212,8 @@ rtc_static_library("red") { deps = [ "../../api/audio_codecs:audio_codecs_api", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", ] } @@ -238,7 +238,7 @@ rtc_static_library("g711") { ":legacy_encoded_audio_frame", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] public_deps = [ ":g711_c", @@ -280,7 +280,7 @@ rtc_static_library("g722") { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/g722:audio_encoder_g722_config", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] public_deps = [ ":g722_c", @@ -323,8 +323,8 @@ rtc_static_library("ilbc") { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/ilbc:audio_encoder_ilbc_config", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", ] public_deps = [ ":ilbc_c", @@ -480,8 +480,8 @@ rtc_source_set("ilbc_c") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", ] } @@ -495,7 +495,7 @@ rtc_static_library("isac_common") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } @@ -587,9 +587,9 @@ rtc_static_library("isac_c") { deps = [ ":isac_common", "../..:webrtc_common", + "../../base:compile_assert_c", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:compile_assert_c", - "../../rtc_base:rtc_base_approved", ] } @@ -697,9 +697,9 @@ rtc_source_set("isac_fix_c") { ":isac_common", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", + "../../base:compile_assert_c", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:compile_assert_c", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -773,8 +773,8 @@ if (rtc_build_with_neon) { deps = [ ":isac_fix_common", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", ] } } @@ -799,7 +799,7 @@ rtc_static_library("pcm16b") { ":legacy_encoded_audio_frame", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] public_deps = [ ":pcm16b_c", @@ -837,10 +837,10 @@ rtc_static_library("webrtc_opus") { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/opus:audio_encoder_opus_config", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", + "../../base:rtc_numerics", "../../common_audio", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_numerics", "../../system_wrappers", ] public_deps = [ @@ -876,7 +876,7 @@ rtc_source_set("webrtc_opus_c") { deps = [ "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } @@ -926,10 +926,10 @@ rtc_static_library("audio_network_adaptor") { deps = [ "../..:webrtc_common", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", "../../common_audio", "../../logging:rtc_event_log_api", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -953,7 +953,7 @@ rtc_source_set("neteq_decoder_enum") { ] deps = [ "../../api/audio_codecs:audio_codecs_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } @@ -1042,9 +1042,9 @@ rtc_static_library("neteq") { "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", + "../../base:gtest_prod", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:gtest_prod", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -1102,7 +1102,7 @@ rtc_source_set("neteq_tools_minimal") { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../rtp_rtcp", ] } @@ -1134,9 +1134,9 @@ rtc_source_set("neteq_test_tools") { ":pcm16b", "..:module_api", "../..:webrtc_common", + "../../base:rtc_base_approved", + "../../base:rtc_base_tests_utils", "../../common_audio", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_base_tests_utils", "../../test:rtp_test_utils", "../rtp_rtcp", ] @@ -1183,8 +1183,8 @@ rtc_source_set("neteq_tools") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", "../rtp_rtcp", ] @@ -1212,8 +1212,8 @@ if (rtc_enable_protobuf) { } deps = [ + "../../base:rtc_base_approved", "../../logging:rtc_event_log_parser", - "../../rtc_base:rtc_base_approved", ] public_deps = [ "../../logging:rtc_event_log_proto", @@ -1307,7 +1307,7 @@ if (rtc_include_tests) { "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", ] @@ -1342,8 +1342,8 @@ if (rtc_include_tests) { ":neteq_test_tools", ":webrtc_opus", "../..:webrtc_common", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", ] @@ -1369,7 +1369,7 @@ if (rtc_include_tests) { "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", ":neteq_tools", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_support", "//testing/gtest", ] @@ -1388,7 +1388,7 @@ if (rtc_include_tests) { ":audio_coding", ":neteq_tools", "../../api/audio_codecs:audio_codecs_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_support", "//testing/gtest", ] @@ -1412,7 +1412,7 @@ if (rtc_include_tests) { ":audio_format_conversion", "..:module_api", "../../:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", "../../system_wrappers:system_wrappers_default", "../../test:test_support", @@ -1442,7 +1442,7 @@ if (rtc_include_tests) { ":audio_format_conversion", "..:module_api", "../../:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", "../../system_wrappers:system_wrappers_default", "../../test:test_support", @@ -1489,8 +1489,8 @@ if (rtc_include_tests) { ":neteq_tools", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/opus:audio_encoder_opus", + "../../base:protobuf_utils", "../../common_audio", - "../../rtc_base:protobuf_utils", "../../test:test_main", "//testing/gtest", ] @@ -1540,7 +1540,7 @@ if (rtc_include_tests) { ":neteq", ":neteq_test_tools", "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "../../test:test_support", "//third_party/gflags", @@ -1573,7 +1573,7 @@ if (rtc_include_tests) { ":isac_fix", ":webrtc_opus", "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "../../test:test_main", "../audio_processing", @@ -1603,7 +1603,7 @@ if (rtc_include_tests) { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", "../../test:test_support", "//testing/gtest", @@ -1628,7 +1628,7 @@ if (rtc_include_tests) { "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_support", "//testing/gtest", "//third_party/gflags", @@ -1705,8 +1705,8 @@ if (rtc_include_tests) { ":pcm16b", ":webrtc_opus", "../..:webrtc_common", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", ] configs += [ ":RTPencode_config" ] @@ -1749,7 +1749,7 @@ if (rtc_include_tests) { ] deps = [ - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "../../test:rtp_test_utils", "//testing/gtest", @@ -1774,7 +1774,7 @@ if (rtc_include_tests) { testonly = true deps = [ "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_support", "//testing/gtest", ] @@ -1853,7 +1853,7 @@ if (rtc_include_tests) { ":neteq_quality_test_support", ":neteq_tools", "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "../../test:test_main", "//testing/gtest", @@ -1872,7 +1872,7 @@ if (rtc_include_tests) { ":isac_fix", ":neteq", ":neteq_quality_test_support", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_main", "//testing/gtest", "//third_party/gflags", @@ -1890,7 +1890,7 @@ if (rtc_include_tests) { ":g711", ":neteq", ":neteq_quality_test_support", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_main", "//testing/gtest", "//third_party/gflags", @@ -1950,7 +1950,7 @@ if (rtc_include_tests) { deps = [ ":isac", ":isac_test_util", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] configs += [ ":isac_test_warnings_config" ] @@ -1991,7 +1991,7 @@ if (rtc_include_tests) { deps = [ ":isac", ":isac_test_util", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] include_dirs = [ @@ -2042,8 +2042,8 @@ if (rtc_include_tests) { deps = [ ":webrtc_opus", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", "../../test:test_main", "//testing/gtest", ] @@ -2167,11 +2167,11 @@ if (rtc_include_tests) { "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", "../../api/audio_codecs:builtin_audio_encoder_factory", + "../../base:protobuf_utils", + "../../base:rtc_base", + "../../base:rtc_base_approved", + "../../base:rtc_base_tests_utils", "../../common_audio", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_base_tests_utils", "../../system_wrappers:system_wrappers", "../../test:audio_codec_mocks", "../../test:field_trial", diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn index 56b20193d0..8939da222e 100644 --- a/webrtc/modules/audio_conference_mixer/BUILD.gn +++ b/webrtc/modules/audio_conference_mixer/BUILD.gn @@ -42,7 +42,7 @@ rtc_static_library("audio_conference_mixer") { "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", "../audio_processing", ] diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn index a60087efd6..c565165192 100644 --- a/webrtc/modules/audio_device/BUILD.gn +++ b/webrtc/modules/audio_device/BUILD.gn @@ -51,9 +51,9 @@ rtc_static_library("audio_device") { deps = [ "..:module_api", "../..:webrtc_common", + "../../base:rtc_base_approved", + "../../base:rtc_task_queue", "../../common_audio", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_task_queue", "../../system_wrappers", "../utility", ] @@ -177,8 +177,8 @@ rtc_static_library("audio_device") { } if (is_ios) { public_deps = [ - "../../rtc_base:gtest_prod", - "../../rtc_base:rtc_base", + "../../base:gtest_prod", + "../../base:rtc_base", "../../sdk:objc_audio", "../../sdk:objc_common", ] @@ -281,7 +281,7 @@ if (rtc_include_tests) { deps = [ ":audio_device", ":mock_audio_device", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", "../utility:utility", @@ -331,7 +331,7 @@ if (rtc_include_tests) { deps = [ ":audio_device", "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", "../../test:test_main", "../../test:test_support", diff --git a/webrtc/modules/audio_mixer/BUILD.gn b/webrtc/modules/audio_mixer/BUILD.gn index 0410e1f0e9..cd3b768ee1 100644 --- a/webrtc/modules/audio_mixer/BUILD.gn +++ b/webrtc/modules/audio_mixer/BUILD.gn @@ -41,7 +41,7 @@ rtc_static_library("audio_mixer_impl") { "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", "../audio_processing", ] @@ -61,7 +61,7 @@ rtc_static_library("audio_frame_manipulator") { deps = [ "..:module_api", "../../audio/utility", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } @@ -90,8 +90,8 @@ if (rtc_include_tests) { "..:module_api", "../../api:audio_mixer_api", "../../audio/utility:audio_frame_operations", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base", + "../../base:rtc_base_approved", "../../test:test_support", "//testing/gmock", ] diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn index fe83596277..3af019e49e 100644 --- a/webrtc/modules/audio_processing/BUILD.gn +++ b/webrtc/modules/audio_processing/BUILD.gn @@ -238,8 +238,8 @@ rtc_static_library("audio_processing") { "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", - "../../rtc_base:gtest_prod", - "../../rtc_base:protobuf_utils", + "../../base:gtest_prod", + "../../base:protobuf_utils", "../audio_coding:isac", ] public_deps = [ @@ -303,8 +303,8 @@ rtc_static_library("audio_processing") { configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] deps += [ + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] } @@ -316,7 +316,7 @@ rtc_source_set("aec_dump_interface") { ] deps = [ - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } @@ -356,8 +356,8 @@ rtc_source_set("audio_processing_c") { deps = [ "../..:webrtc_common", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -470,7 +470,7 @@ if (rtc_build_with_neon) { ] } deps = [ - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] } } @@ -550,11 +550,11 @@ if (rtc_include_tests) { ":audioproc_test_utils", "..:module_api", "../..:webrtc_common", + "../../base:gtest_prod", + "../../base:protobuf_utils", + "../../base:rtc_base", + "../../base:rtc_base_approved", "../../common_audio:common_audio", - "../../rtc_base:gtest_prod", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", "../audio_coding:neteq_tools", @@ -594,7 +594,7 @@ if (rtc_include_tests) { ":audioproc_debug_proto", ":audioproc_protobuf_utils", ":audioproc_unittest_proto", - "../../rtc_base:rtc_task_queue", + "../../base:rtc_task_queue", "aec_dump", "aec_dump:aec_dump_unittests", ] @@ -696,7 +696,7 @@ if (rtc_include_tests) { deps = [ ":audio_processing", ":audioproc_test_utils", - "../../rtc_base:protobuf_utils", + "../../base:protobuf_utils", "//testing/gtest", ] @@ -720,9 +720,9 @@ if (rtc_include_tests) { ":audioproc_protobuf_utils", ":audioproc_test_utils", "../..:webrtc_common", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "//third_party/gflags:gflags", ] @@ -745,10 +745,10 @@ if (rtc_include_tests) { ":audioproc_debug_proto", ":audioproc_protobuf_utils", ":audioproc_test_utils", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", + "../../base:rtc_task_queue", "../../common_audio:common_audio", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_task_queue", "../../system_wrappers", "../../system_wrappers:system_wrappers_default", "../../test:test_support", @@ -776,8 +776,8 @@ if (rtc_include_tests) { deps = [ ":audio_processing", "..:module_api", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", ] } @@ -825,8 +825,8 @@ if (rtc_include_tests) { deps = [ ":audio_processing", ":audioproc_test_utils", + "../../base:rtc_base_approved", "../../common_audio:common_audio", - "../../rtc_base:rtc_base_approved", "../../system_wrappers:metrics_default", "//third_party/gflags", ] @@ -866,8 +866,8 @@ if (rtc_include_tests) { deps = [ ":audioproc_debug_proto", "../..:webrtc_common", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_base_approved", + "../../base:protobuf_utils", + "../../base:rtc_base_approved", ] } } diff --git a/webrtc/modules/audio_processing/aec_dump/BUILD.gn b/webrtc/modules/audio_processing/aec_dump/BUILD.gn index 818a9bf27d..950dd68b17 100644 --- a/webrtc/modules/audio_processing/aec_dump/BUILD.gn +++ b/webrtc/modules/audio_processing/aec_dump/BUILD.gn @@ -18,7 +18,7 @@ rtc_source_set("aec_dump") { ] deps = [ - "../../../rtc_base:rtc_base_approved", + "../../../base:rtc_base_approved", ] } @@ -49,7 +49,7 @@ rtc_source_set("mock_aec_dump_unittests") { deps = [ ":mock_aec_dump", "..:audio_processing", - "../../../rtc_base:rtc_base_approved", + "../../../base:rtc_base_approved", "//testing/gtest", ] } @@ -73,10 +73,10 @@ if (rtc_enable_protobuf) { ] deps = [ + "../../../base:protobuf_utils", + "../../../base:rtc_base_approved", + "../../../base:rtc_task_queue", "../../../modules:module_api", - "../../../rtc_base:protobuf_utils", - "../../../rtc_base:rtc_base_approved", - "../../../rtc_base:rtc_task_queue", "../../../system_wrappers", ] @@ -90,8 +90,8 @@ if (rtc_enable_protobuf) { ":aec_dump_impl", "..:aec_dump_interface", "..:audioproc_debug_proto", + "../../../base:rtc_task_queue", "../../../modules:module_api", - "../../../rtc_base:rtc_task_queue", "../../../test:test_support", "//testing/gtest", ] diff --git a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn index 587663b7fc..af24f8ab5e 100644 --- a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn +++ b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn @@ -22,8 +22,8 @@ rtc_executable("conversational_speech_generator") { ] deps = [ ":lib", - "../../../../rtc_base:rtc_base_approved", - "../../../../test:test_support", + "../../../../../webrtc/base:rtc_base_approved", + "../../../../../webrtc/test:test_support", "//third_party/gflags", ] } @@ -45,9 +45,9 @@ rtc_static_library("lib") { "wavreader_interface.h", ] deps = [ - "../../../..:webrtc_common", - "../../../../common_audio", - "../../../../rtc_base:rtc_base_approved", + "../../../../../webrtc:webrtc_common", + "../../../../../webrtc/base:rtc_base_approved", + "../../../../../webrtc/common_audio", ] visibility = [ ":*" ] # Only targets in this file can depend on this. } @@ -63,11 +63,14 @@ rtc_source_set("unittest") { ] deps = [ ":lib", - "../../../..:webrtc_common", - "../../../../common_audio", - "../../../../rtc_base:rtc_base_approved", - "../../../../test:test_support", + "../../../../../webrtc:webrtc_common", + "../../../../../webrtc/base:rtc_base_approved", + "../../../../../webrtc/common_audio", + "../../../../../webrtc/test:test_support", "//testing/gmock", "//testing/gtest", + "//webrtc:webrtc_common", + "//webrtc/base:rtc_base_approved", + "//webrtc/test:test_support", ] } diff --git a/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn b/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn index 74d5eee877..154219bf53 100644 --- a/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn +++ b/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn @@ -105,7 +105,7 @@ rtc_executable("fake_polqa") { output_name = "py_quality_assessment/quality_assessment/fake_polqa" deps = [ "//webrtc:webrtc_common", - "//webrtc/rtc_base:rtc_base_approved", + "//webrtc/base:rtc_base_approved", ] } diff --git a/webrtc/modules/bitrate_controller/BUILD.gn b/webrtc/modules/bitrate_controller/BUILD.gn index f31025bd05..33a28869ba 100644 --- a/webrtc/modules/bitrate_controller/BUILD.gn +++ b/webrtc/modules/bitrate_controller/BUILD.gn @@ -37,7 +37,7 @@ rtc_static_library("bitrate_controller") { } deps = [ - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", "../rtp_rtcp", ] diff --git a/webrtc/modules/congestion_controller/BUILD.gn b/webrtc/modules/congestion_controller/BUILD.gn index 02123724f9..ef45297031 100644 --- a/webrtc/modules/congestion_controller/BUILD.gn +++ b/webrtc/modules/congestion_controller/BUILD.gn @@ -49,10 +49,10 @@ rtc_static_library("congestion_controller") { deps = [ "..:module_api", "../..:webrtc_common", + "../../base:rtc_base", + "../../base:rtc_base_approved", + "../../base:rtc_numerics", "../../logging:rtc_event_log_api", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_numerics", "../../system_wrappers", "../bitrate_controller", "../pacing", @@ -88,8 +88,8 @@ if (rtc_include_tests) { deps = [ ":congestion_controller", ":mock_congestion_controller", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:field_trial", "../../test:test_support", diff --git a/webrtc/modules/desktop_capture/BUILD.gn b/webrtc/modules/desktop_capture/BUILD.gn index 47b186d835..93ceb752c2 100644 --- a/webrtc/modules/desktop_capture/BUILD.gn +++ b/webrtc/modules/desktop_capture/BUILD.gn @@ -28,7 +28,7 @@ rtc_static_library("primitives") { deps = [ "../..:webrtc_common", - "../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806. + "../../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806. ] } @@ -49,8 +49,8 @@ if (rtc_include_tests) { ":desktop_capture_mock", ":primitives", ":screen_drawer", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base", + "../../base:rtc_base_approved", "../../system_wrappers", "../../test:test_support", "../../test:video_test_support", @@ -94,7 +94,7 @@ if (rtc_include_tests) { ":desktop_capture_mock", ":primitives", "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", "../../test:test_support", "//testing/gmock", @@ -131,7 +131,7 @@ if (rtc_include_tests) { deps = [ ":primitives", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", ] } @@ -155,7 +155,7 @@ if (rtc_include_tests) { deps = [ ":primitives", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_support", ] } @@ -290,7 +290,7 @@ rtc_static_library("desktop_capture") { deps = [ ":primitives", "../..:webrtc_common", - "../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806. + "../../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806. "../../system_wrappers", "//third_party/libyuv", ] diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn index 989305c704..7ab897f28f 100644 --- a/webrtc/modules/media_file/BUILD.gn +++ b/webrtc/modules/media_file/BUILD.gn @@ -35,8 +35,8 @@ rtc_static_library("media_file") { deps = [ "..:module_api", "../..:webrtc_common", + "../../base:rtc_base_approved", "../../common_audio", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] } diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn index 3d1d495fa2..57126d7143 100644 --- a/webrtc/modules/pacing/BUILD.gn +++ b/webrtc/modules/pacing/BUILD.gn @@ -28,8 +28,8 @@ rtc_static_library("pacing") { deps = [ "..:module_api", "../../:webrtc_common", + "../../base:rtc_base_approved", "../../logging:rtc_event_log_api", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../remote_bitrate_estimator", "../rtp_rtcp", @@ -55,8 +55,8 @@ if (rtc_include_tests) { ] deps = [ ":pacing", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_base_tests_utils", + "../../base:rtc_base_approved", + "../../base:rtc_base_tests_utils", "../../system_wrappers:system_wrappers", "../../test:test_support", "../rtp_rtcp", diff --git a/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/webrtc/modules/remote_bitrate_estimator/BUILD.gn index 1ecf630b1d..5a3afc65fd 100644 --- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn +++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn @@ -51,8 +51,8 @@ rtc_static_library("remote_bitrate_estimator") { deps = [ "../..:webrtc_common", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base", + "../../base:rtc_base_approved", "../../system_wrappers", ] } @@ -117,9 +117,9 @@ if (rtc_include_tests) { ":remote_bitrate_estimator", "..:module_api", "../..:webrtc_common", - "../../rtc_base:gtest_prod", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", + "../../base:gtest_prod", + "../../base:rtc_base", + "../../base:rtc_base_approved", "../../system_wrappers", "../../test:test_support", "../../voice_engine", @@ -147,7 +147,7 @@ if (rtc_include_tests) { deps = [ ":bwe_simulator_lib", ":remote_bitrate_estimator", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_support", ] if (!build_with_chromium && is_clang) { @@ -185,8 +185,8 @@ if (rtc_include_tests) { ":mock_remote_bitrate_observer", ":remote_bitrate_estimator", "../..:webrtc_common", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base", + "../../base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:field_trial", "../../test:test_support", @@ -227,7 +227,7 @@ if (rtc_include_tests) { ":bwe_simulator_lib", ":remote_bitrate_estimator", "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_main", "//testing/gmock", "//testing/gtest", diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 3b0bc35b2b..dc623cefb4 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -170,11 +170,11 @@ rtc_static_library("rtp_rtcp") { "../../api:libjingle_peerconnection_api", "../../api:transport_api", "../../api/audio_codecs:audio_codecs_api", + "../../base:gtest_prod", + "../../base:rtc_base_approved", + "../../base:sequenced_task_checker", "../../common_video", "../../logging:rtc_event_log_api", - "../../rtc_base:gtest_prod", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:sequenced_task_checker", "../../system_wrappers", "../audio_coding:audio_format_conversion", "../remote_bitrate_estimator", @@ -200,7 +200,7 @@ rtc_source_set("fec_test_helper") { deps = [ ":rtp_rtcp", "..:module_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", ] # TODO(jschuh): bugs.webrtc.org/1348: fix this warning. @@ -221,7 +221,7 @@ rtc_source_set("mock_rtp_rtcp") { deps = [ ":rtp_rtcp", "..:module_api", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_support", ] } @@ -256,7 +256,7 @@ if (rtc_include_tests) { ] deps = [ ":rtp_rtcp", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../test:test_support", ] if (!build_with_chromium && is_clang) { @@ -342,8 +342,8 @@ if (rtc_include_tests) { "..:module_api", "../..:webrtc_common", "../../api:transport_api", + "../../base:rtc_base_approved", "../../common_video:common_video", - "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:field_trial", "../../test:rtp_test_utils", diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn index e98b30dc53..71238905dc 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn @@ -33,8 +33,8 @@ rtc_static_library("utility") { "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", + "../../base:rtc_task_queue", "../../common_audio", - "../../rtc_base:rtc_task_queue", "../../system_wrappers", "../media_file", ] @@ -56,7 +56,7 @@ if (rtc_include_tests) { deps = [ ":utility", "..:module_api", - "../../rtc_base:rtc_task_queue", + "../../base:rtc_task_queue", "../../test:test_support", "//testing/gmock", ] diff --git a/webrtc/modules/video_capture/BUILD.gn b/webrtc/modules/video_capture/BUILD.gn index b150dffe0b..5865688e20 100644 --- a/webrtc/modules/video_capture/BUILD.gn +++ b/webrtc/modules/video_capture/BUILD.gn @@ -28,8 +28,8 @@ rtc_static_library("video_capture_module") { deps = [ "..:module_api", "../..:webrtc_common", + "../../base:rtc_base_approved", "../../common_video", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -47,7 +47,7 @@ rtc_static_library("video_capture") { deps = [ ":video_capture_module", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", ] @@ -91,7 +91,7 @@ if (!build_with_chromium) { deps = [ ":video_capture_module", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", ] @@ -175,8 +175,8 @@ if (!build_with_chromium) { deps = [ ":video_capture_internal_impl", ":video_capture_module", + "../../base:rtc_base_approved", "../../common_video:common_video", - "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../system_wrappers:system_wrappers_default", "../../test:video_test_common", diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn index 94cb1334a8..67c8ccd239 100644 --- a/webrtc/modules/video_coding/BUILD.gn +++ b/webrtc/modules/video_coding/BUILD.gn @@ -97,12 +97,12 @@ rtc_static_library("video_coding") { "..:module_api", "../..:video_stream_api", "../..:webrtc_common", + "../../base:rtc_base", + "../../base:rtc_base_approved", + "../../base:rtc_numerics", + "../../base:rtc_task_queue", + "../../base:sequenced_task_checker", "../../common_video", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_numerics", - "../../rtc_base:rtc_task_queue", - "../../rtc_base:sequenced_task_checker", "../../system_wrappers", "../rtp_rtcp:rtp_rtcp", "../utility:utility", @@ -136,12 +136,12 @@ rtc_static_library("video_coding_utility") { "..:module_api", "../..:webrtc_common", "../../api/video_codecs:video_codecs_api", + "../../base:rtc_base_approved", + "../../base:rtc_numerics", + "../../base:rtc_task_queue", + "../../base:sequenced_task_checker", "../../common_video", "../../modules/rtp_rtcp:rtp_rtcp", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_numerics", - "../../rtc_base:rtc_task_queue", - "../../rtc_base:sequenced_task_checker", "../../system_wrappers", ] } @@ -160,8 +160,8 @@ rtc_static_library("webrtc_h264") { defines = [] deps = [ ":video_coding_utility", + "../../base:rtc_base_approved", "../../media:rtc_media_base", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -198,8 +198,8 @@ rtc_static_library("webrtc_i420") { deps = [ "../..:webrtc_common", + "../../base:rtc_base_approved", "../../common_video:common_video", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] } @@ -232,9 +232,9 @@ rtc_static_library("webrtc_vp8") { "..:module_api", "../..:webrtc_common", "../../api/video_codecs:video_codecs_api", + "../../base:rtc_base_approved", + "../../base:sequenced_task_checker", "../../common_video", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:sequenced_task_checker", "../../system_wrappers", ] if (rtc_build_libvpx) { @@ -267,8 +267,8 @@ rtc_static_library("webrtc_vp9") { deps = [ ":video_coding_utility", "..:module_api", + "../../base:rtc_base_approved", "../../common_video", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] if (rtc_build_libvpx) { @@ -292,8 +292,8 @@ if (rtc_include_tests) { ":video_coding", ":webrtc_vp8", "../../api:video_frame_api", + "../../base:rtc_base_approved", "../../common_video:common_video", - "../../rtc_base:rtc_base_approved", "../../test:test_support", ] } @@ -315,7 +315,7 @@ if (rtc_include_tests) { ":video_coding", ":webrtc_vp8", "../..:webrtc_common", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers:field_trial_default", "../../system_wrappers:metrics_default", "../../system_wrappers:system_wrappers", @@ -354,8 +354,8 @@ if (rtc_include_tests) { ":webrtc_vp8", "../..:webrtc_common", "../../api/video_codecs:video_codecs_api", + "../../base:rtc_base_approved", "../../common_video:common_video", - "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", "../../test:video_test_common", @@ -378,8 +378,8 @@ if (rtc_include_tests) { ":webrtc_vp8", ":webrtc_vp9", "../..:webrtc_common", + "../../base:rtc_base_approved", "../../media:rtc_media", - "../../rtc_base:rtc_base_approved", "../../test:test_support", "../../test:video_test_support", ] @@ -391,7 +391,7 @@ if (rtc_include_tests) { ] deps += [ - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../sdk/android:libjingle_peerconnection_jni", "//base", ] @@ -428,8 +428,8 @@ if (rtc_include_tests) { ":webrtc_vp8", ":webrtc_vp9", "../../api:video_frame_api", + "../../base:rtc_base_approved", "../../common_video:common_video", - "../../rtc_base:rtc_base_approved", "../../test:test_support", "../../test:video_test_common", "../video_capture", @@ -483,7 +483,7 @@ if (rtc_include_tests) { if (is_android) { deps += [ - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", # TODO(brandtr): Figure out if the java dep below could be moved into # :video_coding_videoprocessor_integration_test, where it belongs. @@ -575,10 +575,10 @@ if (rtc_include_tests) { "../..:webrtc_common", "../../api:video_frame_api", "../../api/video_codecs:video_codecs_api", + "../../base:rtc_base", + "../../base:rtc_base_approved", + "../../base:rtc_task_queue", "../../common_video:common_video", - "../../rtc_base:rtc_base", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_task_queue", "../../system_wrappers:metrics_default", "../../system_wrappers:system_wrappers", "../../test:field_trial", diff --git a/webrtc/modules/video_processing/BUILD.gn b/webrtc/modules/video_processing/BUILD.gn index 6afc5f7f35..c4c9c3b894 100644 --- a/webrtc/modules/video_processing/BUILD.gn +++ b/webrtc/modules/video_processing/BUILD.gn @@ -27,10 +27,10 @@ rtc_static_library("video_processing") { deps = [ ":denoiser_filter", "..:module_api", + "../../base:rtc_base_approved", "../../common_audio", "../../common_video", "../../modules/utility", - "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] if (build_video_processing_sse2) { @@ -66,7 +66,7 @@ if (build_video_processing_sse2) { deps = [ ":denoiser_filter", - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers", ] diff --git a/webrtc/ortc/BUILD.gn b/webrtc/ortc/BUILD.gn index 40c0b2a22b..b6a2cc9459 100644 --- a/webrtc/ortc/BUILD.gn +++ b/webrtc/ortc/BUILD.gn @@ -35,6 +35,8 @@ rtc_static_library("ortc") { deps = [ "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", + "../base:rtc_base", + "../base:rtc_base_approved", "../call:call_interfaces", "../logging:rtc_event_log_api", "../media:rtc_media", @@ -43,8 +45,6 @@ rtc_static_library("ortc") { "../p2p:rtc_p2p", "../pc:libjingle_peerconnection", "../pc:rtc_pc", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", ] public_deps = [ @@ -76,14 +76,14 @@ if (rtc_include_tests) { deps = [ ":ortc", + "../base:rtc_base", + "../base:rtc_base_approved", + "../base:rtc_base_tests_main", + "../base:rtc_base_tests_utils", "../media:rtc_media_tests_utils", "../p2p:p2p_test_utils", "../p2p:rtc_p2p", "../pc:pc_test_utils", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_main", - "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", ] diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn index 5963c925b5..f7d59058ba 100644 --- a/webrtc/p2p/BUILD.gn +++ b/webrtc/p2p/BUILD.gn @@ -87,7 +87,7 @@ rtc_static_library("rtc_p2p") { deps = [ "../api:libjingle_peerconnection_api", "../api:ortc_api", - "../rtc_base:rtc_base", + "../base:rtc_base", "../system_wrappers:field_trial_api", ] @@ -155,9 +155,9 @@ if (rtc_include_tests) { deps = [ ":rtc_p2p", "../api:ortc_api", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", + "../base:rtc_base", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", "../test:test_support", "//testing/gmock", ] @@ -209,9 +209,9 @@ if (rtc_include_tests) { ":rtc_p2p", "../api:fakemetricsobserver", "../api:ortc_api", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", + "../base:rtc_base", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", "../test:test_support", "//testing/gmock", "//testing/gtest", @@ -238,7 +238,7 @@ rtc_static_library("libstunprober") { deps = [ ":rtc_p2p", "..:webrtc_common", - "../rtc_base:rtc_base", + "../base:rtc_base", ] } @@ -259,8 +259,8 @@ if (rtc_include_tests) { ":libstunprober", ":p2p_test_utils", ":rtc_p2p", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_tests_utils", + "../base:rtc_base", + "../base:rtc_base_tests_utils", "//testing/gmock", "//testing/gtest", ] diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn index 768a25cf41..2ff1a0af50 100644 --- a/webrtc/pc/BUILD.gn +++ b/webrtc/pc/BUILD.gn @@ -60,12 +60,12 @@ rtc_static_library("rtc_pc_base") { "../api:call_api", "../api:libjingle_peerconnection_api", "../api:ortc_api", + "../base:rtc_base", + "../base:rtc_task_queue", "../media:rtc_data", "../media:rtc_h264_profile_id", "../media:rtc_media_base", "../p2p:rtc_p2p", - "../rtc_base:rtc_base", - "../rtc_base:rtc_task_queue", ] if (rtc_build_libsrtp) { @@ -165,13 +165,13 @@ rtc_static_library("peerconnection") { "../api:call_api", "../api:rtc_stats_api", "../api/video_codecs:video_codecs_api", + "../base:rtc_base", + "../base:rtc_base_approved", "../call:call_interfaces", "../logging:rtc_event_log_api", "../media:rtc_data", "../media:rtc_media_base", "../p2p:rtc_p2p", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", "../stats", "../system_wrappers:system_wrappers", ] @@ -198,14 +198,14 @@ rtc_static_library("create_pc_factory") { "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", + "../base:rtc_base", + "../base:rtc_base_approved", "../call", "../call:call_interfaces", "../logging:rtc_event_log_api", "../media:rtc_audio_video", "../modules/audio_device:audio_device", "../modules/audio_processing:audio_processing", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", ] configs += [ ":libjingle_peerconnection_warnings_config" ] @@ -279,15 +279,15 @@ if (rtc_include_tests) { deps = [ ":libjingle_peerconnection", ":rtc_pc", + "../base:rtc_base", + "../base:rtc_base_approved", + "../base:rtc_base_tests_main", + "../base:rtc_base_tests_utils", "../logging:rtc_event_log_api", "../media:rtc_media_base", "../media:rtc_media_tests_utils", "../p2p:p2p_test_utils", "../p2p:rtc_p2p", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_main", - "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", ] @@ -325,15 +325,15 @@ if (rtc_include_tests) { "..:webrtc_common", "../api:libjingle_peerconnection_test_api", "../api:rtc_stats_api", + "../base:rtc_base", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", "../call:call_interfaces", "../logging:rtc_event_log_api", "../media:rtc_media", "../media:rtc_media_tests_utils", "../modules/audio_device:audio_device", "../p2p:p2p_test_utils", - "../rtc_base:rtc_base", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", "../test:test_support", "//testing/gmock", ] @@ -442,10 +442,10 @@ if (rtc_include_tests) { ":pc_test_utils", "..:webrtc_common", "../api:fakemetricsobserver", + "../base:rtc_base_tests_main", + "../base:rtc_base_tests_utils", "../media:rtc_media_tests_utils", "../pc:rtc_pc", - "../rtc_base:rtc_base_tests_main", - "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", "../test:audio_codec_mocks", "//testing/gmock", diff --git a/webrtc/rtc_base/BUILD.gn b/webrtc/rtc_base/BUILD.gn index 98accaf131..9e905820d2 100644 --- a/webrtc/rtc_base/BUILD.gn +++ b/webrtc/rtc_base/BUILD.gn @@ -203,8 +203,8 @@ rtc_static_library("rtc_base_approved") { # Dependency on chromium's logging (in //base). deps += [ "//base:base" ] sources += [ - "../../webrtc_overrides/webrtc/rtc_base/logging.cc", - "../../webrtc_overrides/webrtc/rtc_base/logging.h", + "../../webrtc_overrides/webrtc/base/logging.cc", + "../../webrtc_overrides/webrtc/base/logging.h", ] } else { sources += [ @@ -301,8 +301,8 @@ rtc_static_library("rtc_task_queue") { if (build_with_chromium) { sources = [ - "../../webrtc_overrides/webrtc/rtc_base/task_queue.cc", - "../../webrtc_overrides/webrtc/rtc_base/task_queue.h", + "../../webrtc_overrides/webrtc/base/task_queue.cc", + "../../webrtc_overrides/webrtc/base/task_queue.h", ] } else { sources = [ @@ -517,7 +517,7 @@ rtc_static_library("rtc_base") { if (build_with_chromium) { if (is_win) { - sources += [ "../../webrtc_overrides/webrtc/rtc_base/win32socketinit.cc" ] + sources += [ "../../webrtc_overrides/webrtc/base/win32socketinit.cc" ] } include_dirs = [ "../../boringssl/src/include" ] public_configs += [ ":rtc_base_chromium_config" ] diff --git a/webrtc/rtc_base/callback.h.pump b/webrtc/rtc_base/callback.h.pump index cceddf7343..23899526b9 100644 --- a/webrtc/rtc_base/callback.h.pump +++ b/webrtc/rtc_base/callback.h.pump @@ -57,8 +57,8 @@ #ifndef WEBRTC_RTC_BASE_CALLBACK_H_ #define WEBRTC_RTC_BASE_CALLBACK_H_ -#include "webrtc/rtc_base/refcount.h" -#include "webrtc/rtc_base/scoped_ref_ptr.h" +#include "webrtc/base/refcount.h" +#include "webrtc/base/scoped_ref_ptr.h" namespace rtc { diff --git a/webrtc/rtc_base/sigslottester.h.pump b/webrtc/rtc_base/sigslottester.h.pump index 381b7914fd..a88f0c6616 100755 --- a/webrtc/rtc_base/sigslottester.h.pump +++ b/webrtc/rtc_base/sigslottester.h.pump @@ -35,8 +35,8 @@ // EXPECT_EQ("hello", capture); // /* See unit-tests for more examples */ -#include "webrtc/rtc_base/constructormagic.h" -#include "webrtc/rtc_base/sigslot.h" +#include "webrtc/base/constructormagic.h" +#include "webrtc/base/sigslot.h" namespace rtc { diff --git a/webrtc/rtc_tools/BUILD.gn b/webrtc/rtc_tools/BUILD.gn index 21a4aa82b0..e224380cc7 100644 --- a/webrtc/rtc_tools/BUILD.gn +++ b/webrtc/rtc_tools/BUILD.gn @@ -48,8 +48,8 @@ rtc_static_library("command_line_parser") { "simple_command_line_parser.h", ] deps = [ - "../rtc_base:gtest_prod", - "../rtc_base:rtc_base_approved", + "../base:gtest_prod", + "../base:rtc_base_approved", ] } @@ -206,13 +206,13 @@ if (rtc_enable_protobuf) { defines = [ "ENABLE_RTC_EVENT_LOG" ] deps = [ "..:video_stream_api", + "../base:rtc_base_approved", "../call:call_interfaces", "../logging:rtc_event_log_impl", "../logging:rtc_event_log_parser", "../modules:module_api", "../modules/audio_coding:ana_debug_dump_proto", "../modules/audio_coding:neteq_tools", - "../rtc_base:rtc_base_approved", # TODO(kwiberg): Remove this dependency. "../api/audio_codecs:audio_codecs_api", @@ -245,7 +245,7 @@ if (rtc_include_tests) { defines = [ "ENABLE_RTC_EVENT_LOG" ] deps = [ ":event_log_visualizer_utils", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "../test:field_trial", "../test:test_support", ] @@ -264,9 +264,9 @@ if (rtc_include_tests) { } deps = [ + "../base:rtc_base_approved", "../modules:module_api", "../modules/audio_processing", - "../rtc_base:rtc_base_approved", "../system_wrappers:metrics_default", "../test:test_support", "//build/win:default_exe_manifest", diff --git a/webrtc/rtc_tools/network_tester/BUILD.gn b/webrtc/rtc_tools/network_tester/BUILD.gn index 49a625d527..bd069d6109 100644 --- a/webrtc/rtc_tools/network_tester/BUILD.gn +++ b/webrtc/rtc_tools/network_tester/BUILD.gn @@ -41,10 +41,10 @@ if (rtc_enable_protobuf) { deps = [ ":network_tester_config_proto", ":network_tester_packet_proto", + "../../base:protobuf_utils", + "../../base:rtc_task_queue", + "../../base:sequenced_task_checker", "../../p2p", - "../../rtc_base:protobuf_utils", - "../../rtc_base:rtc_task_queue", - "../../rtc_base:sequenced_task_checker", ] if (!build_with_chromium && is_clang) { @@ -84,7 +84,7 @@ if (rtc_enable_protobuf) { deps = [ ":network_tester", "//testing/gtest", - "//webrtc/rtc_base:rtc_base_tests_utils", + "//webrtc/base:rtc_base_tests_utils", "//webrtc/test:test_support", ] diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn index b2b396bb11..0a15fc527b 100644 --- a/webrtc/sdk/BUILD.gn +++ b/webrtc/sdk/BUILD.gn @@ -63,7 +63,7 @@ if (is_ios || is_mac) { ] deps = [ - "../rtc_base:rtc_base", + "../base:rtc_base", ] configs += [ "..:common_objc" ] @@ -98,7 +98,7 @@ if (is_ios || is_mac) { deps = [ ":objc_common", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] if (is_clang) { @@ -127,9 +127,9 @@ if (is_ios || is_mac) { ":objc_common", "../api:libjingle_peerconnection_api", "../api:video_frame_api", + "../base:rtc_base", "../common_video", "../media:rtc_media_base", - "../rtc_base:rtc_base", ] configs += [ "..:common_objc" ] @@ -181,9 +181,9 @@ if (is_ios || is_mac) { ":objc_common", ":objc_videotracksource", "../api:libjingle_peerconnection_api", + "../base:rtc_base", "../common_video", "../media:rtc_media_base", - "../rtc_base:rtc_base", ] configs += [ "..:common_objc" ] @@ -247,7 +247,7 @@ if (is_ios || is_mac) { deps = [ ":objc_video", "../api:video_frame_api", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] configs += [ "..:common_objc" ] public_configs = [ ":objc_common_config" ] @@ -289,9 +289,9 @@ if (is_ios || is_mac) { ":objc_peerconnectionfactory", ":objc_video", "../api:video_frame_api", + "../base:rtc_base", "../media:rtc_media_base", "../pc:libjingle_peerconnection", - "../rtc_base:rtc_base", ] if (rtc_use_metal_rendering) { @@ -334,12 +334,12 @@ if (is_ios || is_mac) { ":objc_videotracksource", "../api:video_frame_api", "../api/video_codecs:video_codecs_api", + "../base:rtc_base", "../media:rtc_audio_video", "../media:rtc_media_base", "../modules:module_api", "../pc:create_pc_factory", "../pc:peerconnection", - "../rtc_base:rtc_base", "../system_wrappers:field_trial_api", ] } @@ -371,7 +371,7 @@ if (is_ios || is_mac) { deps = [ ":objc_peerconnectionfactory_base", "../api:libjingle_peerconnection_api", - "../rtc_base:rtc_base", + "../base:rtc_base", ] } @@ -484,11 +484,11 @@ if (is_ios || is_mac) { ":objc_corevideoframebuffer", ":objc_videotracksource", "../api:video_frame_api", + "../base:rtc_base", "../common_video", "../media:rtc_media_base", "../modules:module_api", "../pc:peerconnection", - "../rtc_base:rtc_base", ] } @@ -530,7 +530,7 @@ if (is_ios || is_mac) { deps = [ ":objc_peerconnection", "..//system_wrappers:system_wrappers_default", - "../rtc_base:rtc_base_tests_utils", + "../base:rtc_base_tests_utils", "../system_wrappers:system_wrappers_default", "//third_party/ocmock", ] @@ -632,7 +632,7 @@ if (is_ios || is_mac) { ":objc_audio", ":objc_peerconnection", ":objc_ui", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "../system_wrappers:field_trial_default", "../system_wrappers:metrics_default", ] @@ -673,8 +673,8 @@ if (is_ios || is_mac) { ] deps = [ + "../base:rtc_base_approved", "../common_video", - "../rtc_base:rtc_base_approved", ] if (!build_with_chromium && is_clang) { @@ -705,13 +705,13 @@ if (is_ios || is_mac) { ":objc_common", ":objc_video", ":objc_videotracksource", + "../base:rtc_base_approved", "../common_video", "../media:rtc_media", "../media:rtc_media_base", "../modules:module_api", "../modules/video_coding:video_coding_utility", "../modules/video_coding:webrtc_h264", - "../rtc_base:rtc_base_approved", "../system_wrappers", ] diff --git a/webrtc/sdk/android/BUILD.gn b/webrtc/sdk/android/BUILD.gn index d9cf609db9..689d6cfb40 100644 --- a/webrtc/sdk/android/BUILD.gn +++ b/webrtc/sdk/android/BUILD.gn @@ -49,8 +49,8 @@ rtc_source_set("base_jni") { deps = [ "//webrtc/api:libjingle_peerconnection_api", - "//webrtc/rtc_base:rtc_base", - "//webrtc/rtc_base:rtc_base_approved", + "//webrtc/base:rtc_base", + "//webrtc/base:rtc_base_approved", "//webrtc/system_wrappers:metrics_api", ] @@ -139,16 +139,16 @@ rtc_static_library("video_jni") { "//webrtc/api:libjingle_peerconnection_api", "//webrtc/api:video_frame_api", "//webrtc/api/video_codecs:video_codecs_api", + "//webrtc/base:rtc_base", + "//webrtc/base:rtc_base_approved", + "//webrtc/base:rtc_task_queue", + "//webrtc/base:sequenced_task_checker", + "//webrtc/base:weak_ptr", "//webrtc/common_video:common_video", "//webrtc/media:rtc_audio_video", "//webrtc/media:rtc_media_base", "//webrtc/modules/utility:utility", "//webrtc/modules/video_coding:video_coding_utility", - "//webrtc/rtc_base:rtc_base", - "//webrtc/rtc_base:rtc_base_approved", - "//webrtc/rtc_base:rtc_task_queue", - "//webrtc/rtc_base:sequenced_task_checker", - "//webrtc/rtc_base:weak_ptr", "//webrtc/system_wrappers:system_wrappers", ] @@ -237,13 +237,13 @@ rtc_static_library("peerconnection_jni") { deps = [ ":base_jni", "../..:webrtc_common", + "//webrtc/base:rtc_base", + "//webrtc/base:rtc_base_approved", + "//webrtc/base:rtc_task_queue", "//webrtc/media:rtc_data", "//webrtc/media:rtc_media_base", "//webrtc/modules/utility:utility", "//webrtc/pc:peerconnection", - "//webrtc/rtc_base:rtc_base", - "//webrtc/rtc_base:rtc_base_approved", - "//webrtc/rtc_base:rtc_task_queue", "//webrtc/system_wrappers:system_wrappers", ] } @@ -294,9 +294,9 @@ rtc_shared_library("libjingle_peerconnection_datachannelonly_so") { ":null_media_jni", ":null_video_jni", ":peerconnection_jni", + "//webrtc/base:rtc_base", + "//webrtc/base:rtc_base_approved", "//webrtc/pc:peerconnection", - "//webrtc/rtc_base:rtc_base", - "//webrtc/rtc_base:rtc_base_approved", ] output_extension = "so" } @@ -312,8 +312,8 @@ rtc_shared_library("libjingle_peerconnection_so") { deps = [ ":libjingle_peerconnection_jni", ":libjingle_peerconnection_metrics_default_jni", + "//webrtc/base:rtc_base", "//webrtc/pc:libjingle_peerconnection", - "//webrtc/rtc_base:rtc_base", ] output_extension = "so" } diff --git a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm index 5d6b6c60b1..4131a45242 100644 --- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm +++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm @@ -17,7 +17,7 @@ #import "WebRTC/RTCVideoFrame.h" #import "WebRTC/RTCVideoFrameBuffer.h" -#include "webrtc/rtc_base/timeutils.h" +#include "webrtc/base/timeutils.h" #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h" #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.h" #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.h" diff --git a/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm b/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm index c4d9bd14fa..61c8032206 100644 --- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm +++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm @@ -18,11 +18,11 @@ #import "WebRTC/RTCVideoFrameBuffer.h" #include "webrtc/api/video_codecs/video_decoder.h" +#include "webrtc/base/logging.h" +#include "webrtc/base/timeutils.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/include/video_codec_interface.h" #include "webrtc/modules/video_coding/include/video_error_codes.h" -#include "webrtc/rtc_base/logging.h" -#include "webrtc/rtc_base/timeutils.h" #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h" namespace webrtc { diff --git a/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm b/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm index a063237c7c..e63c527cb3 100644 --- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm +++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm @@ -19,11 +19,11 @@ #include "webrtc/api/video/video_frame.h" #include "webrtc/api/video_codecs/video_encoder.h" +#include "webrtc/base/logging.h" +#include "webrtc/base/timeutils.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/include/video_codec_interface.h" #include "webrtc/modules/video_coding/include/video_error_codes.h" -#include "webrtc/rtc_base/logging.h" -#include "webrtc/rtc_base/timeutils.h" #include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h" #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h" diff --git a/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm b/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm index 49cc812b81..ec3e226a03 100644 --- a/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm +++ b/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm @@ -12,7 +12,7 @@ #include -#include "webrtc/rtc_base/gunit.h" +#include "webrtc/base/gunit.h" #import "NSString+StdString.h" #import "WebRTC/RTCTracing.h" diff --git a/webrtc/stats/BUILD.gn b/webrtc/stats/BUILD.gn index eaa6f5d491..4a2f578e5b 100644 --- a/webrtc/stats/BUILD.gn +++ b/webrtc/stats/BUILD.gn @@ -24,7 +24,7 @@ rtc_static_library("rtc_stats") { deps = [ "../api:rtc_stats_api", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } @@ -58,9 +58,9 @@ if (rtc_include_tests) { ":rtc_stats", ":rtc_stats_test_utils", "../api:rtc_stats_api", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_main", - "../rtc_base:rtc_base_tests_utils", + "../base:rtc_base_approved", + "../base:rtc_base_tests_main", + "../base:rtc_base_tests_utils", "../system_wrappers:metrics_default", "//testing/gmock", ] diff --git a/webrtc/system_wrappers/BUILD.gn b/webrtc/system_wrappers/BUILD.gn index 7dfcff7a3d..1cf1b6f3b0 100644 --- a/webrtc/system_wrappers/BUILD.gn +++ b/webrtc/system_wrappers/BUILD.gn @@ -107,10 +107,10 @@ rtc_static_library("system_wrappers") { cflags = [ "/wd4334" ] # Ignore warning on shift operator promotion. - # Windows needs //webrtc/rtc_base:rtc_base due to include of - # webrtc/rtc_base/win32.h in source/clock.cc. + # Windows needs //webrtc/base:rtc_base due to include of webrtc/base/win32.h + # in source/clock.cc. # TODO(kjellander): Remove (bugs.webrtc.org/6828) - deps += [ "../rtc_base:rtc_base" ] + deps += [ "../base:rtc_base" ] } if (is_win && is_clang) { @@ -118,7 +118,7 @@ rtc_static_library("system_wrappers") { suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] } - deps += [ "../rtc_base:rtc_base_approved" ] + deps += [ "../base:rtc_base_approved" ] } rtc_source_set("cpu_features_api") { @@ -148,7 +148,7 @@ rtc_source_set("metrics_api") { ] deps = [ "..:webrtc_common", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } @@ -169,7 +169,7 @@ rtc_static_library("metrics_default") { ] deps = [ ":metrics_api", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] } @@ -228,7 +228,7 @@ if (rtc_include_tests) { ":metrics_default", ":system_wrappers", "..:webrtc_common", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "../test:test_main", "//testing/gtest", ] diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn index 29d02e7981..f54a62252d 100644 --- a/webrtc/test/BUILD.gn +++ b/webrtc/test/BUILD.gn @@ -60,11 +60,11 @@ rtc_source_set("video_test_common") { deps = [ "..:video_stream_api", "..:webrtc_common", + "../base:rtc_base_approved", + "../base:rtc_task_queue", "../common_video", "../media:rtc_media_base", "../modules/video_capture:video_capture_module", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", "../system_wrappers", ] } @@ -87,8 +87,8 @@ rtc_source_set("rtp_test_utils") { deps = [ "..:webrtc_common", + "../base:rtc_base_approved", "../modules/rtp_rtcp", - "../rtc_base:rtc_base_approved", "//testing/gtest", ] } @@ -131,9 +131,9 @@ rtc_source_set("test_support") { deps = [ "..:webrtc_common", + "../base:gtest_prod", + "../base:rtc_base_approved", "../common_video", - "../rtc_base:gtest_prod", - "../rtc_base:rtc_base_approved", "../system_wrappers", "//testing/gmock", "//testing/gtest", @@ -178,7 +178,7 @@ if (!build_with_chromium) { ] deps = [ ":field_trial", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "../system_wrappers:metrics_default", "//testing/gmock", "//testing/gtest", @@ -205,8 +205,8 @@ if (!build_with_chromium) { ":test_support", ":video_test_common", "..:webrtc_common", + "../base:rtc_base_approved", "../common_video", - "../rtc_base:rtc_base_approved", "../system_wrappers", "//testing/gmock", "//testing/gtest", @@ -243,7 +243,7 @@ if (!build_with_chromium) { ] deps = [ ":fileutils", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "//third_party/gflags", ] } @@ -273,10 +273,10 @@ if (!build_with_chromium) { ":fake_audio_device", ":rtp_test_utils", "../api:video_frame_api", + "../base:rtc_base_approved", "../call:call_interfaces", "../common_audio", "../modules/rtp_rtcp", - "../rtc_base:rtc_base_approved", "../system_wrappers", ] sources = [ @@ -342,14 +342,14 @@ rtc_source_set("fileutils") { ] deps = [ "..:webrtc_common", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", ] if (is_ios) { sources += [ "testsupport/iosfileutils.mm" ] deps += [ "../sdk:objc_common" ] } if (is_win) { - deps += [ "../rtc_base:rtc_base" ] + deps += [ "../base:rtc_base" ] } visibility = [ ":*" ] } @@ -375,7 +375,7 @@ rtc_source_set("fileutils_unittests") { deps = [ ":fileutils", ":test_support", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "//testing/gmock", "//testing/gtest", ] @@ -396,9 +396,9 @@ rtc_source_set("direct_transport") { deps = [ "..:webrtc_common", "../api:transport_api", + "../base:rtc_base_approved", "../call", "../modules/rtp_rtcp", - "../rtc_base:rtc_base_approved", "../system_wrappers", ] } @@ -415,9 +415,9 @@ rtc_source_set("fake_audio_device") { } deps = [ "..:webrtc_common", + "../base:rtc_base_approved", "../common_audio:common_audio", "../modules/audio_device:audio_device", - "../rtc_base:rtc_base_approved", "../system_wrappers:system_wrappers", ] } @@ -478,6 +478,9 @@ rtc_source_set("test_common") { "../api/audio_codecs:builtin_audio_encoder_factory", "../api/video_codecs:video_codecs_api", "../audio", + "../base:rtc_base_approved", + "../base:rtc_task_queue", + "../base:sequenced_task_checker", "../call", "../common_video", "../logging:rtc_event_log_api", @@ -489,9 +492,6 @@ rtc_source_set("test_common") { "../modules/video_coding:webrtc_h264", "../modules/video_coding:webrtc_vp8", "../modules/video_coding:webrtc_vp9", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", - "../rtc_base:sequenced_task_checker", "../system_wrappers", "../video", "../voice_engine", @@ -571,9 +571,9 @@ rtc_source_set("test_renderer") { deps = [ ":test_support", "..:webrtc_common", + "../base:rtc_base_approved", "../common_video", "../modules/media_file", - "../rtc_base:rtc_base_approved", "//testing/gtest", ] } @@ -593,7 +593,7 @@ rtc_source_set("audio_codec_mocks") { ":test_support", "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_decoder_factory", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "//testing/gmock", ] } diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn index b206c72d6c..3e68470749 100644 --- a/webrtc/test/fuzzers/BUILD.gn +++ b/webrtc/test/fuzzers/BUILD.gn @@ -15,7 +15,7 @@ rtc_static_library("webrtc_fuzzer_main") { "webrtc_fuzzer_main.cc", ] deps = [ - "../../rtc_base:rtc_base_approved", + "../../base:rtc_base_approved", "../../system_wrappers:field_trial_default", "../../system_wrappers:metrics_default", "//testing/libfuzzer:libfuzzer_main", @@ -95,8 +95,8 @@ webrtc_fuzzer_test("flexfec_header_reader_fuzzer") { "flexfec_header_reader_fuzzer.cc", ] deps = [ + "../../base:rtc_base_approved", "../../modules/rtp_rtcp", - "../../rtc_base:rtc_base_approved", ] } @@ -116,9 +116,9 @@ webrtc_fuzzer_test("ulpfec_header_reader_fuzzer") { "ulpfec_header_reader_fuzzer.cc", ] deps = [ + "../../base:rtc_base_approved", "../../modules/rtp_rtcp", "../../modules/rtp_rtcp:fec_test_helper", - "../../rtc_base:rtc_base_approved", ] } @@ -127,9 +127,9 @@ webrtc_fuzzer_test("ulpfec_generator_fuzzer") { "ulpfec_generator_fuzzer.cc", ] deps = [ + "../../base:rtc_base_approved", "../../modules/rtp_rtcp", "../../modules/rtp_rtcp:fec_test_helper", - "../../rtc_base:rtc_base_approved", ] } @@ -138,8 +138,8 @@ webrtc_fuzzer_test("flexfec_receiver_fuzzer") { "flexfec_receiver_fuzzer.cc", ] deps = [ + "../../base:rtc_base_approved", "../../modules/rtp_rtcp", - "../../rtc_base:rtc_base_approved", ] libfuzzer_options = [ "max_len=2000" ] } @@ -160,8 +160,8 @@ webrtc_fuzzer_test("rtcp_receiver_fuzzer") { "rtcp_receiver_fuzzer.cc", ] deps = [ + "../../base:rtc_base_approved", "../../modules/rtp_rtcp", - "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", ] seed_corpus = "corpora/rtcp-corpus" @@ -207,8 +207,8 @@ rtc_static_library("audio_decoder_fuzzer") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", + "../../base:rtc_base_approved", "../../modules/rtp_rtcp", - "../../rtc_base:rtc_base_approved", ] } @@ -286,13 +286,13 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") { "neteq_rtp_fuzzer.cc", ] deps = [ + "../../base:rtc_base_approved", + "../../base:rtc_base_tests_utils", "../../modules/audio_coding:neteq", "../../modules/audio_coding:neteq_test_tools", "../../modules/audio_coding:neteq_tools_minimal", "../../modules/audio_coding:pcm16b", "../../modules/rtp_rtcp", - "../../rtc_base:rtc_base_approved", - "../../rtc_base:rtc_base_tests_utils", ] } @@ -301,8 +301,8 @@ webrtc_fuzzer_test("residual_echo_detector_fuzzer") { "residual_echo_detector_fuzzer.cc", ] deps = [ + "../../base:rtc_base_approved", "../../modules/audio_processing:audio_processing", - "../../rtc_base:rtc_base_approved", ] } @@ -343,8 +343,8 @@ webrtc_fuzzer_test("pseudotcp_parser_fuzzer") { "pseudotcp_parser_fuzzer.cc", ] deps = [ + "../../base:rtc_base", "../../p2p:rtc_p2p", - "../../rtc_base:rtc_base", ] } @@ -353,8 +353,8 @@ webrtc_fuzzer_test("transport_feedback_packet_loss_tracker_fuzzer") { "transport_feedback_packet_loss_tracker_fuzzer.cc", ] deps = [ + "../../base:rtc_base_approved", "../../modules/rtp_rtcp", - "../../rtc_base:rtc_base_approved", "../../voice_engine", ] } @@ -366,8 +366,8 @@ webrtc_fuzzer_test("audio_processing_fuzzer") { "audio_processing_fuzzer_configs.cc", ] deps = [ + "../../base:rtc_base_approved", "../../modules:module_api", "../../modules/audio_processing", - "../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn index 529a73c100..5faf28048e 100644 --- a/webrtc/video/BUILD.gn +++ b/webrtc/video/BUILD.gn @@ -58,6 +58,11 @@ rtc_static_library("video") { "..:webrtc_common", "../api:transport_api", "../api/video_codecs:video_codecs_api", + "../base:rtc_base_approved", + "../base:rtc_numerics", + "../base:rtc_task_queue", + "../base:sequenced_task_checker", + "../base:weak_ptr", "../call:call_interfaces", "../call:rtp_interfaces", "../common_video", @@ -74,11 +79,6 @@ rtc_static_library("video") { "../modules/video_coding:video_coding_utility", "../modules/video_coding:webrtc_vp8", "../modules/video_processing", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_numerics", - "../rtc_base:rtc_task_queue", - "../rtc_base:sequenced_task_checker", - "../rtc_base:weak_ptr", "../system_wrappers", "../voice_engine", ] @@ -93,6 +93,8 @@ if (rtc_include_tests) { "video_quality_test.h", ] deps = [ + "../base:rtc_base_tests_utils", + "../base:rtc_task_queue", "../call:call_interfaces", "../common_video", "../logging:rtc_event_log_api", @@ -103,8 +105,6 @@ if (rtc_include_tests) { "../modules/video_coding:webrtc_h264", "../modules/video_coding:webrtc_vp8", "../modules/video_coding:webrtc_vp9", - "../rtc_base:rtc_base_tests_utils", - "../rtc_base:rtc_task_queue", "../system_wrappers", "../test:test_common", "../test:test_support", @@ -155,7 +155,7 @@ if (rtc_include_tests) { ] deps = [ ":video_quality_test", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "../system_wrappers:metrics_default", "../test:field_trial", "../test:run_test", @@ -180,7 +180,7 @@ if (rtc_include_tests) { deps = [ ":video_quality_test", - "../rtc_base:rtc_base_approved", + "../base:rtc_base_approved", "../system_wrappers:metrics_default", "../test:field_trial", "../test:run_test", @@ -203,11 +203,11 @@ if (rtc_include_tests) { deps = [ "..:webrtc_common", "../api/video_codecs:video_codecs_api", + "../base:rtc_base_approved", "../call:call_interfaces", "../common_video", "../logging:rtc_event_log_api", "../modules/rtp_rtcp", - "../rtc_base:rtc_base_approved", "../system_wrappers", "../system_wrappers:metrics_default", "../test:field_trial", @@ -260,6 +260,8 @@ if (rtc_include_tests) { "..:video_stream_api", "../api:video_frame_api", "../api/video_codecs:video_codecs_api", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", "../call:call_interfaces", "../call:rtp_receiver", "../common_video", @@ -277,8 +279,6 @@ if (rtc_include_tests) { "../modules/video_coding:webrtc_h264", "../modules/video_coding:webrtc_vp8", "../modules/video_coding:webrtc_vp9", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", "../system_wrappers", "../system_wrappers:field_trial_default", "../system_wrappers:metrics_api", diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index 78c92f6ec6..e16b1762a7 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -37,10 +37,10 @@ rtc_static_library("file_player") { deps = [ ":audio_coder", "..:webrtc_common", + "../base:rtc_base_approved", "../common_audio", "../modules:module_api", "../modules/media_file", - "../rtc_base:rtc_base_approved", ] if (!build_with_chromium && is_clang) { @@ -58,10 +58,10 @@ rtc_static_library("file_recorder") { ":audio_coder", "..:webrtc_common", "../audio/utility:audio_frame_operations", + "../base:rtc_base_approved", "../common_audio", "../modules:module_api", "../modules/media_file:media_file", - "../rtc_base:rtc_base_approved", "../system_wrappers", ] @@ -143,6 +143,8 @@ rtc_static_library("voice_engine") { "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", "../audio/utility:audio_frame_operations", + "../base:rtc_base_approved", + "../base:rtc_task_queue", "../call:rtp_interfaces", "../common_audio", "../logging:rtc_event_log_api", @@ -157,8 +159,6 @@ rtc_static_library("voice_engine") { "../modules/pacing", "../modules/rtp_rtcp", "../modules/utility", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", "../system_wrappers", ] } @@ -171,9 +171,9 @@ rtc_static_library("audio_level") { deps = [ "..:webrtc_common", + "../base:rtc_base_approved", "../common_audio", "../modules:module_api", - "../rtc_base:rtc_base_approved", ] } @@ -182,9 +182,9 @@ if (rtc_include_tests) { deps = [ ":file_player", ":voice_engine", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", "../modules:module_api", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", "../test:test_common", "//testing/gmock", "//testing/gtest", @@ -247,11 +247,11 @@ if (rtc_include_tests) { deps = [ ":voice_engine", "..:webrtc_common", + "../base:rtc_base_approved", "../modules:module_api", "../modules/audio_device:audio_device", "../modules/audio_processing:audio_processing", "../modules/rtp_rtcp:rtp_rtcp", - "../rtc_base:rtc_base_approved", "//testing/gmock", "//testing/gtest", "//third_party/gflags",