Remove remains of webrtc/base
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index ca6a70a..d29c4c4 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -18,7 +18,6 @@
'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',
@@ -74,7 +73,6 @@
# 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',
@@ -91,6 +89,7 @@
'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 a4cc0ee..0863ea8 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -232,8 +232,8 @@
deps = [
":webrtc_common",
"api:transport_api",
- "base:rtc_base_approved",
"common_video:common_video",
+ "rtc_base:rtc_base_approved",
]
}
@@ -252,7 +252,6 @@
"api",
"api:transport_api",
"audio",
- "base",
"call",
"common_audio",
"common_video",
@@ -291,7 +290,6 @@
":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",
@@ -306,6 +304,7 @@
"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",
@@ -393,16 +392,16 @@
":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",
]
@@ -440,12 +439,12 @@
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",
@@ -518,7 +517,7 @@
rtc_test("webrtc_nonparallel_tests") {
testonly = true
deps = [
- "base:rtc_base_nonparallel_tests",
+ "rtc_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 e87509b..7c43808 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -28,7 +28,7 @@
":audio_mixer_api",
":transport_api",
"..:webrtc_common",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"audio_codecs:audio_codecs_api",
]
}
@@ -83,8 +83,8 @@
deps = [
":rtc_stats_api",
"..:webrtc_common",
- "../base:rtc_base",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
"audio_codecs:audio_codecs_api",
]
@@ -143,7 +143,7 @@
]
deps = [
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -153,8 +153,8 @@
]
deps = [
- "../base:rtc_base_approved",
"../modules:module_api",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -178,7 +178,7 @@
]
deps = [
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers",
]
@@ -206,7 +206,7 @@
]
deps = [
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -235,7 +235,7 @@
]
deps = [
":libjingle_peerconnection_api",
- "../base:rtc_base_approved",
+ "../rtc_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 416ccbb..2174fb1 100644
--- a/webrtc/api/audio_codecs/BUILD.gn
+++ b/webrtc/api/audio_codecs/BUILD.gn
@@ -27,7 +27,7 @@
]
deps = [
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -38,8 +38,8 @@
]
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 @@
]
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 d2470a2..2c1349a 100644
--- a/webrtc/api/audio_codecs/g722/BUILD.gn
+++ b/webrtc/api/audio_codecs/g722/BUILD.gn
@@ -26,8 +26,8 @@
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 @@
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 bba2662..6ef8856 100644
--- a/webrtc/api/audio_codecs/ilbc/BUILD.gn
+++ b/webrtc/api/audio_codecs/ilbc/BUILD.gn
@@ -26,8 +26,8 @@
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 @@
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 c7f7ac8..29a68ff 100644
--- a/webrtc/api/audio_codecs/opus/BUILD.gn
+++ b/webrtc/api/audio_codecs/opus/BUILD.gn
@@ -18,7 +18,7 @@
"audio_encoder_opus_config.h",
]
deps = [
- "../../../base:rtc_base_approved",
+ "../../../rtc_base:rtc_base_approved",
]
defines = []
if (rtc_opus_variable_complexity) {
@@ -35,9 +35,9 @@
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 @@
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 32cef2d..4a0c878 100644
--- a/webrtc/api/audio_codecs/test/BUILD.gn
+++ b/webrtc/api/audio_codecs/test/BUILD.gn
@@ -21,8 +21,8 @@
]
deps = [
"..:audio_codecs_api",
- "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed?
- "../../../base:rtc_base_approved",
+ "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed?
+ "../../../rtc_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 d435534..5e27c78 100644
--- a/webrtc/api/video_codecs/BUILD.gn
+++ b/webrtc/api/video_codecs/BUILD.gn
@@ -21,7 +21,7 @@
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 495875d..a75d8ef 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -39,8 +39,6 @@
"../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",
@@ -52,6 +50,8 @@
"../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",
]
@@ -80,14 +80,14 @@
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",
@@ -151,8 +151,8 @@
"test/audio_bwe_integration_test.h",
]
deps = [
- "../base:rtc_base_approved",
"../common_audio",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:fake_audio_device",
"../test:field_trial",
diff --git a/webrtc/audio/utility/BUILD.gn b/webrtc/audio/utility/BUILD.gn
index ac477e4..65f9cb0 100644
--- a/webrtc/audio/utility/BUILD.gn
+++ b/webrtc/audio/utility/BUILD.gn
@@ -21,9 +21,9 @@
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 @@
]
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
deleted file mode 100644
index c786f15..0000000
--- a/webrtc/base/BUILD.gn
+++ /dev/null
@@ -1,135 +0,0 @@
-# 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
deleted file mode 100644
index 60cd440..0000000
--- a/webrtc/base/Dummy.java
+++ /dev/null
@@ -1,9 +0,0 @@
-/**
- * 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/README.md b/webrtc/base/README.md
new file mode 100644
index 0000000..3d0bf66
--- /dev/null
+++ b/webrtc/base/README.md
@@ -0,0 +1,20 @@
+# What?
+
+The contents of base have moved to rtc_base.
+
+# Why?
+
+We want to move all the contents in the webrtc directory to top-level, because:
+* When we migrate from Rietveld to PolyGerrit we won't be able to apply WebRTC
+patches on Chromium trybots (which is a very useful feature, especially as we
+plan to have such trybots in our default trybot set).
+This is because PolyGerrit needs the repo in Chromium to be the same as the
+WebRTC repo, otherwise it won’t be able to apply the patch.
+* To fully automate rolling into Chromium DEPS, we wish to use the LKGR finder,
+but doing so is blocked on this (needs the same revision hashes to avoid ugly
+hacks). See [this bug](http://crbug.com/666726).
+
+# Tracking Bugs
+
+[Chromium tracking bug](http://crbug.com/611808)
+[WebRTC tracking bug](https://bugs.chromium.org/p/webrtc/issues/detail?id=7634)
diff --git a/webrtc/base/array_view.h b/webrtc/base/array_view.h
deleted file mode 100644
index a451b59..0000000
--- a/webrtc/base/array_view.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 8b37efa..0000000
--- a/webrtc/base/arraysize.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index cce4226..0000000
--- a/webrtc/base/asyncinvoker-inl.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 0fcfc04..0000000
--- a/webrtc/base/asyncinvoker.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 809f178..0000000
--- a/webrtc/base/asyncpacketsocket.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index b2a172f..0000000
--- a/webrtc/base/asyncresolverinterface.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 9c97139..0000000
--- a/webrtc/base/asyncsocket.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d64927b..0000000
--- a/webrtc/base/asynctcpsocket.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index c3212c0..0000000
--- a/webrtc/base/asyncudpsocket.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3c36848..0000000
--- a/webrtc/base/atomicops.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1e28357..0000000
--- a/webrtc/base/base64.h
+++ /dev/null
@@ -1,20 +0,0 @@
-
-//*********************************************************************
-//* 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
deleted file mode 100644
index 42ffa5a..0000000
--- a/webrtc/base/basictypes.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 39d441f..0000000
--- a/webrtc/base/bind.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * 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> bar = new rtc::RefCountedObject<Bar>();
-// auto functor = rtc::Bind(&Bar::Test, bar);
-// bar = nullptr;
-// // The functor stores an internal scoped_refptr<Bar>, 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
deleted file mode 100644
index 09cba3c..0000000
--- a/webrtc/base/bitbuffer.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 92c85d9..0000000
--- a/webrtc/base/buffer.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3142ae3..0000000
--- a/webrtc/base/bufferqueue.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 0cc9a12..0000000
--- a/webrtc/base/bytebuffer.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 28cbaa5..0000000
--- a/webrtc/base/byteorder.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 4da1e6d..0000000
--- a/webrtc/base/callback.h
+++ /dev/null
@@ -1,70 +0,0 @@
-// 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<ReturnType, ParamType1, ..., ParamTypeN>
-// 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<int, int> my_callback;
-// cout << my_callback.empty() << endl; // true
-//
-// my_callback = Callback1<int, int>(&sqr);
-// cout << my_callback.empty() << endl; // false
-// cout << my_callback(3) << endl; // 9
-//
-// my_callback = Callback1<int, int>(add_k);
-// cout << my_callback(10) << endl; // 15
-//
-// my_callback = Callback1<int, int>();
-// 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
deleted file mode 100644
index f56f157..0000000
--- a/webrtc/base/checks.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 934cc9b..0000000
--- a/webrtc/base/compile_assert_c.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 21652c2..0000000
--- a/webrtc/base/constructormagic.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6a95b31..0000000
--- a/webrtc/base/copyonwritebuffer.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index f627790..0000000
--- a/webrtc/base/cpu_time.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6854567..0000000
--- a/webrtc/base/crc32.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index ab3f542..0000000
--- a/webrtc/base/criticalsection.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1a474b4..0000000
--- a/webrtc/base/cryptstring.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d6c5124..0000000
--- a/webrtc/base/deprecation.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1cf2756..0000000
--- a/webrtc/base/dscp.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 28ff731..0000000
--- a/webrtc/base/event.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index b6da14a..0000000
--- a/webrtc/base/event_tracer.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 22d640d..0000000
--- a/webrtc/base/fakeclock.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index c2c8e6d..0000000
--- a/webrtc/base/fakenetwork.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index da204b2..0000000
--- a/webrtc/base/fakesslidentity.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 5a4465f..0000000
--- a/webrtc/base/file.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 26306db..0000000
--- a/webrtc/base/filerotatingstream.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 18de30c..0000000
--- a/webrtc/base/fileutils.h
+++ /dev/null
@@ -1,20 +0,0 @@
-
-/*
- * 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
deleted file mode 100644
index 18ad9bc..0000000
--- a/webrtc/base/firewallsocketserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 9094466..0000000
--- a/webrtc/base/flags.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 844e71e..0000000
--- a/webrtc/base/format_macros.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1230026..0000000
--- a/webrtc/base/function_view.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 0c25943..0000000
--- a/webrtc/base/gtest_prod_util.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d6c092e..0000000
--- a/webrtc/base/gunit.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 436abee..0000000
--- a/webrtc/base/gunit_prod.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 86a388e..0000000
--- a/webrtc/base/helpers.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index a66ce15..0000000
--- a/webrtc/base/httpbase.h
+++ /dev/null
@@ -1,20 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 7dfe182..0000000
--- a/webrtc/base/httpcommon-inl.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3946dfc..0000000
--- a/webrtc/base/httpcommon.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 4fd75a2..0000000
--- a/webrtc/base/httpserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 9c49c9f..0000000
--- a/webrtc/base/ifaddrs-android.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index de7ad87..0000000
--- a/webrtc/base/ifaddrs_converter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index fdfba9b..0000000
--- a/webrtc/base/ignore_wundef.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 44e432d..0000000
--- a/webrtc/base/ipaddress.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 175028f..0000000
--- a/webrtc/base/json.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 171e048..0000000
--- a/webrtc/base/keep_ref_until_done.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 432471c..0000000
--- a/webrtc/base/location.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 594d9c9..0000000
--- a/webrtc/base/logging.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 95e6dc6..0000000
--- a/webrtc/base/logsinks.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index ed0c4f5..0000000
--- a/webrtc/base/macutils.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 9e5c3ca..0000000
--- a/webrtc/base/mathutils.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index fd17541..0000000
--- a/webrtc/base/md5.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 66d6ee1..0000000
--- a/webrtc/base/md5digest.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 5c22559..0000000
--- a/webrtc/base/memory_usage.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * 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
deleted file mode 100644
index b73f907..0000000
--- a/webrtc/base/messagedigest.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 943d0d7..0000000
--- a/webrtc/base/messagehandler.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 353a4b7..0000000
--- a/webrtc/base/messagequeue.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d61bd05..0000000
--- a/webrtc/base/mod_ops.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index b803ad8..0000000
--- a/webrtc/base/natserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 31c29ab..0000000
--- a/webrtc/base/natsocketfactory.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 001f57f..0000000
--- a/webrtc/base/nattypes.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 9a8e607..0000000
--- a/webrtc/base/nethelpers.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 2953098..0000000
--- a/webrtc/base/network.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 290da4f..0000000
--- a/webrtc/base/networkmonitor.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index b5e8c13..0000000
--- a/webrtc/base/networkroute.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 214c542..0000000
--- a/webrtc/base/nullsocketserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index a4eaea2..0000000
--- a/webrtc/base/numerics/exp_filter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index a9058a2..0000000
--- a/webrtc/base/numerics/percentile_filter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6849bac..0000000
--- a/webrtc/base/onetimeevent.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 795af70..0000000
--- a/webrtc/base/openssl.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6444215..0000000
--- a/webrtc/base/openssladapter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 031c0b1..0000000
--- a/webrtc/base/openssldigest.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 59fa571..0000000
--- a/webrtc/base/opensslidentity.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index e17e029..0000000
--- a/webrtc/base/opensslstreamadapter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 7657ec3..0000000
--- a/webrtc/base/optional.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index e77fd8a..0000000
--- a/webrtc/base/optionsfile.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index b45ca04..0000000
--- a/webrtc/base/pathutils.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 63e6dfa..0000000
--- a/webrtc/base/physicalsocketserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index c7396ec..0000000
--- a/webrtc/base/platform_file.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 626d66f..0000000
--- a/webrtc/base/platform_thread.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index f2dbd58..0000000
--- a/webrtc/base/platform_thread_types.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3d2dd86..0000000
--- a/webrtc/base/protobuf_utils.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * 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 <string>
-
-#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
deleted file mode 100644
index f0ae182..0000000
--- a/webrtc/base/proxyinfo.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1bf580a..0000000
--- a/webrtc/base/proxyserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index aa6f3b4..0000000
--- a/webrtc/base/ptr_util.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 474fdb5..0000000
--- a/webrtc/base/race_checker.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 12a4902..0000000
--- a/webrtc/base/random.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 0cba5fb..0000000
--- a/webrtc/base/rate_limiter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1a17500..0000000
--- a/webrtc/base/rate_statistics.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 0e372db..0000000
--- a/webrtc/base/ratelimiter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d1fd75d..0000000
--- a/webrtc/base/ratetracker.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 4a7cea3..0000000
--- a/webrtc/base/refcount.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 78304fa..0000000
--- a/webrtc/base/refcountedobject.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * 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
deleted file mode 100644
index a7de4f1..0000000
--- a/webrtc/base/rollingaccumulator.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 22d8fe7..0000000
--- a/webrtc/base/rtccertificate.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index fac1cec..0000000
--- a/webrtc/base/rtccertificategenerator.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index acdd9ce..0000000
--- a/webrtc/base/safe_compare.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * 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
deleted file mode 100644
index ac0bb65..0000000
--- a/webrtc/base/safe_conversions.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 497e156..0000000
--- a/webrtc/base/safe_conversions_impl.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 54d99b7..0000000
--- a/webrtc/base/safe_minmax.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 56a5e10..0000000
--- a/webrtc/base/sanitizer.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 2599562..0000000
--- a/webrtc/base/scoped_ref_ptr.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * 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<MyFoo> {
-// ...
-// };
-//
-// void some_function() {
-// scoped_refptr<MyFoo> foo = new MyFoo();
-// foo->Method(param);
-// // |foo| is released when this function returns
-// }
-//
-// void some_other_function() {
-// scoped_refptr<MyFoo> foo = new MyFoo();
-// ...
-// foo = nullptr; // explicitly releases |foo|
-// ...
-// if (foo)
-// foo->Method(param);
-// }
-//
-// The above examples show how scoped_refptr<T> acts like a pointer to T.
-// Given two scoped_refptr<T> classes, it is also possible to exchange
-// references between the two objects, like so:
-//
-// {
-// scoped_refptr<MyFoo> a = new MyFoo();
-// scoped_refptr<MyFoo> 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<MyFoo> a = new MyFoo();
-// scoped_refptr<MyFoo> 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
deleted file mode 100644
index e586b8d..0000000
--- a/webrtc/base/sequenced_task_checker.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 4972539..0000000
--- a/webrtc/base/sequenced_task_checker_impl.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index fde3e59..0000000
--- a/webrtc/base/sha1.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * SHA-1 in C
- * By Steve Reid <sreid@sea-to-sky.net>
- * 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
deleted file mode 100644
index e3b4ef8..0000000
--- a/webrtc/base/sha1digest.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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/signalthread.h b/webrtc/base/signalthread.h
deleted file mode 100644
index f5fcf2c..0000000
--- a/webrtc/base/signalthread.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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_SIGNALTHREAD_H_
-#define WEBRTC_BASE_SIGNALTHREAD_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/signalthread.h"
-
-#endif // WEBRTC_BASE_SIGNALTHREAD_H_
diff --git a/webrtc/base/sigslot.h b/webrtc/base/sigslot.h
deleted file mode 100644
index 9d31441..0000000
--- a/webrtc/base/sigslot.h
+++ /dev/null
@@ -1,104 +0,0 @@
-// 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<single_threaded> and signal0<multi_threaded_local> or
-// has_slots<multi_threaded_local> and signal0<single_threaded>.
-// 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
deleted file mode 100644
index 545bf9e..0000000
--- a/webrtc/base/sigslottester.h
+++ /dev/null
@@ -1,23 +0,0 @@
-// 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
deleted file mode 100644
index 19ea7a0..0000000
--- a/webrtc/base/socket.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index f6769f9..0000000
--- a/webrtc/base/socket_unittest.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 7df0f3a..0000000
--- a/webrtc/base/socketadapters.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 20199ad..0000000
--- a/webrtc/base/socketaddress.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3f53f10..0000000
--- a/webrtc/base/socketaddresspair.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3a829ac..0000000
--- a/webrtc/base/socketfactory.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 55b427d..0000000
--- a/webrtc/base/socketserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index a76ffb3..0000000
--- a/webrtc/base/socketstream.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3d432ec..0000000
--- a/webrtc/base/ssladapter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6be82fd..0000000
--- a/webrtc/base/sslfingerprint.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1cedfa0..0000000
--- a/webrtc/base/sslidentity.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 9fa706b..0000000
--- a/webrtc/base/sslroots.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d7c062e..0000000
--- a/webrtc/base/sslstreamadapter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 18dd865..0000000
--- a/webrtc/base/stream.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index fa88ba4..0000000
--- a/webrtc/base/string_to_number.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 27b810e..0000000
--- a/webrtc/base/stringencode.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 5f8a5b1..0000000
--- a/webrtc/base/stringize_macros.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * 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
deleted file mode 100644
index e3b5e07..0000000
--- a/webrtc/base/stringutils.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 7111147..0000000
--- a/webrtc/base/swap_queue.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 12f5cbb..0000000
--- a/webrtc/base/task_queue.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6cb51a0..0000000
--- a/webrtc/base/task_queue_posix.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 9a05643..0000000
--- a/webrtc/base/template_util.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * 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
deleted file mode 100644
index fc9846f..0000000
--- a/webrtc/base/testbase64.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 378e2b8..0000000
--- a/webrtc/base/testclient.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 21365e2..0000000
--- a/webrtc/base/testechoserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 74f2160..0000000
--- a/webrtc/base/testutils.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6a6887a..0000000
--- a/webrtc/base/thread.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 5b94ffe..0000000
--- a/webrtc/base/thread_annotations.h
+++ /dev/null
@@ -1,27 +0,0 @@
-//
-// 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
deleted file mode 100644
index ade5256..0000000
--- a/webrtc/base/thread_checker.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3a0a6c7..0000000
--- a/webrtc/base/thread_checker_impl.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * 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
deleted file mode 100644
index f2e98a8..0000000
--- a/webrtc/base/timedelta.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 60c3631..0000000
--- a/webrtc/base/timestampaligner.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1569b58..0000000
--- a/webrtc/base/timeutils.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 1bea5f4..0000000
--- a/webrtc/base/trace_event.h
+++ /dev/null
@@ -1,14 +0,0 @@
-// 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
deleted file mode 100644
index 3d9c86b..0000000
--- a/webrtc/base/transformadapter.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6a4ac8d..0000000
--- a/webrtc/base/type_traits.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 7a18205..0000000
--- a/webrtc/base/unixfilesystem.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 31ce96d..0000000
--- a/webrtc/base/virtualsocketserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 282a551..0000000
--- a/webrtc/base/weak_ptr.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 413bd11..0000000
--- a/webrtc/base/win32.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d647c44..0000000
--- a/webrtc/base/win32filesystem.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d7017e1..0000000
--- a/webrtc/base/win32socketinit.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index c143692..0000000
--- a/webrtc/base/win32socketserver.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index ffffdf9..0000000
--- a/webrtc/base/win32window.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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
deleted file mode 100644
index d515f7c..0000000
--- a/webrtc/base/window.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * 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 c26c761..c40b557 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -28,8 +28,8 @@
"../api:libjingle_peerconnection_api",
"../api:transport_api",
"../api/audio_codecs:audio_codecs_api",
- "../base:rtc_base",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -43,7 +43,7 @@
"rtp_transport_controller_send_interface.h",
]
deps = [
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -64,8 +64,8 @@
deps = [
":rtp_interfaces",
"..:webrtc_common",
- "../base:rtc_base_approved",
"../modules/rtp_rtcp",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -76,8 +76,8 @@
]
deps = [
":rtp_interfaces",
- "../base:rtc_base_approved",
"../modules/congestion_controller",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -109,7 +109,6 @@
"..:webrtc_common",
"../api:transport_api",
"../audio",
- "../base:rtc_task_queue",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl",
"../modules/bitrate_controller",
@@ -117,6 +116,7 @@
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/utility",
+ "../rtc_base:rtc_task_queue",
"../system_wrappers",
"../video",
]
@@ -149,7 +149,6 @@
":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",
@@ -158,6 +157,7 @@
"../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 @@
":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 ff0aa26..f7f3efb 100644
--- a/webrtc/common_audio/BUILD.gn
+++ b/webrtc/common_audio/BUILD.gn
@@ -63,8 +63,8 @@
deps = [
":sinc_resampler",
"..:webrtc_common",
- "../base:gtest_prod",
- "../base:rtc_base_approved",
+ "../rtc_base:gtest_prod",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers",
]
public_deps = [
@@ -209,8 +209,8 @@
":common_audio_c_arm_asm",
":common_audio_cc",
"..:webrtc_common",
- "../base:compile_assert_c",
- "../base:rtc_base_approved",
+ "../rtc_base:compile_assert_c",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:system_wrappers",
]
}
@@ -225,7 +225,7 @@
public_configs = [ ":common_audio_config" ]
deps = [
"..:webrtc_common",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:system_wrappers",
]
}
@@ -236,8 +236,8 @@
]
deps = [
"..:webrtc_common",
- "../base:gtest_prod",
- "../base:rtc_base_approved",
+ "../rtc_base:gtest_prod",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers",
]
}
@@ -344,7 +344,7 @@
}
deps = [
":common_audio_c",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
}
@@ -401,8 +401,8 @@
":common_audio",
":sinc_resampler",
"..:webrtc_common",
- "../base:rtc_base_approved",
- "../base:rtc_base_tests_utils",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_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 8f0afab..93906ef 100644
--- a/webrtc/common_video/BUILD.gn
+++ b/webrtc/common_video/BUILD.gn
@@ -57,10 +57,10 @@
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 @@
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 726b5ba..004b9a2 100644
--- a/webrtc/examples/BUILD.gn
+++ b/webrtc/examples/BUILD.gn
@@ -523,12 +523,12 @@
deps += [
"../api:libjingle_peerconnection_test_api",
"../api:video_frame_api",
- "../base:rtc_base",
- "../base:rtc_base_approved",
- "../base:rtc_json",
"../media:rtc_media",
"../modules/video_capture:video_capture_module",
"../pc:libjingle_peerconnection",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_base:rtc_json",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"//third_party/libyuv",
@@ -548,11 +548,8 @@
]
deps = [
"..:webrtc_common",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../rtc_tools:command_line_parser",
- "//webrtc:webrtc_common",
- "//webrtc/base:rtc_base_approved",
- "//webrtc/rtc_tools:command_line_parser",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@@ -565,10 +562,10 @@
"relayserver/relayserver_main.cc",
]
deps = [
- "../base:rtc_base",
- "../base:rtc_base_approved",
"../p2p:rtc_p2p",
"../pc:rtc_pc",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
]
@@ -583,10 +580,10 @@
"turnserver/turnserver_main.cc",
]
deps = [
- "../base:rtc_base",
- "../base:rtc_base_approved",
"../p2p:rtc_p2p",
"../pc:rtc_pc",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
]
@@ -601,10 +598,10 @@
"stunserver/stunserver_main.cc",
]
deps = [
- "../base:rtc_base",
- "../base:rtc_base_approved",
"../p2p:rtc_p2p",
"../pc:rtc_pc",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
]
@@ -636,13 +633,13 @@
deps = [
"../api:libjingle_peerconnection_test_api",
"../api:video_frame_api",
- "../base:rtc_base",
- "../base:rtc_base_approved",
- "../base:rtc_json",
"../media:rtc_media",
"../media:rtc_media_base",
"../modules/video_capture:video_capture_module",
"../pc:libjingle_peerconnection",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_base:rtc_json",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
]
@@ -664,10 +661,10 @@
}
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 f3c3469..6a70324 100644
--- a/webrtc/logging/BUILD.gn
+++ b/webrtc/logging/BUILD.gn
@@ -30,7 +30,7 @@
deps = [
"..:video_stream_api",
"..:webrtc_common",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -48,11 +48,11 @@
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 @@
}
deps = [
"..:video_stream_api",
- "../base:protobuf_utils",
- "../base:rtc_base_approved",
+ "../rtc_base:protobuf_utils",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -111,12 +111,12 @@
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 @@
":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 @@
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_parser",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
# TODO(kwiberg): Remove this dependency.
"../api/audio_codecs:audio_codecs_api",
@@ -182,7 +182,7 @@
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_proto",
- "../base:rtc_base_approved",
+ "../rtc_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 cb5bf2b..f905d8a 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -45,8 +45,8 @@
deps = [
"..:webrtc_common",
- "../base:rtc_base",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -115,9 +115,9 @@
":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) {
@@ -227,10 +227,6 @@
"../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",
@@ -245,6 +241,10 @@
"../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 @@
"..: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 @@
"../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",
]
@@ -508,10 +508,6 @@
"../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",
@@ -521,6 +517,10 @@
"../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 94e2d4f..359bd8c 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -37,7 +37,7 @@
deps = [
"..:webrtc_common",
"../api:video_frame_api",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 3ea9215..d69d978 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -47,7 +47,7 @@
deps = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -58,8 +58,8 @@
]
deps = [
"../..:webrtc_common",
- "../../base:protobuf_utils",
- "../../base:rtc_base_approved",
+ "../../rtc_base:protobuf_utils",
+ "../../rtc_base:rtc_base_approved",
"../../api/audio_codecs:audio_codecs_api",
] + audio_codec_deps
defines = audio_codec_defines
@@ -72,8 +72,8 @@
]
deps = [
"../..:webrtc_common",
- "../../base:protobuf_utils",
- "../../base:rtc_base_approved",
+ "../../rtc_base:protobuf_utils",
+ "../../rtc_base:rtc_base_approved",
"../../api/audio_codecs:audio_codecs_api",
] + audio_codec_deps
defines = audio_codec_defines
@@ -89,8 +89,8 @@
deps = [
"../../api/audio_codecs:audio_codecs_api",
"../..:webrtc_common",
- "../../base:protobuf_utils",
- "../../base:rtc_base_approved",
+ "../../rtc_base:protobuf_utils",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
":audio_coding_module_typedefs",
":isac_common",
@@ -156,7 +156,7 @@
":audio_coding_module_typedefs",
":neteq",
":rent_a_codec",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../logging:rtc_event_log_api",
]
defines = audio_coding_defines
@@ -169,7 +169,7 @@
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -193,8 +193,8 @@
deps = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
"../../common_audio",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -212,8 +212,8 @@
deps = [
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
"../../common_audio",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -238,7 +238,7 @@
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
public_deps = [
":g711_c",
@@ -280,7 +280,7 @@
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs/g722:audio_encoder_g722_config",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
public_deps = [
":g722_c",
@@ -323,8 +323,8 @@
"../..: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 @@
deps = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
"../../common_audio",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -495,7 +495,7 @@
deps = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -587,9 +587,9 @@
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 @@
":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 @@
deps = [
":isac_fix_common",
- "../../base:rtc_base_approved",
"../../common_audio",
+ "../../rtc_base:rtc_base_approved",
]
}
}
@@ -799,7 +799,7 @@
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
public_deps = [
":pcm16b_c",
@@ -837,10 +837,10 @@
"../..: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 @@
deps = [
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -926,10 +926,10 @@
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 @@
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -1042,9 +1042,9 @@
"..: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 @@
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../rtp_rtcp",
]
}
@@ -1134,9 +1134,9 @@
":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 @@
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 @@
}
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 @@
"..:module_api",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
]
@@ -1342,8 +1342,8 @@
":neteq_test_tools",
":webrtc_opus",
"../..:webrtc_common",
- "../../base:protobuf_utils",
- "../../base:rtc_base_approved",
+ "../../rtc_base:protobuf_utils",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
]
@@ -1369,7 +1369,7 @@
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
":neteq_tools",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_support",
"//testing/gtest",
]
@@ -1388,7 +1388,7 @@
":audio_coding",
":neteq_tools",
"../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_support",
"//testing/gtest",
]
@@ -1412,7 +1412,7 @@
":audio_format_conversion",
"..:module_api",
"../../:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@@ -1442,7 +1442,7 @@
":audio_format_conversion",
"..:module_api",
"../../:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@@ -1489,8 +1489,8 @@
":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 @@
":neteq",
":neteq_test_tools",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
"//third_party/gflags",
@@ -1573,7 +1573,7 @@
":isac_fix",
":webrtc_opus",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_main",
"../audio_processing",
@@ -1603,7 +1603,7 @@
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
"//testing/gtest",
@@ -1628,7 +1628,7 @@
"..:module_api",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_support",
"//testing/gtest",
"//third_party/gflags",
@@ -1705,8 +1705,8 @@
":pcm16b",
":webrtc_opus",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
"../../common_audio",
+ "../../rtc_base:rtc_base_approved",
]
configs += [ ":RTPencode_config" ]
@@ -1749,7 +1749,7 @@
]
deps = [
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:rtp_test_utils",
"//testing/gtest",
@@ -1774,7 +1774,7 @@
testonly = true
deps = [
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_support",
"//testing/gtest",
]
@@ -1853,7 +1853,7 @@
":neteq_quality_test_support",
":neteq_tools",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_main",
"//testing/gtest",
@@ -1872,7 +1872,7 @@
":isac_fix",
":neteq",
":neteq_quality_test_support",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_main",
"//testing/gtest",
"//third_party/gflags",
@@ -1890,7 +1890,7 @@
":g711",
":neteq",
":neteq_quality_test_support",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_main",
"//testing/gtest",
"//third_party/gflags",
@@ -1950,7 +1950,7 @@
deps = [
":isac",
":isac_test_util",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
configs += [ ":isac_test_warnings_config" ]
@@ -1991,7 +1991,7 @@
deps = [
":isac",
":isac_test_util",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
include_dirs = [
@@ -2042,8 +2042,8 @@
deps = [
":webrtc_opus",
- "../../base:rtc_base_approved",
"../../common_audio",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_main",
"//testing/gtest",
]
@@ -2167,11 +2167,11 @@
"../../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 ab2fc0d..16a62b8 100644
--- a/webrtc/modules/audio_conference_mixer/BUILD.gn
+++ b/webrtc/modules/audio_conference_mixer/BUILD.gn
@@ -42,7 +42,7 @@
"..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
- "../../base:rtc_base_approved",
+ "../../rtc_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 2c262ba..6cf2ea7 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -51,9 +51,9 @@
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 @@
}
if (is_ios) {
public_deps = [
- "../../base:gtest_prod",
- "../../base:rtc_base",
+ "../../rtc_base:gtest_prod",
+ "../../rtc_base:rtc_base",
"../../sdk:objc_audio",
"../../sdk:objc_common",
]
@@ -281,7 +281,7 @@
deps = [
":audio_device",
":mock_audio_device",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
"../utility:utility",
@@ -331,7 +331,7 @@
deps = [
":audio_device",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_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 b6569f7..e7323f1 100644
--- a/webrtc/modules/audio_mixer/BUILD.gn
+++ b/webrtc/modules/audio_mixer/BUILD.gn
@@ -41,7 +41,7 @@
"..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../audio_processing",
]
@@ -61,7 +61,7 @@
deps = [
"..:module_api",
"../../audio/utility",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -90,8 +90,8 @@
"..:module_api",
"../../api:audio_mixer_api",
"../../audio/utility:audio_frame_operations",
- "../../base:rtc_base_approved",
- "../../base:rtc_task_queue",
+ "../../rtc_base:rtc_base_approved",
+ "../../rtc_base:rtc_task_queue",
"../../test:test_support",
"//testing/gmock",
]
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 92a8f17..4fd6e9f 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -238,8 +238,8 @@
"..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
- "../../base:gtest_prod",
- "../../base:protobuf_utils",
+ "../../rtc_base:gtest_prod",
+ "../../rtc_base:protobuf_utils",
"../audio_coding:isac",
]
public_deps = [
@@ -303,8 +303,8 @@
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 @@
]
deps = [
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -356,8 +356,8 @@
deps = [
"../..:webrtc_common",
- "../../base:rtc_base_approved",
"../../common_audio",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
@@ -470,7 +470,7 @@
]
}
deps = [
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
}
}
@@ -550,11 +550,11 @@
":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 @@
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_unittest_proto",
- "../../base:rtc_task_queue",
+ "../../rtc_base:rtc_task_queue",
"aec_dump",
"aec_dump:aec_dump_unittests",
]
@@ -696,7 +696,7 @@
deps = [
":audio_processing",
":audioproc_test_utils",
- "../../base:protobuf_utils",
+ "../../rtc_base:protobuf_utils",
"//testing/gtest",
]
@@ -720,9 +720,9 @@
":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 @@
":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 @@
deps = [
":audio_processing",
"..:module_api",
- "../../base:rtc_base_approved",
"../../common_audio",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
]
}
@@ -825,8 +825,8 @@
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 @@
deps = [
":audioproc_debug_proto",
"../..:webrtc_common",
- "../../base:protobuf_utils",
- "../../base:rtc_base_approved",
+ "../../rtc_base:protobuf_utils",
+ "../../rtc_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 950dd68..818a9bf 100644
--- a/webrtc/modules/audio_processing/aec_dump/BUILD.gn
+++ b/webrtc/modules/audio_processing/aec_dump/BUILD.gn
@@ -18,7 +18,7 @@
]
deps = [
- "../../../base:rtc_base_approved",
+ "../../../rtc_base:rtc_base_approved",
]
}
@@ -49,7 +49,7 @@
deps = [
":mock_aec_dump",
"..:audio_processing",
- "../../../base:rtc_base_approved",
+ "../../../rtc_base:rtc_base_approved",
"//testing/gtest",
]
}
@@ -73,10 +73,10 @@
]
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 @@
":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 af24f8a..587663b 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 @@
]
deps = [
":lib",
- "../../../../../webrtc/base:rtc_base_approved",
- "../../../../../webrtc/test:test_support",
+ "../../../../rtc_base:rtc_base_approved",
+ "../../../../test:test_support",
"//third_party/gflags",
]
}
@@ -45,9 +45,9 @@
"wavreader_interface.h",
]
deps = [
- "../../../../../webrtc:webrtc_common",
- "../../../../../webrtc/base:rtc_base_approved",
- "../../../../../webrtc/common_audio",
+ "../../../..:webrtc_common",
+ "../../../../common_audio",
+ "../../../../rtc_base:rtc_base_approved",
]
visibility = [ ":*" ] # Only targets in this file can depend on this.
}
@@ -63,14 +63,11 @@
]
deps = [
":lib",
- "../../../../../webrtc:webrtc_common",
- "../../../../../webrtc/base:rtc_base_approved",
- "../../../../../webrtc/common_audio",
- "../../../../../webrtc/test:test_support",
+ "../../../..:webrtc_common",
+ "../../../../common_audio",
+ "../../../../rtc_base:rtc_base_approved",
+ "../../../../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 2d1b1db..44fbd2f 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 @@
output_name = "py_quality_assessment/quality_assessment/fake_polqa"
deps = [
"../../../..:webrtc_common",
- "../../../../base:rtc_base_approved",
+ "../../../../rtc_base:rtc_base_approved",
]
}
diff --git a/webrtc/modules/bitrate_controller/BUILD.gn b/webrtc/modules/bitrate_controller/BUILD.gn
index 4ab057a..7c2063e 100644
--- a/webrtc/modules/bitrate_controller/BUILD.gn
+++ b/webrtc/modules/bitrate_controller/BUILD.gn
@@ -37,7 +37,7 @@
}
deps = [
- "../../base:rtc_base_approved",
+ "../../rtc_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 c3ee11f..27921b7 100644
--- a/webrtc/modules/congestion_controller/BUILD.gn
+++ b/webrtc/modules/congestion_controller/BUILD.gn
@@ -51,10 +51,10 @@
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",
@@ -91,9 +91,9 @@
deps = [
":congestion_controller",
":mock_congestion_controller",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
- "../../base:rtc_base_tests_utils",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
+ "../../rtc_base:rtc_base_tests_utils",
"../../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 889f0fa..24140fa 100644
--- a/webrtc/modules/desktop_capture/BUILD.gn
+++ b/webrtc/modules/desktop_capture/BUILD.gn
@@ -28,7 +28,7 @@
deps = [
"../..:webrtc_common",
- "../../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
+ "../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
]
}
@@ -49,8 +49,8 @@
":desktop_capture_mock",
":primitives",
":screen_drawer",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
"../../test:video_test_support",
@@ -96,7 +96,7 @@
":desktop_capture_mock",
":primitives",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
"//testing/gmock",
@@ -133,7 +133,7 @@
deps = [
":primitives",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
}
@@ -157,7 +157,7 @@
deps = [
":primitives",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_support",
]
}
@@ -292,7 +292,7 @@
deps = [
":primitives",
"../..:webrtc_common",
- "../../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
+ "../../rtc_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 baea158..62cd1ad 100644
--- a/webrtc/modules/media_file/BUILD.gn
+++ b/webrtc/modules/media_file/BUILD.gn
@@ -35,8 +35,8 @@
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 77c2a81..b22c178 100644
--- a/webrtc/modules/pacing/BUILD.gn
+++ b/webrtc/modules/pacing/BUILD.gn
@@ -30,8 +30,8 @@
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",
@@ -58,8 +58,8 @@
]
deps = [
":pacing",
- "../../base:rtc_base_approved",
- "../../base:rtc_base_tests_utils",
+ "../../rtc_base:rtc_base_approved",
+ "../../rtc_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 ffe5a83..8a48077 100644
--- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn
+++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
@@ -51,8 +51,8 @@
deps = [
"../..:webrtc_common",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
}
@@ -118,9 +118,9 @@
":remote_bitrate_estimator",
"..:module_api",
"../..:webrtc_common",
- "../../base:gtest_prod",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
+ "../../rtc_base:gtest_prod",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
"../../voice_engine",
@@ -148,7 +148,7 @@
deps = [
":bwe_simulator_lib",
":remote_bitrate_estimator",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_support",
]
if (!build_with_chromium && is_clang) {
@@ -189,8 +189,8 @@
":mock_remote_bitrate_observer",
":remote_bitrate_estimator",
"../..:webrtc_common",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:field_trial",
"../../test:test_support",
@@ -231,7 +231,7 @@
":bwe_simulator_lib",
":remote_bitrate_estimator",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_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 91bfd85..75e11f3 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -170,11 +170,11 @@
"../../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 @@
deps = [
":rtp_rtcp",
"..:module_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
]
# TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
@@ -221,7 +221,7 @@
deps = [
":rtp_rtcp",
"..:module_api",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_support",
]
}
@@ -256,7 +256,7 @@
]
deps = [
":rtp_rtcp",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../test:test_support",
]
if (!build_with_chromium && is_clang) {
@@ -342,8 +342,8 @@
"..: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 54730a7..b40cbd5 100644
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@@ -33,8 +33,8 @@
"..: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 @@
deps = [
":utility",
"..:module_api",
- "../../base:rtc_task_queue",
+ "../../rtc_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 b908923..86c1336 100644
--- a/webrtc/modules/video_capture/BUILD.gn
+++ b/webrtc/modules/video_capture/BUILD.gn
@@ -28,8 +28,8 @@
deps = [
"..:module_api",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
"../../common_video",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
@@ -47,7 +47,7 @@
deps = [
":video_capture_module",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
@@ -91,7 +91,7 @@
deps = [
":video_capture_module",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
@@ -175,8 +175,8 @@
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 8d15e8b..8ff2bcb 100644
--- a/webrtc/modules/video_coding/BUILD.gn
+++ b/webrtc/modules/video_coding/BUILD.gn
@@ -97,12 +97,12 @@
"..: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 @@
"..: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 @@
defines = []
deps = [
":video_coding_utility",
- "../../base:rtc_base_approved",
"../../media:rtc_media_base",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
@@ -198,8 +198,8 @@
deps = [
"../..:webrtc_common",
- "../../base:rtc_base_approved",
"../../common_video:common_video",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
}
@@ -232,9 +232,9 @@
"..: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 @@
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 @@
":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 @@
":video_coding",
":webrtc_vp8",
"../..:webrtc_common",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:field_trial_default",
"../../system_wrappers:metrics_default",
"../../system_wrappers:system_wrappers",
@@ -354,8 +354,8 @@
":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 @@
":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 @@
]
deps += [
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../sdk/android:libjingle_peerconnection_jni",
"//base",
]
@@ -428,8 +428,8 @@
":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 (is_android) {
deps += [
- "../../base:rtc_base_approved",
+ "../../rtc_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 @@
"../..: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 3f0fb76..efe5458 100644
--- a/webrtc/modules/video_processing/BUILD.gn
+++ b/webrtc/modules/video_processing/BUILD.gn
@@ -27,10 +27,10 @@
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 @@
deps = [
":denoiser_filter",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers",
]
diff --git a/webrtc/ortc/BUILD.gn b/webrtc/ortc/BUILD.gn
index b6a2cc9..40c0b2a 100644
--- a/webrtc/ortc/BUILD.gn
+++ b/webrtc/ortc/BUILD.gn
@@ -35,8 +35,6 @@
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",
@@ -45,6 +43,8 @@
"../p2p:rtc_p2p",
"../pc:libjingle_peerconnection",
"../pc:rtc_pc",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
]
public_deps = [
@@ -76,14 +76,14 @@
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 daa2d84..16b8762 100644
--- a/webrtc/p2p/BUILD.gn
+++ b/webrtc/p2p/BUILD.gn
@@ -87,7 +87,7 @@
deps = [
"../api:libjingle_peerconnection_api",
"../api:ortc_api",
- "../base:rtc_base",
+ "../rtc_base:rtc_base",
"../system_wrappers:field_trial_api",
]
@@ -155,9 +155,9 @@
deps = [
":rtc_p2p",
"../api:ortc_api",
- "../base:rtc_base",
- "../base:rtc_base_approved",
- "../base:rtc_base_tests_utils",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_base:rtc_base_tests_utils",
"../test:test_support",
"//testing/gmock",
]
@@ -209,9 +209,9 @@
":rtc_p2p",
"../api:fakemetricsobserver",
"../api:ortc_api",
- "../base:rtc_base",
- "../base:rtc_base_approved",
- "../base:rtc_base_tests_utils",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_base:rtc_base_tests_utils",
"../test:test_support",
"//testing/gmock",
"//testing/gtest",
@@ -238,7 +238,7 @@
deps = [
":rtc_p2p",
"..:webrtc_common",
- "../base:rtc_base",
+ "../rtc_base:rtc_base",
]
}
@@ -259,8 +259,8 @@
":libstunprober",
":p2p_test_utils",
":rtc_p2p",
- "../base:rtc_base",
- "../base:rtc_base_tests_utils",
+ "../rtc_base:rtc_base",
+ "../rtc_base:rtc_base_tests_utils",
"//testing/gmock",
"//testing/gtest",
]
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index 783fd67..2585416 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -60,12 +60,12 @@
"../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 @@
"../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 @@
"../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 @@
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 @@
"..: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 @@
":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 851973b..925f915 100644
--- a/webrtc/rtc_base/BUILD.gn
+++ b/webrtc/rtc_base/BUILD.gn
@@ -203,8 +203,8 @@
# Dependency on chromium's logging (in //base).
deps += [ "//base:base" ]
sources += [
- "../../webrtc_overrides/webrtc/base/logging.cc",
- "../../webrtc_overrides/webrtc/base/logging.h",
+ "../../webrtc_overrides/webrtc/rtc_base/logging.cc",
+ "../../webrtc_overrides/webrtc/rtc_base/logging.h",
]
} else {
sources += [
@@ -301,8 +301,8 @@
if (build_with_chromium) {
sources = [
- "../../webrtc_overrides/webrtc/base/task_queue.cc",
- "../../webrtc_overrides/webrtc/base/task_queue.h",
+ "../../webrtc_overrides/webrtc/rtc_base/task_queue.cc",
+ "../../webrtc_overrides/webrtc/rtc_base/task_queue.h",
]
} else {
sources = [
@@ -518,7 +518,7 @@
if (build_with_chromium) {
if (is_win) {
- sources += [ "../../webrtc_overrides/webrtc/base/win32socketinit.cc" ]
+ sources += [ "../../webrtc_overrides/webrtc/rtc_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 2389952..cceddf7 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/base/refcount.h"
-#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/rtc_base/refcount.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
namespace rtc {
diff --git a/webrtc/rtc_base/signalthread.cc b/webrtc/rtc_base/signalthread.cc
index be2741e..93fa73d 100644
--- a/webrtc/rtc_base/signalthread.cc
+++ b/webrtc/rtc_base/signalthread.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/base/signalthread.h"
+#include "webrtc/rtc_base/signalthread.h"
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
namespace rtc {
diff --git a/webrtc/rtc_base/signalthread.h b/webrtc/rtc_base/signalthread.h
index bc8c98e..f6722a7 100644
--- a/webrtc/rtc_base/signalthread.h
+++ b/webrtc/rtc_base/signalthread.h
@@ -13,11 +13,11 @@
#include <string>
-#include "webrtc/base/checks.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/nullsocketserver.h"
-#include "webrtc/base/sigslot.h"
-#include "webrtc/base/thread.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/nullsocketserver.h"
+#include "webrtc/rtc_base/sigslot.h"
+#include "webrtc/rtc_base/thread.h"
namespace rtc {
diff --git a/webrtc/rtc_base/signalthread_unittest.cc b/webrtc/rtc_base/signalthread_unittest.cc
index 53bb006..99d3206 100644
--- a/webrtc/rtc_base/signalthread_unittest.cc
+++ b/webrtc/rtc_base/signalthread_unittest.cc
@@ -10,10 +10,10 @@
#include <memory>
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/signalthread.h"
-#include "webrtc/base/thread.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/gunit.h"
+#include "webrtc/rtc_base/signalthread.h"
+#include "webrtc/rtc_base/thread.h"
using namespace rtc;
diff --git a/webrtc/rtc_base/sigslottester.h.pump b/webrtc/rtc_base/sigslottester.h.pump
index a88f0c6..381b791 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/base/constructormagic.h"
-#include "webrtc/base/sigslot.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/sigslot.h"
namespace rtc {
diff --git a/webrtc/rtc_tools/BUILD.gn b/webrtc/rtc_tools/BUILD.gn
index fac42f3..dde557e 100644
--- a/webrtc/rtc_tools/BUILD.gn
+++ b/webrtc/rtc_tools/BUILD.gn
@@ -48,8 +48,8 @@
"simple_command_line_parser.h",
]
deps = [
- "../base:gtest_prod",
- "../base:rtc_base_approved",
+ "../rtc_base:gtest_prod",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -206,13 +206,13 @@
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 @@
defines = [ "ENABLE_RTC_EVENT_LOG" ]
deps = [
":event_log_visualizer_utils",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../test:field_trial",
"../test:test_support",
]
@@ -264,9 +264,9 @@
}
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 4a262fc..847db66 100644
--- a/webrtc/rtc_tools/network_tester/BUILD.gn
+++ b/webrtc/rtc_tools/network_tester/BUILD.gn
@@ -41,10 +41,10 @@
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) {
@@ -83,7 +83,7 @@
deps = [
":network_tester",
- "../../base:rtc_base_tests_utils",
+ "../../rtc_base:rtc_base_tests_utils",
"../../test:test_support",
"//testing/gtest",
]
diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn
index d08a2f7..e691a8e 100644
--- a/webrtc/sdk/BUILD.gn
+++ b/webrtc/sdk/BUILD.gn
@@ -63,7 +63,7 @@
]
deps = [
- "../base:rtc_base",
+ "../rtc_base:rtc_base",
]
configs += [ "..:common_objc" ]
@@ -98,7 +98,7 @@
deps = [
":objc_common",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
if (is_clang) {
@@ -127,9 +127,9 @@
":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 @@
":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 @@
deps = [
":objc_video",
"../api:video_frame_api",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
configs += [ "..:common_objc" ]
public_configs = [ ":objc_common_config" ]
@@ -289,9 +289,9 @@
":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) {
@@ -327,10 +327,10 @@
":objc_video",
":objc_videotoolbox",
"../api:video_frame_api",
- "../base:rtc_base",
"../media:rtc_media_base",
"../pc:create_pc_factory",
"../pc:peerconnection",
+ "../rtc_base:rtc_base",
]
}
@@ -361,7 +361,7 @@
deps = [
":objc_peerconnectionfactory_base",
"../api:libjingle_peerconnection_api",
- "../base:rtc_base",
+ "../rtc_base:rtc_base",
]
}
@@ -473,10 +473,10 @@
":objc_corevideoframebuffer",
":objc_videotracksource",
"../api:video_frame_api",
- "../base:rtc_base",
"../common_video",
"../media:rtc_media_base",
"../pc:peerconnection",
+ "../rtc_base:rtc_base",
]
}
@@ -519,7 +519,7 @@
deps = [
":objc_peerconnection",
"..//system_wrappers:system_wrappers_default",
- "../base:rtc_base_tests_utils",
+ "../rtc_base:rtc_base_tests_utils",
"../system_wrappers:system_wrappers_default",
"//third_party/ocmock",
]
@@ -619,7 +619,7 @@
":objc_audio",
":objc_peerconnection",
":objc_ui",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
]
@@ -660,8 +660,8 @@
]
deps = [
- "../base:rtc_base_approved",
"../common_video",
+ "../rtc_base:rtc_base_approved",
]
if (!build_with_chromium && is_clang) {
@@ -692,13 +692,13 @@
":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 1291e52..401b11f 100644
--- a/webrtc/sdk/android/BUILD.gn
+++ b/webrtc/sdk/android/BUILD.gn
@@ -49,8 +49,8 @@
deps = [
"../../api:libjingle_peerconnection_api",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_api",
]
@@ -139,16 +139,16 @@
"../../api:libjingle_peerconnection_api",
"../../api:video_frame_api",
"../../api/video_codecs:video_codecs_api",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
- "../../base:rtc_task_queue",
- "../../base:sequenced_task_checker",
- "../../base:weak_ptr",
"../../common_video:common_video",
"../../media:rtc_audio_video",
"../../media:rtc_media_base",
"../../modules/utility:utility",
"../../modules/video_coding:video_coding_utility",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
+ "../../rtc_base:rtc_task_queue",
+ "../../rtc_base:sequenced_task_checker",
+ "../../rtc_base:weak_ptr",
"../../system_wrappers:system_wrappers",
]
@@ -237,13 +237,13 @@
deps = [
":base_jni",
"../..:webrtc_common",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
- "../../base:rtc_task_queue",
"../../media:rtc_data",
"../../media:rtc_media_base",
"../../modules/utility:utility",
"../../pc:peerconnection",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
+ "../../rtc_base:rtc_task_queue",
"../../system_wrappers:system_wrappers",
]
}
@@ -294,9 +294,9 @@
":null_media_jni",
":null_video_jni",
":peerconnection_jni",
- "../../base:rtc_base",
- "../../base:rtc_base_approved",
"../../pc:peerconnection",
+ "../../rtc_base:rtc_base",
+ "../../rtc_base:rtc_base_approved",
]
output_extension = "so"
}
@@ -312,8 +312,8 @@
deps = [
":libjingle_peerconnection_jni",
":libjingle_peerconnection_metrics_default_jni",
- "../../base:rtc_base",
"../../pc:libjingle_peerconnection",
+ "../../rtc_base:rtc_base",
]
output_extension = "so"
}
diff --git a/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm b/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm
index ec3e226..49cc812 100644
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm
+++ b/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm
@@ -12,7 +12,7 @@
#include <vector>
-#include "webrtc/base/gunit.h"
+#include "webrtc/rtc_base/gunit.h"
#import "NSString+StdString.h"
#import "WebRTC/RTCTracing.h"
diff --git a/webrtc/stats/BUILD.gn b/webrtc/stats/BUILD.gn
index 4a2f578..eaa6f5d 100644
--- a/webrtc/stats/BUILD.gn
+++ b/webrtc/stats/BUILD.gn
@@ -24,7 +24,7 @@
deps = [
"../api:rtc_stats_api",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -58,9 +58,9 @@
":rtc_stats",
":rtc_stats_test_utils",
"../api:rtc_stats_api",
- "../base:rtc_base_approved",
- "../base:rtc_base_tests_main",
- "../base:rtc_base_tests_utils",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_base:rtc_base_tests_main",
+ "../rtc_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 1cf1b6f..7dfcff7 100644
--- a/webrtc/system_wrappers/BUILD.gn
+++ b/webrtc/system_wrappers/BUILD.gn
@@ -107,10 +107,10 @@
cflags = [ "/wd4334" ] # Ignore warning on shift operator promotion.
- # Windows needs //webrtc/base:rtc_base due to include of webrtc/base/win32.h
- # in source/clock.cc.
+ # Windows needs //webrtc/rtc_base:rtc_base due to include of
+ # webrtc/rtc_base/win32.h in source/clock.cc.
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
- deps += [ "../base:rtc_base" ]
+ deps += [ "../rtc_base:rtc_base" ]
}
if (is_win && is_clang) {
@@ -118,7 +118,7 @@
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
- deps += [ "../base:rtc_base_approved" ]
+ deps += [ "../rtc_base:rtc_base_approved" ]
}
rtc_source_set("cpu_features_api") {
@@ -148,7 +148,7 @@
]
deps = [
"..:webrtc_common",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -169,7 +169,7 @@
]
deps = [
":metrics_api",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -228,7 +228,7 @@
":metrics_default",
":system_wrappers",
"..:webrtc_common",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../test:test_main",
"//testing/gtest",
]
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 2719db4..8f50c11 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -60,11 +60,11 @@
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 @@
deps = [
"..:webrtc_common",
- "../base:rtc_base_approved",
"../modules/rtp_rtcp",
+ "../rtc_base:rtc_base_approved",
"//testing/gtest",
]
}
@@ -131,9 +131,9 @@
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 @@
]
deps = [
":field_trial",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:metrics_default",
"//testing/gmock",
"//testing/gtest",
@@ -205,8 +205,8 @@
":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 @@
]
deps = [
":fileutils",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"//third_party/gflags",
]
}
@@ -273,10 +273,10 @@
":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 @@
]
deps = [
"..:webrtc_common",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
]
if (is_ios) {
sources += [ "testsupport/iosfileutils.mm" ]
deps += [ "../sdk:objc_common" ]
}
if (is_win) {
- deps += [ "../base:rtc_base" ]
+ deps += [ "../rtc_base:rtc_base" ]
}
visibility = [ ":*" ]
}
@@ -375,7 +375,7 @@
deps = [
":fileutils",
":test_support",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"//testing/gmock",
"//testing/gtest",
]
@@ -396,9 +396,9 @@
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 @@
}
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",
]
}
@@ -478,9 +478,6 @@
"../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",
@@ -492,6 +489,9 @@
"../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 @@
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 @@
":test_support",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"//testing/gmock",
]
}
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index 3e68470..b206c72 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -15,7 +15,7 @@
"webrtc_fuzzer_main.cc",
]
deps = [
- "../../base:rtc_base_approved",
+ "../../rtc_base:rtc_base_approved",
"../../system_wrappers:field_trial_default",
"../../system_wrappers:metrics_default",
"//testing/libfuzzer:libfuzzer_main",
@@ -95,8 +95,8 @@
"flexfec_header_reader_fuzzer.cc",
]
deps = [
- "../../base:rtc_base_approved",
"../../modules/rtp_rtcp",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -116,9 +116,9 @@
"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 @@
"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 @@
"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 @@
"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 @@
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 @@
"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 @@
"residual_echo_detector_fuzzer.cc",
]
deps = [
- "../../base:rtc_base_approved",
"../../modules/audio_processing:audio_processing",
+ "../../rtc_base:rtc_base_approved",
]
}
@@ -343,8 +343,8 @@
"pseudotcp_parser_fuzzer.cc",
]
deps = [
- "../../base:rtc_base",
"../../p2p:rtc_p2p",
+ "../../rtc_base:rtc_base",
]
}
@@ -353,8 +353,8 @@
"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 @@
"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 430b00d..990a7ae 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -58,11 +58,6 @@
"..: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",
@@ -79,6 +74,11 @@
"../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",
]
@@ -93,8 +93,6 @@
"video_quality_test.h",
]
deps = [
- "../base:rtc_base_tests_utils",
- "../base:rtc_task_queue",
"../call:call_interfaces",
"../common_video",
"../logging:rtc_event_log_api",
@@ -105,6 +103,8 @@
"../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_renderer",
@@ -158,7 +158,7 @@
]
deps = [
":video_quality_test",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:metrics_default",
"../test:field_trial",
"../test:run_test",
@@ -183,7 +183,7 @@
deps = [
":video_quality_test",
- "../base:rtc_base_approved",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers:metrics_default",
"../test:field_trial",
"../test:run_test",
@@ -206,11 +206,11 @@
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",
@@ -263,8 +263,6 @@
"..: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",
@@ -282,6 +280,8 @@
"../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 e98a89b..e82691a 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -37,10 +37,10 @@
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 @@
":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",
]
@@ -143,8 +143,6 @@
"../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",
@@ -159,6 +157,8 @@
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/utility",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_base:rtc_task_queue",
"../system_wrappers",
]
}
@@ -171,9 +171,9 @@
deps = [
"..:webrtc_common",
- "../base:rtc_base_approved",
"../common_audio",
"../modules:module_api",
+ "../rtc_base:rtc_base_approved",
]
}
@@ -182,8 +182,6 @@
deps = [
":file_player",
":voice_engine",
- "../base:rtc_base_approved",
- "../base:rtc_base_tests_utils",
"../common_audio",
"../modules:module_api",
"../modules/audio_coding",
@@ -194,6 +192,8 @@
"../modules/rtp_rtcp",
"../modules/utility",
"../modules/video_capture:video_capture",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_base:rtc_base_tests_utils",
"../system_wrappers",
"../test:test_common",
"../test:test_main",
@@ -247,13 +247,13 @@
deps = [
":voice_engine",
"..:webrtc_common",
- "../base:rtc_base_approved",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/audio_device:audio_device",
"../modules/audio_processing:audio_processing",
"../modules/rtp_rtcp:rtp_rtcp",
"../modules/video_capture",
+ "../rtc_base:rtc_base_approved",
"../system_wrappers",
"../system_wrappers/:system_wrappers_default",
"../test/:test_common",