Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
Reason for revert:
Breaks lots of downstream projects.
Original issue's description:
> 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/2973183002
> Cr-Commit-Position: refs/heads/master@{#18948}
> Committed:
https://chromium.googlesource.com/external/webrtc/+/9483b49bafc681a8360dff7217e7651a74dea71d
TBR=kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7634
Review-Url: https://codereview.webrtc.org/2976633002
Cr-Original-Commit-Position: refs/heads/master@{#18949}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 370dd479730b4918b8b81842c4c36d25d0633b50
diff --git a/BUILD.gn b/BUILD.gn
index 12ee93c..9280d54 100644
--- a/BUILD.gn
+++ b/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",
]
}
@@ -252,6 +252,7 @@
"api",
"api:transport_api",
"audio",
+ "base",
"call",
"common_audio",
"common_video",
@@ -290,6 +291,7 @@
":video_engine_tests",
":webrtc_nonparallel_tests",
":webrtc_perf_tests",
+ "base:rtc_base_tests_utils",
"common_audio:common_audio_unittests",
"common_video:common_video_unittests",
"media:rtc_media_unittests",
@@ -304,7 +306,6 @@
"ortc:ortc_unittests",
"pc:peerconnection_unittests",
"pc:rtc_pc_unittests",
- "rtc_base:rtc_base_tests_utils",
"stats:rtc_stats_unittests",
"system_wrappers:system_wrappers_unittests",
"test",
@@ -392,16 +393,16 @@
":webrtc_common",
"api:rtc_api_unittests",
"api/audio_codecs/test:audio_codecs_api_unittests",
+ "base:rtc_base_approved_unittests",
+ "base:rtc_base_tests_main",
+ "base:rtc_base_tests_utils",
+ "base:rtc_base_unittests",
+ "base:rtc_numerics_unittests",
+ "base:rtc_task_queue_unittests",
+ "base:sequenced_task_checker_unittests",
+ "base:weak_ptr_unittests",
"p2p:libstunprober_unittests",
"p2p:rtc_p2p_unittests",
- "rtc_base:rtc_base_approved_unittests",
- "rtc_base:rtc_base_tests_main",
- "rtc_base:rtc_base_tests_utils",
- "rtc_base:rtc_base_unittests",
- "rtc_base:rtc_numerics_unittests",
- "rtc_base:rtc_task_queue_unittests",
- "rtc_base:sequenced_task_checker_unittests",
- "rtc_base:weak_ptr_unittests",
"system_wrappers:metrics_default",
]
@@ -439,12 +440,12 @@
testonly = true
deps = [
"audio:audio_tests",
+ "base:rtc_base_tests_utils",
# TODO(eladalon): call_tests aren't actually video-specific, so we
# should move them to a more appropriate test suite.
"call:call_tests",
"modules/video_capture",
- "rtc_base:rtc_base_tests_utils",
"test:test_common",
"test:test_main",
"test:video_test_common",
@@ -516,7 +517,7 @@
rtc_test("webrtc_nonparallel_tests") {
testonly = true
deps = [
- "rtc_base:rtc_base_nonparallel_tests",
+ "base:rtc_base_nonparallel_tests",
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
diff --git a/api/BUILD.gn b/api/BUILD.gn
index c413f28..249411b 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -28,7 +28,7 @@
":audio_mixer_api",
":transport_api",
"..:webrtc_common",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"audio_codecs:audio_codecs_api",
]
}
@@ -83,8 +83,8 @@
deps = [
":rtc_stats_api",
"..:webrtc_common",
- "../rtc_base:rtc_base",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
"audio_codecs:audio_codecs_api",
]
@@ -143,7 +143,7 @@
]
deps = [
- "../rtc_base:rtc_base_approved",
+ "../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 = [
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"../system_wrappers",
]
@@ -206,7 +206,7 @@
]
deps = [
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
}
@@ -235,7 +235,7 @@
]
deps = [
":libjingle_peerconnection_api",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index 2174fb1..416ccbb 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -27,7 +27,7 @@
]
deps = [
"../..:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../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/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn
index 2c1349a..d2470a2 100644
--- a/api/audio_codecs/g722/BUILD.gn
+++ b/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/api/audio_codecs/ilbc/BUILD.gn b/api/audio_codecs/ilbc/BUILD.gn
index 6ef8856..bba2662 100644
--- a/api/audio_codecs/ilbc/BUILD.gn
+++ b/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/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
index 29a68ff..c7f7ac8 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -18,7 +18,7 @@
"audio_encoder_opus_config.h",
]
deps = [
- "../../../rtc_base:rtc_base_approved",
+ "../../../base:rtc_base_approved",
]
defines = []
if (rtc_opus_variable_complexity) {
@@ -35,9 +35,9 @@
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/api/audio_codecs/test/BUILD.gn b/api/audio_codecs/test/BUILD.gn
index 4a0c878..32cef2d 100644
--- a/api/audio_codecs/test/BUILD.gn
+++ b/api/audio_codecs/test/BUILD.gn
@@ -21,8 +21,8 @@
]
deps = [
"..:audio_codecs_api",
- "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed?
- "../../../rtc_base:rtc_base_approved",
+ "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed?
+ "../../../base:rtc_base_approved",
"../../../test:audio_codec_mocks",
"../../../test:test_support",
"../g722:audio_decoder_g722",
diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
index 5e27c78..d435534 100644
--- a/api/video_codecs/BUILD.gn
+++ b/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/audio/BUILD.gn b/audio/BUILD.gn
index 2b7d06f..1577316 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -37,6 +37,8 @@
"../api:call_api",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
+ "../base:rtc_base_approved",
+ "../base:rtc_task_queue",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_audio",
@@ -48,8 +50,6 @@
"../modules/pacing:pacing",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_task_queue",
"../system_wrappers",
"../voice_engine",
]
@@ -77,14 +77,14 @@
deps = [
":audio",
"../api:mock_audio_mixer",
+ "../base:rtc_base_approved",
+ "../base:rtc_task_queue",
"../call:rtp_receiver",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/congestion_controller:congestion_controller",
"../modules/congestion_controller:mock_congestion_controller",
"../modules/pacing:pacing",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_task_queue",
"../test:test_common",
"../test:test_support",
"utility:utility_tests",
diff --git a/audio/utility/BUILD.gn b/audio/utility/BUILD.gn
index 65f9cb0..ac477e4 100644
--- a/audio/utility/BUILD.gn
+++ b/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/base/BUILD.gn b/base/BUILD.gn
new file mode 100644
index 0000000..c786f15
--- /dev/null
+++ b/base/BUILD.gn
@@ -0,0 +1,135 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//build/config/crypto.gni")
+import("//build/config/ui.gni")
+import("../webrtc.gni")
+
+if (is_android) {
+ import("//build/config/android/config.gni")
+ import("//build/config/android/rules.gni")
+}
+if (is_win) {
+ import("//build/config/clang/clang.gni")
+}
+
+group("base") {
+ public_deps = [
+ ":rtc_base",
+ ":rtc_base_approved",
+ ":rtc_task_queue",
+ ":sequenced_task_checker",
+ ":weak_ptr",
+ ]
+}
+
+if (!rtc_build_ssl) {
+ config("external_ssl_library") {
+ assert(rtc_ssl_root != "",
+ "You must specify rtc_ssl_root when rtc_build_ssl==0.")
+ include_dirs = [ rtc_ssl_root ]
+ }
+}
+
+# The targets below are deprecated and only exist here temporarily during
+# refactoring. See https://bugs.webrtc.org/7634 for more details.
+
+group("protobuf_utils") {
+ public_deps = [ "../rtc_base:protobuf_utils" ]
+}
+
+group("compile_assert_c") {
+ public_deps = [ "../rtc_base:compile_assert_c" ]
+}
+
+group("rtc_base_approved") {
+ public_deps = [ "../rtc_base:rtc_base_approved" ]
+}
+
+group("rtc_task_queue") {
+ public_deps = [ "../rtc_base:rtc_task_queue" ]
+}
+
+group("sequenced_task_checker") {
+ public_deps = [ "../rtc_base:sequenced_task_checker" ]
+}
+
+group("weak_ptr") {
+ public_deps = [ "../rtc_base:weak_ptr" ]
+}
+
+group("rtc_numerics") {
+ public_deps = [ "../rtc_base:rtc_numerics" ]
+}
+
+group("rtc_json") {
+ public_deps = [ "../rtc_base:rtc_json" ]
+}
+
+group("rtc_base") {
+ public_deps = [ "../rtc_base:rtc_base" ]
+}
+
+group("gtest_prod") {
+ public_deps = [ "../rtc_base:gtest_prod" ]
+}
+
+group("rtc_base_tests_utils") {
+ testonly = true
+ public_deps = [ "../rtc_base:rtc_base_tests_utils" ]
+}
+
+if (rtc_include_tests) {
+ group("rtc_base_tests_main") {
+ testonly = true
+ public_deps = [ "../rtc_base:rtc_base_tests_main" ]
+ }
+
+ group("rtc_base_nonparallel_tests") {
+ testonly = true
+ public_deps = [ "../rtc_base:rtc_base_nonparallel_tests" ]
+ }
+
+ group("rtc_base_approved_unittests") {
+ testonly = true
+ public_deps = [ "../rtc_base:rtc_base_approved_unittests" ]
+ }
+
+ group("sequenced_task_checker_unittests") {
+ testonly = true
+ public_deps = [ "../rtc_base:sequenced_task_checker_unittests" ]
+ }
+
+ group("weak_ptr_unittests") {
+ testonly = true
+ public_deps = [ "../rtc_base:weak_ptr_unittests" ]
+ }
+
+ group("rtc_task_queue_unittests") {
+ testonly = true
+ public_deps = [ "../rtc_base:rtc_task_queue_unittests" ]
+ }
+
+
+ group("rtc_numerics_unittests") {
+ testonly = true
+ public_deps = [ "../rtc_base:rtc_numerics_unittests" ]
+ }
+
+ group("rtc_base_unittests") {
+ testonly = true
+ public_deps = [ "../rtc_base:rtc_base_unittests" ]
+ }
+}
+
+if (is_android) {
+ android_library("base_java") {
+ java_files = [ "Dummy.java" ] # Need one file to avoid hitting an assert.
+ deps = [ "../rtc_base:base_java" ]
+ }
+}
diff --git a/base/Dummy.java b/base/Dummy.java
new file mode 100644
index 0000000..60cd440
--- /dev/null
+++ b/base/Dummy.java
@@ -0,0 +1,9 @@
+/**
+ * This class only exists as glue in a transition.
+ * TODO(kjellander): Remove.
+ * See https://bugs.webrtc.org/7634 for more details.
+ */
+class Dummy {
+ Dummy() {
+ }
+}
diff --git a/base/array_view.h b/base/array_view.h
new file mode 100644
index 0000000..a451b59
--- /dev/null
+++ b/base/array_view.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ARRAY_VIEW_H_
+#define WEBRTC_BASE_ARRAY_VIEW_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/array_view.h"
+
+#endif // WEBRTC_BASE_ARRAY_VIEW_H_
diff --git a/base/arraysize.h b/base/arraysize.h
new file mode 100644
index 0000000..8b37efa
--- /dev/null
+++ b/base/arraysize.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ARRAYSIZE_H_
+#define WEBRTC_BASE_ARRAYSIZE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/arraysize.h"
+
+#endif // WEBRTC_BASE_ARRAYSIZE_H_
diff --git a/base/asyncinvoker-inl.h b/base/asyncinvoker-inl.h
new file mode 100644
index 0000000..cce4226
--- /dev/null
+++ b/base/asyncinvoker-inl.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ASYNCINVOKER_INL_H_
+#define WEBRTC_BASE_ASYNCINVOKER_INL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/asyncinvoker-inl.h"
+
+#endif // WEBRTC_BASE_ASYNCINVOKER_INL_H_
diff --git a/base/asyncinvoker.h b/base/asyncinvoker.h
new file mode 100644
index 0000000..0fcfc04
--- /dev/null
+++ b/base/asyncinvoker.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ASYNCINVOKER_H_
+#define WEBRTC_BASE_ASYNCINVOKER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/asyncinvoker.h"
+
+#endif // WEBRTC_BASE_ASYNCINVOKER_H_
diff --git a/base/asyncpacketsocket.h b/base/asyncpacketsocket.h
new file mode 100644
index 0000000..809f178
--- /dev/null
+++ b/base/asyncpacketsocket.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_
+#define WEBRTC_BASE_ASYNCPACKETSOCKET_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/asyncpacketsocket.h"
+
+#endif // WEBRTC_BASE_ASYNCPACKETSOCKET_H_
diff --git a/base/asyncresolverinterface.h b/base/asyncresolverinterface.h
new file mode 100644
index 0000000..b2a172f
--- /dev/null
+++ b/base/asyncresolverinterface.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ASYNCRESOLVERINTERFACE_H_
+#define WEBRTC_BASE_ASYNCRESOLVERINTERFACE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/asyncresolverinterface.h"
+
+#endif
diff --git a/base/asyncsocket.h b/base/asyncsocket.h
new file mode 100644
index 0000000..9c97139
--- /dev/null
+++ b/base/asyncsocket.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ASYNCSOCKET_H_
+#define WEBRTC_BASE_ASYNCSOCKET_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/asyncsocket.h"
+
+#endif // WEBRTC_BASE_ASYNCSOCKET_H_
diff --git a/base/asynctcpsocket.h b/base/asynctcpsocket.h
new file mode 100644
index 0000000..d64927b
--- /dev/null
+++ b/base/asynctcpsocket.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ASYNCTCPSOCKET_H_
+#define WEBRTC_BASE_ASYNCTCPSOCKET_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/asynctcpsocket.h"
+
+#endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_
diff --git a/base/asyncudpsocket.h b/base/asyncudpsocket.h
new file mode 100644
index 0000000..c3212c0
--- /dev/null
+++ b/base/asyncudpsocket.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ASYNCUDPSOCKET_H_
+#define WEBRTC_BASE_ASYNCUDPSOCKET_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/asyncudpsocket.h"
+
+#endif // WEBRTC_BASE_ASYNCUDPSOCKET_H_
diff --git a/base/atomicops.h b/base/atomicops.h
new file mode 100644
index 0000000..3c36848
--- /dev/null
+++ b/base/atomicops.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ATOMICOPS_H_
+#define WEBRTC_BASE_ATOMICOPS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/atomicops.h"
+
+#endif // WEBRTC_BASE_ATOMICOPS_H_
diff --git a/base/base64.h b/base/base64.h
new file mode 100644
index 0000000..1e28357
--- /dev/null
+++ b/base/base64.h
@@ -0,0 +1,20 @@
+
+//*********************************************************************
+//* C_Base64 - a simple base64 encoder and decoder.
+//*
+//* Copyright (c) 1999, Bob Withers - bwit@pobox.com
+//*
+//* This code may be freely used for any purpose, either personal
+//* or commercial, provided the authors copyright notice remains
+//* intact.
+//*********************************************************************
+
+#ifndef WEBRTC_BASE_BASE64_H_
+#define WEBRTC_BASE_BASE64_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/base64.h"
+
+#endif // WEBRTC_BASE_BASE64_H_
diff --git a/base/basictypes.h b/base/basictypes.h
new file mode 100644
index 0000000..42ffa5a
--- /dev/null
+++ b/base/basictypes.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_BASICTYPES_H_
+#define WEBRTC_BASE_BASICTYPES_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/basictypes.h"
+
+#endif // WEBRTC_BASE_BASICTYPES_H_
diff --git a/base/bind.h b/base/bind.h
new file mode 100644
index 0000000..39d441f
--- /dev/null
+++ b/base/bind.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Bind() is an overloaded function that converts method calls into function
+// objects (aka functors). The method object is captured as a scoped_refptr<> if
+// possible, and as a raw pointer otherwise. Any arguments to the method are
+// captured by value. The return value of Bind is a stateful, nullary function
+// object. Care should be taken about the lifetime of objects captured by
+// Bind(); the returned functor knows nothing about the lifetime of a non
+// ref-counted method object or any arguments passed by pointer, and calling the
+// functor with a destroyed object will surely do bad things.
+//
+// To prevent the method object from being captured as a scoped_refptr<>, you
+// can use Unretained. But this should only be done when absolutely necessary,
+// and when the caller knows the extra reference isn't needed.
+//
+// Example usage:
+// struct Foo {
+// int Test1() { return 42; }
+// int Test2() const { return 52; }
+// int Test3(int x) { return x*x; }
+// float Test4(int x, float y) { return x + y; }
+// };
+//
+// int main() {
+// Foo foo;
+// cout << rtc::Bind(&Foo::Test1, &foo)() << endl;
+// cout << rtc::Bind(&Foo::Test2, &foo)() << endl;
+// cout << rtc::Bind(&Foo::Test3, &foo, 3)() << endl;
+// cout << rtc::Bind(&Foo::Test4, &foo, 7, 8.5f)() << endl;
+// }
+//
+// Example usage of ref counted objects:
+// struct Bar {
+// int AddRef();
+// int Release();
+//
+// void Test() {}
+// void BindThis() {
+// // The functor passed to AsyncInvoke() will keep this object alive.
+// invoker.AsyncInvoke(RTC_FROM_HERE,rtc::Bind(&Bar::Test, this));
+// }
+// };
+//
+// int main() {
+// rtc::scoped_refptr<Bar> 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/base/bitbuffer.h b/base/bitbuffer.h
new file mode 100644
index 0000000..09cba3c
--- /dev/null
+++ b/base/bitbuffer.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_BITBUFFER_H_
+#define WEBRTC_BASE_BITBUFFER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/bitbuffer.h"
+
+#endif // WEBRTC_BASE_BITBUFFER_H_
diff --git a/base/buffer.h b/base/buffer.h
new file mode 100644
index 0000000..92c85d9
--- /dev/null
+++ b/base/buffer.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_BUFFER_H_
+#define WEBRTC_BASE_BUFFER_H_
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/buffer.h"
+
+#endif // WEBRTC_BASE_BUFFER_H_
diff --git a/base/bufferqueue.h b/base/bufferqueue.h
new file mode 100644
index 0000000..3142ae3
--- /dev/null
+++ b/base/bufferqueue.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_BUFFERQUEUE_H_
+#define WEBRTC_BASE_BUFFERQUEUE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/bufferqueue.h"
+
+#endif // WEBRTC_BASE_BUFFERQUEUE_H_
diff --git a/base/bytebuffer.h b/base/bytebuffer.h
new file mode 100644
index 0000000..0cc9a12
--- /dev/null
+++ b/base/bytebuffer.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_BYTEBUFFER_H_
+#define WEBRTC_BASE_BYTEBUFFER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/bytebuffer.h"
+
+#endif // WEBRTC_BASE_BYTEBUFFER_H_
diff --git a/base/byteorder.h b/base/byteorder.h
new file mode 100644
index 0000000..28cbaa5
--- /dev/null
+++ b/base/byteorder.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_BYTEORDER_H_
+#define WEBRTC_BASE_BYTEORDER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/byteorder.h"
+
+#endif // WEBRTC_BASE_BYTEORDER_H_
diff --git a/base/callback.h b/base/callback.h
new file mode 100644
index 0000000..4da1e6d
--- /dev/null
+++ b/base/callback.h
@@ -0,0 +1,70 @@
+// This file was GENERATED by command:
+// pump.py callback.h.pump
+// DO NOT EDIT BY HAND!!!
+
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// To generate callback.h from callback.h.pump, execute:
+// /home/build/google3/third_party/gtest/scripts/pump.py callback.h.pump
+
+// Callbacks are callable object containers. They can hold a function pointer
+// or a function object and behave like a value type. Internally, data is
+// reference-counted, making copies and pass-by-value inexpensive.
+//
+// Callbacks are typed using template arguments. The format is:
+// CallbackN<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/base/checks.h b/base/checks.h
new file mode 100644
index 0000000..f56f157
--- /dev/null
+++ b/base/checks.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2006 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_CHECKS_H_
+#define WEBRTC_BASE_CHECKS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/checks.h"
+
+#endif // WEBRTC_BASE_CHECKS_H_
diff --git a/base/compile_assert_c.h b/base/compile_assert_c.h
new file mode 100644
index 0000000..934cc9b
--- /dev/null
+++ b/base/compile_assert_c.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_COMPILE_ASSERT_C_H_
+#define WEBRTC_BASE_COMPILE_ASSERT_C_H_
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/compile_assert_c.h"
+
+#endif // WEBRTC_BASE_COMPILE_ASSERT_C_H_
diff --git a/base/constructormagic.h b/base/constructormagic.h
new file mode 100644
index 0000000..21652c2
--- /dev/null
+++ b/base/constructormagic.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_CONSTRUCTORMAGIC_H_
+#define WEBRTC_BASE_CONSTRUCTORMAGIC_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/constructormagic.h"
+
+#endif // WEBRTC_BASE_CONSTRUCTORMAGIC_H_
diff --git a/base/copyonwritebuffer.h b/base/copyonwritebuffer.h
new file mode 100644
index 0000000..6a95b31
--- /dev/null
+++ b/base/copyonwritebuffer.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_COPYONWRITEBUFFER_H_
+#define WEBRTC_BASE_COPYONWRITEBUFFER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/copyonwritebuffer.h"
+
+#endif // WEBRTC_BASE_COPYONWRITEBUFFER_H_
diff --git a/base/cpu_time.h b/base/cpu_time.h
new file mode 100644
index 0000000..f627790
--- /dev/null
+++ b/base/cpu_time.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_CPU_TIME_H_
+#define WEBRTC_BASE_CPU_TIME_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/cpu_time.h"
+
+#endif // WEBRTC_BASE_CPU_TIME_H_
diff --git a/base/crc32.h b/base/crc32.h
new file mode 100644
index 0000000..6854567
--- /dev/null
+++ b/base/crc32.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_CRC32_H_
+#define WEBRTC_BASE_CRC32_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/crc32.h"
+
+#endif // WEBRTC_BASE_CRC32_H_
diff --git a/base/criticalsection.h b/base/criticalsection.h
new file mode 100644
index 0000000..ab3f542
--- /dev/null
+++ b/base/criticalsection.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_CRITICALSECTION_H_
+#define WEBRTC_BASE_CRITICALSECTION_H_
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/criticalsection.h"
+
+#endif // WEBRTC_BASE_CRITICALSECTION_H_
diff --git a/base/cryptstring.h b/base/cryptstring.h
new file mode 100644
index 0000000..1a474b4
--- /dev/null
+++ b/base/cryptstring.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_CRYPTSTRING_H_
+#define WEBRTC_BASE_CRYPTSTRING_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/cryptstring.h"
+
+#endif // WEBRTC_BASE_CRYPTSTRING_H_
diff --git a/base/deprecation.h b/base/deprecation.h
new file mode 100644
index 0000000..d6c5124
--- /dev/null
+++ b/base/deprecation.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_DEPRECATION_H_
+#define WEBRTC_BASE_DEPRECATION_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/deprecation.h"
+
+#endif // WEBRTC_BASE_DEPRECATION_H_
diff --git a/base/dscp.h b/base/dscp.h
new file mode 100644
index 0000000..1cf2756
--- /dev/null
+++ b/base/dscp.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_DSCP_H_
+#define WEBRTC_BASE_DSCP_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/dscp.h"
+
+#endif // WEBRTC_BASE_DSCP_H_
diff --git a/base/event.h b/base/event.h
new file mode 100644
index 0000000..28ff731
--- /dev/null
+++ b/base/event.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_EVENT_H_
+#define WEBRTC_BASE_EVENT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/event.h"
+
+#endif // WEBRTC_BASE_EVENT_H_
diff --git a/base/event_tracer.h b/base/event_tracer.h
new file mode 100644
index 0000000..b6da14a
--- /dev/null
+++ b/base/event_tracer.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file defines the interface for event tracing in WebRTC.
+//
+// Event log handlers are set through SetupEventTracer(). User of this API will
+// provide two function pointers to handle event tracing calls.
+//
+// * GetCategoryEnabledPtr
+// Event tracing system calls this function to determine if a particular
+// event category is enabled.
+//
+// * AddTraceEventPtr
+// Adds a tracing event. It is the user's responsibility to log the data
+// provided.
+//
+// Parameters for the above two functions are described in trace_event.h.
+
+#ifndef WEBRTC_BASE_EVENT_TRACER_H_
+#define WEBRTC_BASE_EVENT_TRACER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/event_tracer.h"
+
+#endif // WEBRTC_BASE_EVENT_TRACER_H_
diff --git a/base/fakeclock.h b/base/fakeclock.h
new file mode 100644
index 0000000..22d640d
--- /dev/null
+++ b/base/fakeclock.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FAKECLOCK_H_
+#define WEBRTC_BASE_FAKECLOCK_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/fakeclock.h"
+
+#endif // WEBRTC_BASE_FAKECLOCK_H_
diff --git a/base/fakenetwork.h b/base/fakenetwork.h
new file mode 100644
index 0000000..c2c8e6d
--- /dev/null
+++ b/base/fakenetwork.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2009 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FAKENETWORK_H_
+#define WEBRTC_BASE_FAKENETWORK_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/fakenetwork.h"
+
+#endif // WEBRTC_BASE_FAKENETWORK_H_
diff --git a/base/fakesslidentity.h b/base/fakesslidentity.h
new file mode 100644
index 0000000..da204b2
--- /dev/null
+++ b/base/fakesslidentity.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FAKESSLIDENTITY_H_
+#define WEBRTC_BASE_FAKESSLIDENTITY_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/fakesslidentity.h"
+
+#endif // WEBRTC_BASE_FAKESSLIDENTITY_H_
diff --git a/base/file.h b/base/file.h
new file mode 100644
index 0000000..5a4465f
--- /dev/null
+++ b/base/file.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FILE_H_
+#define WEBRTC_BASE_FILE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/file.h"
+
+#endif // WEBRTC_BASE_FILE_H_
diff --git a/base/filerotatingstream.h b/base/filerotatingstream.h
new file mode 100644
index 0000000..26306db
--- /dev/null
+++ b/base/filerotatingstream.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FILEROTATINGSTREAM_H_
+#define WEBRTC_BASE_FILEROTATINGSTREAM_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/filerotatingstream.h"
+
+#endif // WEBRTC_BASE_FILEROTATINGSTREAM_H_
diff --git a/base/fileutils.h b/base/fileutils.h
new file mode 100644
index 0000000..18de30c
--- /dev/null
+++ b/base/fileutils.h
@@ -0,0 +1,20 @@
+
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FILEUTILS_H_
+#define WEBRTC_BASE_FILEUTILS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/fileutils.h"
+
+#endif // WEBRTC_BASE_FILEUTILS_H_
diff --git a/base/firewallsocketserver.h b/base/firewallsocketserver.h
new file mode 100644
index 0000000..18ad9bc
--- /dev/null
+++ b/base/firewallsocketserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FIREWALLSOCKETSERVER_H_
+#define WEBRTC_BASE_FIREWALLSOCKETSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/firewallsocketserver.h"
+
+#endif // WEBRTC_BASE_FIREWALLSOCKETSERVER_H_
diff --git a/base/flags.h b/base/flags.h
new file mode 100644
index 0000000..9094466
--- /dev/null
+++ b/base/flags.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright 2006 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// Originally comes from shared/commandlineflags/flags.h
+
+// Flags are defined and declared using DEFINE_xxx and DECLARE_xxx macros,
+// where xxx is the flag type. Flags are referred to via FLAG_yyy,
+// where yyy is the flag name. For intialization and iteration of flags,
+// see the FlagList class. For full programmatic access to any
+// flag, see the Flag class.
+//
+// The implementation only relies and basic C++ functionality
+// and needs no special library or STL support.
+
+#ifndef WEBRTC_BASE_FLAGS_H_
+#define WEBRTC_BASE_FLAGS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/flags.h"
+
+#endif // SHARED_COMMANDLINEFLAGS_FLAGS_H_
diff --git a/base/format_macros.h b/base/format_macros.h
new file mode 100644
index 0000000..844e71e
--- /dev/null
+++ b/base/format_macros.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FORMAT_MACROS_H_
+#define WEBRTC_BASE_FORMAT_MACROS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/format_macros.h"
+
+#endif // WEBRTC_BASE_FORMAT_MACROS_H_
diff --git a/base/function_view.h b/base/function_view.h
new file mode 100644
index 0000000..1230026
--- /dev/null
+++ b/base/function_view.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_FUNCTION_VIEW_H_
+#define WEBRTC_BASE_FUNCTION_VIEW_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/function_view.h"
+
+#endif // WEBRTC_BASE_FUNCTION_VIEW_H_
diff --git a/base/gtest_prod_util.h b/base/gtest_prod_util.h
new file mode 100644
index 0000000..0c25943
--- /dev/null
+++ b/base/gtest_prod_util.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_GTEST_PROD_UTIL_H_
+#define WEBRTC_BASE_GTEST_PROD_UTIL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/gtest_prod_util.h"
+
+#endif // WEBRTC_BASE_GTEST_PROD_UTIL_H_
diff --git a/base/gunit.h b/base/gunit.h
new file mode 100644
index 0000000..d6c092e
--- /dev/null
+++ b/base/gunit.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_GUNIT_H_
+#define WEBRTC_BASE_GUNIT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/gunit.h"
+
+#endif // WEBRTC_BASE_GUNIT_H_
diff --git a/base/gunit_prod.h b/base/gunit_prod.h
new file mode 100644
index 0000000..436abee
--- /dev/null
+++ b/base/gunit_prod.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_GUNIT_PROD_H_
+#define WEBRTC_BASE_GUNIT_PROD_H_
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/gunit_prod.h"
+
+#endif // WEBRTC_BASE_GUNIT_PROD_H_
diff --git a/base/helpers.h b/base/helpers.h
new file mode 100644
index 0000000..86a388e
--- /dev/null
+++ b/base/helpers.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_HELPERS_H_
+#define WEBRTC_BASE_HELPERS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/helpers.h"
+
+#endif // WEBRTC_BASE_HELPERS_H_
diff --git a/base/httpbase.h b/base/httpbase.h
new file mode 100644
index 0000000..a66ce15
--- /dev/null
+++ b/base/httpbase.h
@@ -0,0 +1,20 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#ifndef WEBRTC_BASE_HTTPBASE_H_
+#define WEBRTC_BASE_HTTPBASE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/httpbase.h"
+
+#endif // WEBRTC_BASE_HTTPBASE_H_
diff --git a/base/httpcommon-inl.h b/base/httpcommon-inl.h
new file mode 100644
index 0000000..7dfe182
--- /dev/null
+++ b/base/httpcommon-inl.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_HTTPCOMMON_INL_H_
+#define WEBRTC_BASE_HTTPCOMMON_INL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/httpcommon-inl.h"
+
+#endif // WEBRTC_BASE_HTTPCOMMON_INL_H_
diff --git a/base/httpcommon.h b/base/httpcommon.h
new file mode 100644
index 0000000..3946dfc
--- /dev/null
+++ b/base/httpcommon.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_HTTPCOMMON_H_
+#define WEBRTC_BASE_HTTPCOMMON_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/httpcommon.h"
+
+#endif // WEBRTC_BASE_HTTPCOMMON_H_
diff --git a/base/httpserver.h b/base/httpserver.h
new file mode 100644
index 0000000..4fd75a2
--- /dev/null
+++ b/base/httpserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_HTTPSERVER_H_
+#define WEBRTC_BASE_HTTPSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/httpserver.h"
+
+#endif // WEBRTC_BASE_HTTPSERVER_H_
diff --git a/base/ifaddrs-android.h b/base/ifaddrs-android.h
new file mode 100644
index 0000000..9c49c9f
--- /dev/null
+++ b/base/ifaddrs-android.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_IFADDRS_ANDROID_H_
+#define WEBRTC_BASE_IFADDRS_ANDROID_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/ifaddrs-android.h"
+
+#endif // WEBRTC_BASE_IFADDRS_ANDROID_H_
diff --git a/base/ifaddrs_converter.h b/base/ifaddrs_converter.h
new file mode 100644
index 0000000..de7ad87
--- /dev/null
+++ b/base/ifaddrs_converter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_IFADDRS_CONVERTER_H_
+#define WEBRTC_BASE_IFADDRS_CONVERTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/ifaddrs_converter.h"
+
+#endif // WEBRTC_BASE_IFADDRS_CONVERTER_H_
diff --git a/base/ignore_wundef.h b/base/ignore_wundef.h
new file mode 100644
index 0000000..fdfba9b
--- /dev/null
+++ b/base/ignore_wundef.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_IGNORE_WUNDEF_H_
+#define WEBRTC_BASE_IGNORE_WUNDEF_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/ignore_wundef.h"
+
+#endif // WEBRTC_BASE_IGNORE_WUNDEF_H_
diff --git a/base/ipaddress.h b/base/ipaddress.h
new file mode 100644
index 0000000..44e432d
--- /dev/null
+++ b/base/ipaddress.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_IPADDRESS_H_
+#define WEBRTC_BASE_IPADDRESS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/ipaddress.h"
+
+#endif // WEBRTC_BASE_IPADDRESS_H_
diff --git a/base/json.h b/base/json.h
new file mode 100644
index 0000000..175028f
--- /dev/null
+++ b/base/json.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_JSON_H_
+#define WEBRTC_BASE_JSON_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/json.h"
+
+#endif // WEBRTC_BASE_JSON_H_
diff --git a/base/keep_ref_until_done.h b/base/keep_ref_until_done.h
new file mode 100644
index 0000000..171e048
--- /dev/null
+++ b/base/keep_ref_until_done.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
+#define WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/keep_ref_until_done.h"
+
+#endif // WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
diff --git a/base/location.h b/base/location.h
new file mode 100644
index 0000000..432471c
--- /dev/null
+++ b/base/location.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_LOCATION_H_
+#define WEBRTC_BASE_LOCATION_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/location.h"
+
+#endif // WEBRTC_BASE_LOCATION_H_
diff --git a/base/logging.h b/base/logging.h
new file mode 100644
index 0000000..594d9c9
--- /dev/null
+++ b/base/logging.h
@@ -0,0 +1,54 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// LOG(...) an ostream target that can be used to send formatted
+// output to a variety of logging targets, such as debugger console, stderr,
+// or any LogSink.
+// The severity level passed as the first argument to the LOGging
+// functions is used as a filter, to limit the verbosity of the logging.
+// Static members of LogMessage documented below are used to control the
+// verbosity and target of the output.
+// There are several variations on the LOG macro which facilitate logging
+// of common error conditions, detailed below.
+
+// LOG(sev) logs the given stream at severity "sev", which must be a
+// compile-time constant of the LoggingSeverity type, without the namespace
+// prefix.
+// LOG_V(sev) Like LOG(), but sev is a run-time variable of the LoggingSeverity
+// type (basically, it just doesn't prepend the namespace).
+// LOG_F(sev) Like LOG(), but includes the name of the current function.
+// LOG_T(sev) Like LOG(), but includes the this pointer.
+// LOG_T_F(sev) Like LOG_F(), but includes the this pointer.
+// LOG_GLE(M)(sev [, mod]) attempt to add a string description of the
+// HRESULT returned by GetLastError. The "M" variant allows searching of a
+// DLL's string table for the error description.
+// LOG_ERRNO(sev) attempts to add a string description of an errno-derived
+// error. errno and associated facilities exist on both Windows and POSIX,
+// but on Windows they only apply to the C/C++ runtime.
+// LOG_ERR(sev) is an alias for the platform's normal error system, i.e. _GLE on
+// Windows and _ERRNO on POSIX.
+// (The above three also all have _EX versions that let you specify the error
+// code, rather than using the last one.)
+// LOG_E(sev, ctx, err, ...) logs a detailed error interpreted using the
+// specified context.
+// LOG_CHECK_LEVEL(sev) (and LOG_CHECK_LEVEL_V(sev)) can be used as a test
+// before performing expensive or sensitive operations whose sole purpose is
+// to output logging data at the desired level.
+// Lastly, PLOG(sev, err) is an alias for LOG_ERR_EX.
+
+#ifndef WEBRTC_BASE_LOGGING_H_
+#define WEBRTC_BASE_LOGGING_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/logging.h"
+
+#endif // WEBRTC_BASE_LOGGING_H_
diff --git a/base/logsinks.h b/base/logsinks.h
new file mode 100644
index 0000000..95e6dc6
--- /dev/null
+++ b/base/logsinks.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_LOGSINKS_H_
+#define WEBRTC_BASE_LOGSINKS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/logsinks.h"
+
+#endif // WEBRTC_BASE_LOGSINKS_H_
diff --git a/base/macutils.h b/base/macutils.h
new file mode 100644
index 0000000..ed0c4f5
--- /dev/null
+++ b/base/macutils.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2007 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_MACUTILS_H_
+#define WEBRTC_BASE_MACUTILS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/macutils.h"
+
+#endif // WEBRTC_BASE_MACUTILS_H_
diff --git a/base/mathutils.h b/base/mathutils.h
new file mode 100644
index 0000000..9e5c3ca
--- /dev/null
+++ b/base/mathutils.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2005 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_MATHUTILS_H_
+#define WEBRTC_BASE_MATHUTILS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/mathutils.h"
+
+#endif // WEBRTC_BASE_MATHUTILS_H_
diff --git a/base/md5.h b/base/md5.h
new file mode 100644
index 0000000..fd17541
--- /dev/null
+++ b/base/md5.h
@@ -0,0 +1,31 @@
+/*
+ * This is the header file for the MD5 message-digest algorithm.
+ * The algorithm is due to Ron Rivest. This code was
+ * written by Colin Plumb in 1993, no copyright is claimed.
+ * This code is in the public domain; do with it what you wish.
+ *
+ * Equivalent code is available from RSA Data Security, Inc.
+ * This code has been tested against that, and is equivalent,
+ * except that you don't need to include two pages of legalese
+ * with every copy.
+ * To compute the message digest of a chunk of bytes, declare an
+ * MD5Context structure, pass it to MD5Init, call MD5Update as
+ * needed on buffers full of bytes, and then call MD5Final, which
+ * will fill a supplied 16-byte array with the digest.
+ *
+ */
+
+// Changes(fbarchard): Ported to C++ and Google style guide.
+// Made context first parameter in MD5Final for consistency with Sha1.
+// Changes(hellner): added rtc namespace
+// Changes(pbos): Reverted types back to uint32(8)_t with _t suffix.
+
+#ifndef WEBRTC_BASE_MD5_H_
+#define WEBRTC_BASE_MD5_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/md5.h"
+
+#endif // WEBRTC_BASE_MD5_H_
diff --git a/base/md5digest.h b/base/md5digest.h
new file mode 100644
index 0000000..66d6ee1
--- /dev/null
+++ b/base/md5digest.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_MD5DIGEST_H_
+#define WEBRTC_BASE_MD5DIGEST_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/md5digest.h"
+
+#endif // WEBRTC_BASE_MD5DIGEST_H_
diff --git a/base/memory_usage.h b/base/memory_usage.h
new file mode 100644
index 0000000..5c22559
--- /dev/null
+++ b/base/memory_usage.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_BASE_MEMORY_USAGE_H_
+#define WEBRTC_BASE_MEMORY_USAGE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/memory_usage.h"
+
+#endif // WEBRTC_BASE_MEMORY_USAGE_H_
diff --git a/base/messagedigest.h b/base/messagedigest.h
new file mode 100644
index 0000000..b73f907
--- /dev/null
+++ b/base/messagedigest.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_MESSAGEDIGEST_H_
+#define WEBRTC_BASE_MESSAGEDIGEST_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/messagedigest.h"
+
+#endif // WEBRTC_BASE_MESSAGEDIGEST_H_
diff --git a/base/messagehandler.h b/base/messagehandler.h
new file mode 100644
index 0000000..943d0d7
--- /dev/null
+++ b/base/messagehandler.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_MESSAGEHANDLER_H_
+#define WEBRTC_BASE_MESSAGEHANDLER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/messagehandler.h"
+
+#endif // WEBRTC_BASE_MESSAGEHANDLER_H_
diff --git a/base/messagequeue.h b/base/messagequeue.h
new file mode 100644
index 0000000..353a4b7
--- /dev/null
+++ b/base/messagequeue.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_MESSAGEQUEUE_H_
+#define WEBRTC_BASE_MESSAGEQUEUE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/messagequeue.h"
+
+#endif // WEBRTC_BASE_MESSAGEQUEUE_H_
diff --git a/base/mod_ops.h b/base/mod_ops.h
new file mode 100644
index 0000000..d61bd05
--- /dev/null
+++ b/base/mod_ops.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_MOD_OPS_H_
+#define WEBRTC_BASE_MOD_OPS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/mod_ops.h"
+
+#endif // WEBRTC_BASE_MOD_OPS_H_
diff --git a/base/natserver.h b/base/natserver.h
new file mode 100644
index 0000000..b803ad8
--- /dev/null
+++ b/base/natserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NATSERVER_H_
+#define WEBRTC_BASE_NATSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/natserver.h"
+
+#endif // WEBRTC_BASE_NATSERVER_H_
diff --git a/base/natsocketfactory.h b/base/natsocketfactory.h
new file mode 100644
index 0000000..31c29ab
--- /dev/null
+++ b/base/natsocketfactory.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NATSOCKETFACTORY_H_
+#define WEBRTC_BASE_NATSOCKETFACTORY_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/natsocketfactory.h"
+
+#endif // WEBRTC_BASE_NATSOCKETFACTORY_H_
diff --git a/base/nattypes.h b/base/nattypes.h
new file mode 100644
index 0000000..001f57f
--- /dev/null
+++ b/base/nattypes.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NATTYPES_H_
+#define WEBRTC_BASE_NATTYPES_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/nattypes.h"
+
+#endif // WEBRTC_BASE_NATTYPES_H_
diff --git a/base/nethelpers.h b/base/nethelpers.h
new file mode 100644
index 0000000..9a8e607
--- /dev/null
+++ b/base/nethelpers.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2008 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NETHELPERS_H_
+#define WEBRTC_BASE_NETHELPERS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/nethelpers.h"
+
+#endif // WEBRTC_BASE_NETHELPERS_H_
diff --git a/base/network.h b/base/network.h
new file mode 100644
index 0000000..2953098
--- /dev/null
+++ b/base/network.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NETWORK_H_
+#define WEBRTC_BASE_NETWORK_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/network.h"
+
+#endif // WEBRTC_BASE_NETWORK_H_
diff --git a/base/networkmonitor.h b/base/networkmonitor.h
new file mode 100644
index 0000000..290da4f
--- /dev/null
+++ b/base/networkmonitor.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NETWORKMONITOR_H_
+#define WEBRTC_BASE_NETWORKMONITOR_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/networkmonitor.h"
+
+#endif // WEBRTC_BASE_NETWORKMONITOR_H_
diff --git a/base/networkroute.h b/base/networkroute.h
new file mode 100644
index 0000000..b5e8c13
--- /dev/null
+++ b/base/networkroute.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NETWORKROUTE_H_
+#define WEBRTC_BASE_NETWORKROUTE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/networkroute.h"
+
+#endif // WEBRTC_BASE_NETWORKROUTE_H_
diff --git a/base/nullsocketserver.h b/base/nullsocketserver.h
new file mode 100644
index 0000000..214c542
--- /dev/null
+++ b/base/nullsocketserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NULLSOCKETSERVER_H_
+#define WEBRTC_BASE_NULLSOCKETSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/nullsocketserver.h"
+
+#endif // WEBRTC_BASE_NULLSOCKETSERVER_H_
diff --git a/base/numerics/exp_filter.h b/base/numerics/exp_filter.h
new file mode 100644
index 0000000..a4eaea2
--- /dev/null
+++ b/base/numerics/exp_filter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NUMERICS_EXP_FILTER_H_
+#define WEBRTC_BASE_NUMERICS_EXP_FILTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/numerics/exp_filter.h"
+
+#endif // WEBRTC_BASE_NUMERICS_EXP_FILTER_H_
diff --git a/base/numerics/percentile_filter.h b/base/numerics/percentile_filter.h
new file mode 100644
index 0000000..a9058a2
--- /dev/null
+++ b/base/numerics/percentile_filter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_
+#define WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/numerics/percentile_filter.h"
+
+#endif // WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_
diff --git a/base/onetimeevent.h b/base/onetimeevent.h
new file mode 100644
index 0000000..6849bac
--- /dev/null
+++ b/base/onetimeevent.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ONETIMEEVENT_H_
+#define WEBRTC_BASE_ONETIMEEVENT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/onetimeevent.h"
+
+#endif // WEBRTC_BASE_ONETIMEEVENT_H_
diff --git a/base/openssl.h b/base/openssl.h
new file mode 100644
index 0000000..795af70
--- /dev/null
+++ b/base/openssl.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPENSSL_H_
+#define WEBRTC_BASE_OPENSSL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/openssl.h"
+
+#endif // WEBRTC_BASE_OPENSSL_H_
diff --git a/base/openssladapter.h b/base/openssladapter.h
new file mode 100644
index 0000000..6444215
--- /dev/null
+++ b/base/openssladapter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPENSSLADAPTER_H_
+#define WEBRTC_BASE_OPENSSLADAPTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/openssladapter.h"
+
+#endif // WEBRTC_BASE_OPENSSLADAPTER_H_
diff --git a/base/openssldigest.h b/base/openssldigest.h
new file mode 100644
index 0000000..031c0b1
--- /dev/null
+++ b/base/openssldigest.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPENSSLDIGEST_H_
+#define WEBRTC_BASE_OPENSSLDIGEST_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/openssldigest.h"
+
+#endif // WEBRTC_BASE_OPENSSLDIGEST_H_
diff --git a/base/opensslidentity.h b/base/opensslidentity.h
new file mode 100644
index 0000000..59fa571
--- /dev/null
+++ b/base/opensslidentity.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPENSSLIDENTITY_H_
+#define WEBRTC_BASE_OPENSSLIDENTITY_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/opensslidentity.h"
+
+#endif // WEBRTC_BASE_OPENSSLIDENTITY_H_
diff --git a/base/opensslstreamadapter.h b/base/opensslstreamadapter.h
new file mode 100644
index 0000000..e17e029
--- /dev/null
+++ b/base/opensslstreamadapter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_
+#define WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/opensslstreamadapter.h"
+
+#endif // WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_
diff --git a/base/optional.h b/base/optional.h
new file mode 100644
index 0000000..7657ec3
--- /dev/null
+++ b/base/optional.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPTIONAL_H_
+#define WEBRTC_BASE_OPTIONAL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/optional.h"
+
+#endif // WEBRTC_BASE_OPTIONAL_H_
diff --git a/base/optionsfile.h b/base/optionsfile.h
new file mode 100644
index 0000000..e77fd8a
--- /dev/null
+++ b/base/optionsfile.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2008 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPTIONSFILE_H_
+#define WEBRTC_BASE_OPTIONSFILE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/optionsfile.h"
+
+#endif // WEBRTC_BASE_OPTIONSFILE_H_
diff --git a/base/pathutils.h b/base/pathutils.h
new file mode 100644
index 0000000..b45ca04
--- /dev/null
+++ b/base/pathutils.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PATHUTILS_H_
+#define WEBRTC_BASE_PATHUTILS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/pathutils.h"
+
+#endif // WEBRTC_BASE_PATHUTILS_H_
diff --git a/base/physicalsocketserver.h b/base/physicalsocketserver.h
new file mode 100644
index 0000000..63e6dfa
--- /dev/null
+++ b/base/physicalsocketserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PHYSICALSOCKETSERVER_H_
+#define WEBRTC_BASE_PHYSICALSOCKETSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/physicalsocketserver.h"
+
+#endif // WEBRTC_BASE_PHYSICALSOCKETSERVER_H_
diff --git a/base/platform_file.h b/base/platform_file.h
new file mode 100644
index 0000000..c7396ec
--- /dev/null
+++ b/base/platform_file.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PLATFORM_FILE_H_
+#define WEBRTC_BASE_PLATFORM_FILE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/platform_file.h"
+
+#endif // WEBRTC_BASE_PLATFORM_FILE_H_
diff --git a/base/platform_thread.h b/base/platform_thread.h
new file mode 100644
index 0000000..626d66f
--- /dev/null
+++ b/base/platform_thread.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PLATFORM_THREAD_H_
+#define WEBRTC_BASE_PLATFORM_THREAD_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/platform_thread.h"
+
+#endif // WEBRTC_BASE_PLATFORM_THREAD_H_
diff --git a/base/platform_thread_types.h b/base/platform_thread_types.h
new file mode 100644
index 0000000..f2dbd58
--- /dev/null
+++ b/base/platform_thread_types.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
+#define WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/platform_thread_types.h"
+
+#endif // WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
diff --git a/base/protobuf_utils.h b/base/protobuf_utils.h
new file mode 100644
index 0000000..3d2dd86
--- /dev/null
+++ b/base/protobuf_utils.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <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/base/proxyinfo.h b/base/proxyinfo.h
new file mode 100644
index 0000000..f0ae182
--- /dev/null
+++ b/base/proxyinfo.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PROXYINFO_H_
+#define WEBRTC_BASE_PROXYINFO_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/proxyinfo.h"
+
+#endif // WEBRTC_BASE_PROXYINFO_H_
diff --git a/base/proxyserver.h b/base/proxyserver.h
new file mode 100644
index 0000000..1bf580a
--- /dev/null
+++ b/base/proxyserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PROXYSERVER_H_
+#define WEBRTC_BASE_PROXYSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/proxyserver.h"
+
+#endif // WEBRTC_BASE_PROXYSERVER_H_
diff --git a/base/ptr_util.h b/base/ptr_util.h
new file mode 100644
index 0000000..aa6f3b4
--- /dev/null
+++ b/base/ptr_util.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This implementation is borrowed from chromium.
+
+#ifndef WEBRTC_BASE_PTR_UTIL_H_
+#define WEBRTC_BASE_PTR_UTIL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/ptr_util.h"
+
+#endif // WEBRTC_BASE_PTR_UTIL_H_
diff --git a/base/race_checker.h b/base/race_checker.h
new file mode 100644
index 0000000..474fdb5
--- /dev/null
+++ b/base/race_checker.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RACE_CHECKER_H_
+#define WEBRTC_BASE_RACE_CHECKER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/race_checker.h"
+
+#endif // WEBRTC_BASE_RACE_CHECKER_H_
diff --git a/base/random.h b/base/random.h
new file mode 100644
index 0000000..12a4902
--- /dev/null
+++ b/base/random.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RANDOM_H_
+#define WEBRTC_BASE_RANDOM_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/random.h"
+
+#endif // WEBRTC_BASE_RANDOM_H_
diff --git a/base/rate_limiter.h b/base/rate_limiter.h
new file mode 100644
index 0000000..0cba5fb
--- /dev/null
+++ b/base/rate_limiter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RATE_LIMITER_H_
+#define WEBRTC_BASE_RATE_LIMITER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/rate_limiter.h"
+
+#endif // WEBRTC_BASE_RATE_LIMITER_H_
diff --git a/base/rate_statistics.h b/base/rate_statistics.h
new file mode 100644
index 0000000..1a17500
--- /dev/null
+++ b/base/rate_statistics.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RATE_STATISTICS_H_
+#define WEBRTC_BASE_RATE_STATISTICS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/rate_statistics.h"
+
+#endif // WEBRTC_BASE_RATE_STATISTICS_H_
diff --git a/base/ratelimiter.h b/base/ratelimiter.h
new file mode 100644
index 0000000..0e372db
--- /dev/null
+++ b/base/ratelimiter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RATELIMITER_H_
+#define WEBRTC_BASE_RATELIMITER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/ratelimiter.h"
+
+#endif // WEBRTC_BASE_RATELIMITER_H_
diff --git a/base/ratetracker.h b/base/ratetracker.h
new file mode 100644
index 0000000..d1fd75d
--- /dev/null
+++ b/base/ratetracker.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RATETRACKER_H_
+#define WEBRTC_BASE_RATETRACKER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/ratetracker.h"
+
+#endif // WEBRTC_BASE_RATETRACKER_H_
diff --git a/base/refcount.h b/base/refcount.h
new file mode 100644
index 0000000..4a7cea3
--- /dev/null
+++ b/base/refcount.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_BASE_REFCOUNT_H_
+#define WEBRTC_BASE_REFCOUNT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/refcount.h"
+
+#endif // WEBRTC_BASE_REFCOUNT_H_
diff --git a/base/refcountedobject.h b/base/refcountedobject.h
new file mode 100644
index 0000000..78304fa
--- /dev/null
+++ b/base/refcountedobject.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_BASE_REFCOUNTEDOBJECT_H_
+#define WEBRTC_BASE_REFCOUNTEDOBJECT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/refcountedobject.h"
+
+#endif // WEBRTC_BASE_REFCOUNTEDOBJECT_H_
diff --git a/base/rollingaccumulator.h b/base/rollingaccumulator.h
new file mode 100644
index 0000000..a7de4f1
--- /dev/null
+++ b/base/rollingaccumulator.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ROLLINGACCUMULATOR_H_
+#define WEBRTC_BASE_ROLLINGACCUMULATOR_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/rollingaccumulator.h"
+
+#endif // WEBRTC_BASE_ROLLINGACCUMULATOR_H_
diff --git a/base/rtccertificate.h b/base/rtccertificate.h
new file mode 100644
index 0000000..22d8fe7
--- /dev/null
+++ b/base/rtccertificate.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RTCCERTIFICATE_H_
+#define WEBRTC_BASE_RTCCERTIFICATE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/rtccertificate.h"
+
+#endif // WEBRTC_BASE_RTCCERTIFICATE_H_
diff --git a/base/rtccertificategenerator.h b/base/rtccertificategenerator.h
new file mode 100644
index 0000000..fac1cec
--- /dev/null
+++ b/base/rtccertificategenerator.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_
+#define WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/rtccertificategenerator.h"
+
+#endif // WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_
diff --git a/base/safe_compare.h b/base/safe_compare.h
new file mode 100644
index 0000000..acdd9ce
--- /dev/null
+++ b/base/safe_compare.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file defines six constexpr functions:
+//
+// rtc::SafeEq // ==
+// rtc::SafeNe // !=
+// rtc::SafeLt // <
+// rtc::SafeLe // <=
+// rtc::SafeGt // >
+// rtc::SafeGe // >=
+//
+// They each accept two arguments of arbitrary types, and in almost all cases,
+// they simply call the appropriate comparison operator. However, if both
+// arguments are integers, they don't compare them using C++'s quirky rules,
+// but instead adhere to the true mathematical definitions. It is as if the
+// arguments were first converted to infinite-range signed integers, and then
+// compared, although of course nothing expensive like that actually takes
+// place. In practice, for signed/signed and unsigned/unsigned comparisons and
+// some mixed-signed comparisons with a compile-time constant, the overhead is
+// zero; in the remaining cases, it is just a few machine instructions (no
+// branches).
+
+#ifndef WEBRTC_BASE_SAFE_COMPARE_H_
+#define WEBRTC_BASE_SAFE_COMPARE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/safe_compare.h"
+
+#endif // WEBRTC_BASE_SAFE_COMPARE_H_
diff --git a/base/safe_conversions.h b/base/safe_conversions.h
new file mode 100644
index 0000000..ac0bb65
--- /dev/null
+++ b/base/safe_conversions.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Borrowed from Chromium's src/base/numerics/safe_conversions.h.
+
+#ifndef WEBRTC_BASE_SAFE_CONVERSIONS_H_
+#define WEBRTC_BASE_SAFE_CONVERSIONS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/safe_conversions.h"
+
+#endif // WEBRTC_BASE_SAFE_CONVERSIONS_H_
diff --git a/base/safe_conversions_impl.h b/base/safe_conversions_impl.h
new file mode 100644
index 0000000..497e156
--- /dev/null
+++ b/base/safe_conversions_impl.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Borrowed from Chromium's src/base/numerics/safe_conversions_impl.h.
+
+#ifndef WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_
+#define WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/safe_conversions_impl.h"
+
+#endif // WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_
diff --git a/base/safe_minmax.h b/base/safe_minmax.h
new file mode 100644
index 0000000..54d99b7
--- /dev/null
+++ b/base/safe_minmax.h
@@ -0,0 +1,18 @@
+/*
+ * Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SAFE_MINMAX_H_
+#define WEBRTC_BASE_SAFE_MINMAX_H_
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/safe_minmax.h"
+
+#endif // WEBRTC_BASE_SAFE_MINMAX_H_
diff --git a/base/sanitizer.h b/base/sanitizer.h
new file mode 100644
index 0000000..56a5e10
--- /dev/null
+++ b/base/sanitizer.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SANITIZER_H_
+#define WEBRTC_BASE_SANITIZER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sanitizer.h"
+
+#endif // WEBRTC_BASE_SANITIZER_H_
diff --git a/base/scoped_ref_ptr.h b/base/scoped_ref_ptr.h
new file mode 100644
index 0000000..2599562
--- /dev/null
+++ b/base/scoped_ref_ptr.h
@@ -0,0 +1,71 @@
+/*
+ * Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Originally these classes are from Chromium.
+// http://src.chromium.org/viewvc/chrome/trunk/src/base/memory/ref_counted.h?view=markup
+
+//
+// A smart pointer class for reference counted objects. Use this class instead
+// of calling AddRef and Release manually on a reference counted object to
+// avoid common memory leaks caused by forgetting to Release an object
+// reference. Sample usage:
+//
+// class MyFoo : public RefCounted<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/base/sequenced_task_checker.h b/base/sequenced_task_checker.h
new file mode 100644
index 0000000..e586b8d
--- /dev/null
+++ b/base/sequenced_task_checker.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_
+#define WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sequenced_task_checker.h"
+
+#endif // WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_
diff --git a/base/sequenced_task_checker_impl.h b/base/sequenced_task_checker_impl.h
new file mode 100644
index 0000000..4972539
--- /dev/null
+++ b/base/sequenced_task_checker_impl.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_
+#define WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sequenced_task_checker_impl.h"
+
+#endif // WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_
diff --git a/base/sha1.h b/base/sha1.h
new file mode 100644
index 0000000..fde3e59
--- /dev/null
+++ b/base/sha1.h
@@ -0,0 +1,18 @@
+/*
+ * 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/base/sha1digest.h b/base/sha1digest.h
new file mode 100644
index 0000000..e3b4ef8
--- /dev/null
+++ b/base/sha1digest.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SHA1DIGEST_H_
+#define WEBRTC_BASE_SHA1DIGEST_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sha1digest.h"
+
+#endif // WEBRTC_BASE_SHA1DIGEST_H_
diff --git a/base/sigslot.h b/base/sigslot.h
new file mode 100644
index 0000000..9d31441
--- /dev/null
+++ b/base/sigslot.h
@@ -0,0 +1,104 @@
+// sigslot.h: Signal/Slot classes
+//
+// Written by Sarah Thompson (sarah@telergy.com) 2002.
+//
+// License: Public domain. You are free to use this code however you like, with
+// the proviso that the author takes on no responsibility or liability for any
+// use.
+//
+// QUICK DOCUMENTATION
+//
+// (see also the full documentation at http://sigslot.sourceforge.net/)
+//
+// #define switches
+// SIGSLOT_PURE_ISO:
+// Define this to force ISO C++ compliance. This also disables all of
+// the thread safety support on platforms where it is available.
+//
+// SIGSLOT_USE_POSIX_THREADS:
+// Force use of Posix threads when using a C++ compiler other than gcc
+// on a platform that supports Posix threads. (When using gcc, this is
+// the default - use SIGSLOT_PURE_ISO to disable this if necessary)
+//
+// SIGSLOT_DEFAULT_MT_POLICY:
+// Where thread support is enabled, this defaults to
+// multi_threaded_global. Otherwise, the default is single_threaded.
+// #define this yourself to override the default. In pure ISO mode,
+// anything other than single_threaded will cause a compiler error.
+//
+// PLATFORM NOTES
+//
+// Win32:
+// On Win32, the WEBRTC_WIN symbol must be #defined. Most mainstream
+// compilers do this by default, but you may need to define it yourself
+// if your build environment is less standard. This causes the Win32
+// thread support to be compiled in and used automatically.
+//
+// Unix/Linux/BSD, etc.:
+// If you're using gcc, it is assumed that you have Posix threads
+// available, so they are used automatically. You can override this (as
+// under Windows) with the SIGSLOT_PURE_ISO switch. If you're using
+// something other than gcc but still want to use Posix threads, you
+// need to #define SIGSLOT_USE_POSIX_THREADS.
+//
+// ISO C++:
+// If none of the supported platforms are detected, or if
+// SIGSLOT_PURE_ISO is defined, all multithreading support is turned
+// off, along with any code that might cause a pure ISO C++ environment
+// to complain. Before you ask, gcc -ansi -pedantic won't compile this
+// library, but gcc -ansi is fine. Pedantic mode seems to throw a lot of
+// errors that aren't really there. If you feel like investigating this,
+// please contact the author.
+//
+//
+// THREADING MODES
+//
+// single_threaded:
+// Your program is assumed to be single threaded from the point of view
+// of signal/slot usage (i.e. all objects using signals and slots are
+// created and destroyed from a single thread). Behaviour if objects are
+// destroyed concurrently is undefined (i.e. you'll get the occasional
+// segmentation fault/memory exception).
+//
+// multi_threaded_global:
+// Your program is assumed to be multi threaded. Objects using signals
+// and slots can be safely created and destroyed from any thread, even
+// when connections exist. In multi_threaded_global mode, this is
+// achieved by a single global mutex (actually a critical section on
+// Windows because they are faster). This option uses less OS resources,
+// but results in more opportunities for contention, possibly resulting
+// in more context switches than are strictly necessary.
+//
+// multi_threaded_local:
+// Behaviour in this mode is essentially the same as
+// multi_threaded_global, except that each signal, and each object that
+// inherits has_slots, all have their own mutex/critical section. In
+// practice, this means that mutex collisions (and hence context
+// switches) only happen if they are absolutely essential. However, on
+// some platforms, creating a lot of mutexes can slow down the whole OS,
+// so use this option with care.
+//
+// USING THE LIBRARY
+//
+// See the full documentation at http://sigslot.sourceforge.net/
+//
+// Libjingle specific:
+//
+// This file has been modified such that has_slots and signalx do not have to be
+// using the same threading requirements. E.g. it is possible to connect a
+// has_slots<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/base/sigslottester.h b/base/sigslottester.h
new file mode 100644
index 0000000..545bf9e
--- /dev/null
+++ b/base/sigslottester.h
@@ -0,0 +1,23 @@
+// This file was GENERATED by command:
+// pump.py sigslottester.h.pump
+// DO NOT EDIT BY HAND!!!
+
+/*
+ * Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SIGSLOTTESTER_H_
+#define WEBRTC_BASE_SIGSLOTTESTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sigslottester.h"
+
+#endif // WEBRTC_BASE_SIGSLOTTESTER_H_
diff --git a/base/socket.h b/base/socket.h
new file mode 100644
index 0000000..19ea7a0
--- /dev/null
+++ b/base/socket.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SOCKET_H_
+#define WEBRTC_BASE_SOCKET_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/socket.h"
+
+#endif // WEBRTC_BASE_SOCKET_H_
diff --git a/base/socket_unittest.h b/base/socket_unittest.h
new file mode 100644
index 0000000..f6769f9
--- /dev/null
+++ b/base/socket_unittest.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2009 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SOCKET_UNITTEST_H_
+#define WEBRTC_BASE_SOCKET_UNITTEST_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/socket_unittest.h"
+
+#endif // WEBRTC_BASE_SOCKET_UNITTEST_H_
diff --git a/base/socketadapters.h b/base/socketadapters.h
new file mode 100644
index 0000000..7df0f3a
--- /dev/null
+++ b/base/socketadapters.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SOCKETADAPTERS_H_
+#define WEBRTC_BASE_SOCKETADAPTERS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/socketadapters.h"
+
+#endif // WEBRTC_BASE_SOCKETADAPTERS_H_
diff --git a/base/socketaddress.h b/base/socketaddress.h
new file mode 100644
index 0000000..20199ad
--- /dev/null
+++ b/base/socketaddress.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SOCKETADDRESS_H_
+#define WEBRTC_BASE_SOCKETADDRESS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/socketaddress.h"
+
+#endif // WEBRTC_BASE_SOCKETADDRESS_H_
diff --git a/base/socketaddresspair.h b/base/socketaddresspair.h
new file mode 100644
index 0000000..3f53f10
--- /dev/null
+++ b/base/socketaddresspair.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SOCKETADDRESSPAIR_H_
+#define WEBRTC_BASE_SOCKETADDRESSPAIR_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/socketaddresspair.h"
+
+#endif // WEBRTC_BASE_SOCKETADDRESSPAIR_H_
diff --git a/base/socketfactory.h b/base/socketfactory.h
new file mode 100644
index 0000000..3a829ac
--- /dev/null
+++ b/base/socketfactory.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SOCKETFACTORY_H_
+#define WEBRTC_BASE_SOCKETFACTORY_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/socketfactory.h"
+
+#endif // WEBRTC_BASE_SOCKETFACTORY_H_
diff --git a/base/socketserver.h b/base/socketserver.h
new file mode 100644
index 0000000..55b427d
--- /dev/null
+++ b/base/socketserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SOCKETSERVER_H_
+#define WEBRTC_BASE_SOCKETSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/socketserver.h"
+
+#endif // WEBRTC_BASE_SOCKETSERVER_H_
diff --git a/base/socketstream.h b/base/socketstream.h
new file mode 100644
index 0000000..a76ffb3
--- /dev/null
+++ b/base/socketstream.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2005 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SOCKETSTREAM_H_
+#define WEBRTC_BASE_SOCKETSTREAM_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/socketstream.h"
+
+#endif // WEBRTC_BASE_SOCKETSTREAM_H_
diff --git a/base/ssladapter.h b/base/ssladapter.h
new file mode 100644
index 0000000..3d432ec
--- /dev/null
+++ b/base/ssladapter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SSLADAPTER_H_
+#define WEBRTC_BASE_SSLADAPTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/ssladapter.h"
+
+#endif // WEBRTC_BASE_SSLADAPTER_H_
diff --git a/base/sslfingerprint.h b/base/sslfingerprint.h
new file mode 100644
index 0000000..6be82fd
--- /dev/null
+++ b/base/sslfingerprint.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SSLFINGERPRINT_H_
+#define WEBRTC_BASE_SSLFINGERPRINT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sslfingerprint.h"
+
+#endif // WEBRTC_BASE_SSLFINGERPRINT_H_
diff --git a/base/sslidentity.h b/base/sslidentity.h
new file mode 100644
index 0000000..1cedfa0
--- /dev/null
+++ b/base/sslidentity.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Handling of certificates and keypairs for SSLStreamAdapter's peer mode.
+
+#ifndef WEBRTC_BASE_SSLIDENTITY_H_
+#define WEBRTC_BASE_SSLIDENTITY_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sslidentity.h"
+
+#endif // WEBRTC_BASE_SSLIDENTITY_H_
diff --git a/base/sslroots.h b/base/sslroots.h
new file mode 100644
index 0000000..9fa706b
--- /dev/null
+++ b/base/sslroots.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SSLROOTS_H_
+#define WEBRTC_BASE_SSLROOTS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sslroots.h"
+
+#endif // WEBRTC_BASE_SSLROOTS_H_
diff --git a/base/sslstreamadapter.h b/base/sslstreamadapter.h
new file mode 100644
index 0000000..d7c062e
--- /dev/null
+++ b/base/sslstreamadapter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SSLSTREAMADAPTER_H_
+#define WEBRTC_BASE_SSLSTREAMADAPTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/sslstreamadapter.h"
+
+#endif // WEBRTC_BASE_SSLSTREAMADAPTER_H_
diff --git a/base/stream.h b/base/stream.h
new file mode 100644
index 0000000..18dd865
--- /dev/null
+++ b/base/stream.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_STREAM_H_
+#define WEBRTC_BASE_STREAM_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/stream.h"
+
+#endif // WEBRTC_BASE_STREAM_H_
diff --git a/base/string_to_number.h b/base/string_to_number.h
new file mode 100644
index 0000000..fa88ba4
--- /dev/null
+++ b/base/string_to_number.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_STRING_TO_NUMBER_H_
+#define WEBRTC_BASE_STRING_TO_NUMBER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/string_to_number.h"
+
+#endif // WEBRTC_BASE_STRING_TO_NUMBER_H_
diff --git a/base/stringencode.h b/base/stringencode.h
new file mode 100644
index 0000000..27b810e
--- /dev/null
+++ b/base/stringencode.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_STRINGENCODE_H_
+#define WEBRTC_BASE_STRINGENCODE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/stringencode.h"
+
+#endif // WEBRTC_BASE_STRINGENCODE_H__
diff --git a/base/stringize_macros.h b/base/stringize_macros.h
new file mode 100644
index 0000000..5f8a5b1
--- /dev/null
+++ b/base/stringize_macros.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Modified from the Chromium original:
+// src/base/strings/stringize_macros.h
+
+// This file defines preprocessor macros for stringizing preprocessor
+// symbols (or their output) and manipulating preprocessor symbols
+// that define strings.
+
+#ifndef WEBRTC_BASE_STRINGIZE_MACROS_H_
+#define WEBRTC_BASE_STRINGIZE_MACROS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/stringize_macros.h"
+
+#endif // WEBRTC_BASE_STRINGIZE_MACROS_H_
diff --git a/base/stringutils.h b/base/stringutils.h
new file mode 100644
index 0000000..e3b5e07
--- /dev/null
+++ b/base/stringutils.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_STRINGUTILS_H_
+#define WEBRTC_BASE_STRINGUTILS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/stringutils.h"
+
+#endif // WEBRTC_BASE_STRINGUTILS_H_
diff --git a/base/swap_queue.h b/base/swap_queue.h
new file mode 100644
index 0000000..7111147
--- /dev/null
+++ b/base/swap_queue.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_SWAP_QUEUE_H_
+#define WEBRTC_BASE_SWAP_QUEUE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/swap_queue.h"
+
+#endif // WEBRTC_BASE_SWAP_QUEUE_H_
diff --git a/base/task_queue.h b/base/task_queue.h
new file mode 100644
index 0000000..12f5cbb
--- /dev/null
+++ b/base/task_queue.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TASK_QUEUE_H_
+#define WEBRTC_BASE_TASK_QUEUE_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/task_queue.h"
+
+#endif // WEBRTC_BASE_TASK_QUEUE_H_
diff --git a/base/task_queue_posix.h b/base/task_queue_posix.h
new file mode 100644
index 0000000..6cb51a0
--- /dev/null
+++ b/base/task_queue_posix.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TASK_QUEUE_POSIX_H_
+#define WEBRTC_BASE_TASK_QUEUE_POSIX_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/task_queue_posix.h"
+
+#endif // WEBRTC_BASE_TASK_QUEUE_POSIX_H_
diff --git a/base/template_util.h b/base/template_util.h
new file mode 100644
index 0000000..9a05643
--- /dev/null
+++ b/base/template_util.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Borrowed from Chromium's src/base/template_util.h.
+
+#ifndef WEBRTC_BASE_TEMPLATE_UTIL_H_
+#define WEBRTC_BASE_TEMPLATE_UTIL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/template_util.h"
+
+#endif // WEBRTC_BASE_TEMPLATE_UTIL_H_
diff --git a/base/testbase64.h b/base/testbase64.h
new file mode 100644
index 0000000..fc9846f
--- /dev/null
+++ b/base/testbase64.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TESTBASE64_H_
+#define WEBRTC_BASE_TESTBASE64_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/testbase64.h"
+
+#endif // WEBRTC_BASE_TESTBASE64_H_
diff --git a/base/testclient.h b/base/testclient.h
new file mode 100644
index 0000000..378e2b8
--- /dev/null
+++ b/base/testclient.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TESTCLIENT_H_
+#define WEBRTC_BASE_TESTCLIENT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/testclient.h"
+
+#endif // WEBRTC_BASE_TESTCLIENT_H_
diff --git a/base/testechoserver.h b/base/testechoserver.h
new file mode 100644
index 0000000..21365e2
--- /dev/null
+++ b/base/testechoserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TESTECHOSERVER_H_
+#define WEBRTC_BASE_TESTECHOSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/testechoserver.h"
+
+#endif // WEBRTC_BASE_TESTECHOSERVER_H_
diff --git a/base/testutils.h b/base/testutils.h
new file mode 100644
index 0000000..74f2160
--- /dev/null
+++ b/base/testutils.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TESTUTILS_H_
+#define WEBRTC_BASE_TESTUTILS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/testutils.h"
+
+#endif // WEBRTC_BASE_TESTUTILS_H_
diff --git a/base/thread.h b/base/thread.h
new file mode 100644
index 0000000..6a6887a
--- /dev/null
+++ b/base/thread.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_THREAD_H_
+#define WEBRTC_BASE_THREAD_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/thread.h"
+
+#endif // WEBRTC_BASE_THREAD_H_
diff --git a/base/thread_annotations.h b/base/thread_annotations.h
new file mode 100644
index 0000000..5b94ffe
--- /dev/null
+++ b/base/thread_annotations.h
@@ -0,0 +1,27 @@
+//
+// Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+//
+// Use of this source code is governed by a BSD-style license
+// that can be found in the LICENSE file in the root of the source
+// tree. An additional intellectual property rights grant can be found
+// in the file PATENTS. All contributing project authors may
+// be found in the AUTHORS file in the root of the source tree.
+//
+// Borrowed from
+// https://code.google.com/p/gperftools/source/browse/src/base/thread_annotations.h
+// but adapted for clang attributes instead of the gcc.
+//
+// This header file contains the macro definitions for thread safety
+// annotations that allow the developers to document the locking policies
+// of their multi-threaded code. The annotations can also help program
+// analysis tools to identify potential thread safety issues.
+
+#ifndef WEBRTC_BASE_THREAD_ANNOTATIONS_H_
+#define WEBRTC_BASE_THREAD_ANNOTATIONS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/thread_annotations.h"
+
+#endif // WEBRTC_BASE_THREAD_ANNOTATIONS_H_
diff --git a/base/thread_checker.h b/base/thread_checker.h
new file mode 100644
index 0000000..ade5256
--- /dev/null
+++ b/base/thread_checker.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Borrowed from Chromium's src/base/threading/thread_checker.h.
+
+#ifndef WEBRTC_BASE_THREAD_CHECKER_H_
+#define WEBRTC_BASE_THREAD_CHECKER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/thread_checker.h"
+
+#endif // WEBRTC_BASE_THREAD_CHECKER_H_
diff --git a/base/thread_checker_impl.h b/base/thread_checker_impl.h
new file mode 100644
index 0000000..3a0a6c7
--- /dev/null
+++ b/base/thread_checker_impl.h
@@ -0,0 +1,21 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Borrowed from Chromium's src/base/threading/thread_checker_impl.h.
+
+#ifndef WEBRTC_BASE_THREAD_CHECKER_IMPL_H_
+#define WEBRTC_BASE_THREAD_CHECKER_IMPL_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/thread_checker_impl.h"
+
+#endif // WEBRTC_BASE_THREAD_CHECKER_IMPL_H_
diff --git a/base/timedelta.h b/base/timedelta.h
new file mode 100644
index 0000000..f2e98a8
--- /dev/null
+++ b/base/timedelta.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TIMEDELTA_H_
+#define WEBRTC_BASE_TIMEDELTA_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/timedelta.h"
+
+#endif // WEBRTC_BASE_TIMEDELTA_H_
diff --git a/base/timestampaligner.h b/base/timestampaligner.h
new file mode 100644
index 0000000..60c3631
--- /dev/null
+++ b/base/timestampaligner.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright (c) 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TIMESTAMPALIGNER_H_
+#define WEBRTC_BASE_TIMESTAMPALIGNER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/timestampaligner.h"
+
+#endif // WEBRTC_BASE_TIMESTAMPALIGNER_H_
diff --git a/base/timeutils.h b/base/timeutils.h
new file mode 100644
index 0000000..1569b58
--- /dev/null
+++ b/base/timeutils.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2005 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TIMEUTILS_H_
+#define WEBRTC_BASE_TIMEUTILS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/timeutils.h"
+
+#endif // WEBRTC_BASE_TIMEUTILS_H_
diff --git a/base/trace_event.h b/base/trace_event.h
new file mode 100644
index 0000000..1bea5f4
--- /dev/null
+++ b/base/trace_event.h
@@ -0,0 +1,14 @@
+// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file under third_party_mods/chromium or at:
+// http://src.chromium.org/svn/trunk/src/LICENSE
+
+#ifndef WEBRTC_BASE_TRACE_EVENT_H_
+#define WEBRTC_BASE_TRACE_EVENT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/trace_event.h"
+
+#endif // WEBRTC_BASE_TRACE_EVENT_H_
diff --git a/base/transformadapter.h b/base/transformadapter.h
new file mode 100644
index 0000000..3d9c86b
--- /dev/null
+++ b/base/transformadapter.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TRANSFORMADAPTER_H_
+#define WEBRTC_BASE_TRANSFORMADAPTER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/transformadapter.h"
+
+#endif // WEBRTC_BASE_TRANSFORMADAPTER_H_
diff --git a/base/type_traits.h b/base/type_traits.h
new file mode 100644
index 0000000..6a4ac8d
--- /dev/null
+++ b/base/type_traits.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TYPE_TRAITS_H_
+#define WEBRTC_BASE_TYPE_TRAITS_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/type_traits.h"
+
+#endif // WEBRTC_BASE_TYPE_TRAITS_H_
diff --git a/base/unixfilesystem.h b/base/unixfilesystem.h
new file mode 100644
index 0000000..7a18205
--- /dev/null
+++ b/base/unixfilesystem.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_UNIXFILESYSTEM_H_
+#define WEBRTC_BASE_UNIXFILESYSTEM_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/unixfilesystem.h"
+
+#endif // WEBRTC_BASE_UNIXFILESYSTEM_H_
diff --git a/base/virtualsocketserver.h b/base/virtualsocketserver.h
new file mode 100644
index 0000000..31ce96d
--- /dev/null
+++ b/base/virtualsocketserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_VIRTUALSOCKETSERVER_H_
+#define WEBRTC_BASE_VIRTUALSOCKETSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/virtualsocketserver.h"
+
+#endif // WEBRTC_BASE_VIRTUALSOCKETSERVER_H_
diff --git a/base/weak_ptr.h b/base/weak_ptr.h
new file mode 100644
index 0000000..282a551
--- /dev/null
+++ b/base/weak_ptr.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_WEAK_PTR_H_
+#define WEBRTC_BASE_WEAK_PTR_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/weak_ptr.h"
+
+#endif // WEBRTC_BASE_WEAK_PTR_H_
diff --git a/base/win32.h b/base/win32.h
new file mode 100644
index 0000000..413bd11
--- /dev/null
+++ b/base/win32.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_WIN32_H_
+#define WEBRTC_BASE_WIN32_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/win32.h"
+
+#endif // WEBRTC_BASE_WIN32_H_
diff --git a/base/win32filesystem.h b/base/win32filesystem.h
new file mode 100644
index 0000000..d647c44
--- /dev/null
+++ b/base/win32filesystem.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_WIN32FILESYSTEM_H_
+#define WEBRTC_BASE_WIN32FILESYSTEM_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/win32filesystem.h"
+
+#endif // WEBRTC_BASE_WIN32FILESYSTEM_H_
diff --git a/base/win32socketinit.h b/base/win32socketinit.h
new file mode 100644
index 0000000..d7017e1
--- /dev/null
+++ b/base/win32socketinit.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2009 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_WIN32SOCKETINIT_H_
+#define WEBRTC_BASE_WIN32SOCKETINIT_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/win32socketinit.h"
+
+#endif // WEBRTC_BASE_WIN32SOCKETINIT_H_
diff --git a/base/win32socketserver.h b/base/win32socketserver.h
new file mode 100644
index 0000000..c143692
--- /dev/null
+++ b/base/win32socketserver.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_WIN32SOCKETSERVER_H_
+#define WEBRTC_BASE_WIN32SOCKETSERVER_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/win32socketserver.h"
+
+#endif // WEBRTC_BASE_WIN32SOCKETSERVER_H_
diff --git a/base/win32window.h b/base/win32window.h
new file mode 100644
index 0000000..ffffdf9
--- /dev/null
+++ b/base/win32window.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_WIN32WINDOW_H_
+#define WEBRTC_BASE_WIN32WINDOW_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/win32window.h"
+
+#endif // WEBRTC_BASE_WIN32WINDOW_H_
diff --git a/base/window.h b/base/window.h
new file mode 100644
index 0000000..d515f7c
--- /dev/null
+++ b/base/window.h
@@ -0,0 +1,19 @@
+/*
+ * Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_WINDOW_H_
+#define WEBRTC_BASE_WINDOW_H_
+
+
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/window.h"
+
+#endif // WEBRTC_BASE_WINDOW_H_
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 2d18530..6a50ea0 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -28,8 +28,8 @@
"../api:libjingle_peerconnection_api",
"../api:transport_api",
"../api/audio_codecs:audio_codecs_api",
- "../rtc_base:rtc_base",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
]
}
@@ -43,7 +43,7 @@
"rtp_transport_controller_send_interface.h",
]
deps = [
- "../rtc_base:rtc_base_approved",
+ "../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",
]
}
@@ -109,6 +109,7 @@
"..:webrtc_common",
"../api:transport_api",
"../audio",
+ "../base:rtc_task_queue",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl",
"../modules/bitrate_controller",
@@ -116,7 +117,6 @@
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/utility",
- "../rtc_base:rtc_task_queue",
"../system_wrappers",
"../video",
]
@@ -149,6 +149,7 @@
":rtp_sender",
"..:webrtc_common",
"../api:mock_audio_mixer",
+ "../base:rtc_base_approved",
"../logging:rtc_event_log_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
@@ -157,7 +158,6 @@
"../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/common_audio/BUILD.gn b/common_audio/BUILD.gn
index f7f3efb..ff0aa26 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -63,8 +63,8 @@
deps = [
":sinc_resampler",
"..:webrtc_common",
- "../rtc_base:gtest_prod",
- "../rtc_base:rtc_base_approved",
+ "../base:gtest_prod",
+ "../base:rtc_base_approved",
"../system_wrappers",
]
public_deps = [
@@ -209,8 +209,8 @@
":common_audio_c_arm_asm",
":common_audio_cc",
"..:webrtc_common",
- "../rtc_base:compile_assert_c",
- "../rtc_base:rtc_base_approved",
+ "../base:compile_assert_c",
+ "../base:rtc_base_approved",
"../system_wrappers:system_wrappers",
]
}
@@ -225,7 +225,7 @@
public_configs = [ ":common_audio_config" ]
deps = [
"..:webrtc_common",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"../system_wrappers:system_wrappers",
]
}
@@ -236,8 +236,8 @@
]
deps = [
"..:webrtc_common",
- "../rtc_base:gtest_prod",
- "../rtc_base:rtc_base_approved",
+ "../base:gtest_prod",
+ "../base:rtc_base_approved",
"../system_wrappers",
]
}
@@ -344,7 +344,7 @@
}
deps = [
":common_audio_c",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
}
}
@@ -401,8 +401,8 @@
":common_audio",
":sinc_resampler",
"..:webrtc_common",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_base_tests_utils",
+ "../base:rtc_base_approved",
+ "../base:rtc_base_tests_utils",
"../system_wrappers",
"../test:test_main",
"//testing/gmock",
diff --git a/common_video/BUILD.gn b/common_video/BUILD.gn
index 20953c8..68b4934 100644
--- a/common_video/BUILD.gn
+++ b/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/examples/BUILD.gn b/examples/BUILD.gn
index 55b72dd..85813fa 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -422,7 +422,7 @@
"objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm",
]
deps = [
- "//webrtc/rtc_base:rtc_base",
+ "//webrtc/base:rtc_base",
]
public_deps = [
":AppRTCMobile_ios_frameworks",
@@ -524,12 +524,12 @@
"//third_party/libyuv",
"//webrtc/api:libjingle_peerconnection_test_api",
"//webrtc/api:video_frame_api",
+ "//webrtc/base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
+ "//webrtc/base:rtc_json",
"//webrtc/media:rtc_media",
"//webrtc/modules/video_capture:video_capture_module",
"//webrtc/pc:libjingle_peerconnection",
- "//webrtc/rtc_base:rtc_base",
- "//webrtc/rtc_base:rtc_base_approved",
- "//webrtc/rtc_base:rtc_json",
"//webrtc/system_wrappers:field_trial_default",
"//webrtc/system_wrappers:metrics_default",
]
@@ -548,7 +548,7 @@
]
deps = [
"//webrtc:webrtc_common",
- "//webrtc/rtc_base:rtc_base_approved",
+ "//webrtc/base:rtc_base_approved",
"//webrtc/rtc_tools:command_line_parser",
]
if (!build_with_chromium && is_clang) {
@@ -562,10 +562,10 @@
"relayserver/relayserver_main.cc",
]
deps = [
- "../rtc_base:rtc_base",
+ "../base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
"//webrtc/p2p:rtc_p2p",
"//webrtc/pc:rtc_pc",
- "//webrtc/rtc_base:rtc_base_approved",
"//webrtc/system_wrappers:field_trial_default",
"//webrtc/system_wrappers:metrics_default",
]
@@ -580,10 +580,10 @@
"turnserver/turnserver_main.cc",
]
deps = [
- "../rtc_base:rtc_base",
+ "../base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
"//webrtc/p2p:rtc_p2p",
"//webrtc/pc:rtc_pc",
- "//webrtc/rtc_base:rtc_base_approved",
"//webrtc/system_wrappers:field_trial_default",
"//webrtc/system_wrappers:metrics_default",
]
@@ -598,10 +598,10 @@
"stunserver/stunserver_main.cc",
]
deps = [
- "../rtc_base:rtc_base",
+ "../base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
"//webrtc/p2p:rtc_p2p",
"//webrtc/pc:rtc_pc",
- "//webrtc/rtc_base:rtc_base_approved",
"//webrtc/system_wrappers:field_trial_default",
"//webrtc/system_wrappers:metrics_default",
]
@@ -633,13 +633,13 @@
deps = [
"//webrtc/api:libjingle_peerconnection_test_api",
"//webrtc/api:video_frame_api",
+ "//webrtc/base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
+ "//webrtc/base:rtc_json",
"//webrtc/media:rtc_media",
"//webrtc/media:rtc_media_base",
"//webrtc/modules/video_capture:video_capture_module",
"//webrtc/pc:libjingle_peerconnection",
- "//webrtc/rtc_base:rtc_base",
- "//webrtc/rtc_base:rtc_base_approved",
- "//webrtc/rtc_base:rtc_json",
"//webrtc/system_wrappers:field_trial_default",
"//webrtc/system_wrappers:metrics_default",
]
@@ -661,10 +661,10 @@
}
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/logging/BUILD.gn b/logging/BUILD.gn
index 6a70324..f3c3469 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -30,7 +30,7 @@
deps = [
"..:video_stream_api",
"..:webrtc_common",
- "../rtc_base:rtc_base_approved",
+ "../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",
- "../rtc_base:protobuf_utils",
- "../rtc_base:rtc_base_approved",
+ "../base:protobuf_utils",
+ "../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",
- "../rtc_base:rtc_base_approved",
+ "../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",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"//third_party/gflags",
]
if (!build_with_chromium && is_clang) {
diff --git a/media/BUILD.gn b/media/BUILD.gn
index ef9b79c..fad410b 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -45,8 +45,8 @@
deps = [
"..:webrtc_common",
- "../rtc_base:rtc_base",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base",
+ "../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) {
@@ -227,6 +227,10 @@
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video_codecs:video_codecs_api",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
+ "../base:rtc_task_queue",
+ "../base:sequenced_task_checker",
"../call",
"../common_video:common_video",
"../modules/audio_coding:rent_a_codec",
@@ -241,10 +245,6 @@
"../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",
]
@@ -508,6 +508,10 @@
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../audio",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
+ "../base:rtc_base_tests_main",
+ "../base:rtc_base_tests_utils",
"../call:call_interfaces",
"../common_video:common_video",
"../logging:rtc_event_log_api",
@@ -517,10 +521,6 @@
"../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/modules/BUILD.gn b/modules/BUILD.gn
index 0ae5041..2586831 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -37,7 +37,7 @@
deps = [
"..:webrtc_common",
"../api:video_frame_api",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index f751963..d5b669c 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -47,7 +47,7 @@
deps = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
]
}
@@ -58,8 +58,8 @@
]
deps = [
"../..:webrtc_common",
- "../../rtc_base:protobuf_utils",
- "../../rtc_base:rtc_base_approved",
+ "../../base:protobuf_utils",
+ "../../base:rtc_base_approved",
"../../api/audio_codecs:audio_codecs_api",
] + audio_codec_deps
defines = audio_codec_defines
@@ -72,8 +72,8 @@
]
deps = [
"../..:webrtc_common",
- "../../rtc_base:protobuf_utils",
- "../../rtc_base:rtc_base_approved",
+ "../../base:protobuf_utils",
+ "../../base:rtc_base_approved",
"../../api/audio_codecs:audio_codecs_api",
] + audio_codec_deps
defines = audio_codec_defines
@@ -89,8 +89,8 @@
deps = [
"../../api/audio_codecs:audio_codecs_api",
"../..:webrtc_common",
- "../../rtc_base:protobuf_utils",
- "../../rtc_base:rtc_base_approved",
+ "../../base:protobuf_utils",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
":audio_coding_module_typedefs",
":isac_common",
@@ -156,7 +156,7 @@
":audio_coding_module_typedefs",
":neteq",
":rent_a_codec",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../logging:rtc_event_log_api",
]
defines = audio_coding_defines
@@ -169,7 +169,7 @@
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
]
@@ -1342,8 +1342,8 @@
":neteq_test_tools",
":webrtc_opus",
"../..:webrtc_common",
- "../../rtc_base:protobuf_utils",
- "../../rtc_base:rtc_base_approved",
+ "../../base:protobuf_utils",
+ "../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
]
@@ -1369,7 +1369,7 @@
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
":neteq_tools",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_support",
"//testing/gtest",
]
@@ -1388,7 +1388,7 @@
":audio_coding",
":neteq_tools",
"../../api/audio_codecs:audio_codecs_api",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_support",
"//testing/gtest",
]
@@ -1412,7 +1412,7 @@
":audio_format_conversion",
"..:module_api",
"../../:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@@ -1442,7 +1442,7 @@
":audio_format_conversion",
"..:module_api",
"../../:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@@ -1489,8 +1489,8 @@
":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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_base_approved",
+ "../../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 = [
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:rtp_test_utils",
"//testing/gtest",
@@ -1774,7 +1774,7 @@
testonly = true
deps = [
"../..:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_support",
"//testing/gtest",
]
@@ -1853,7 +1853,7 @@
":neteq_quality_test_support",
":neteq_tools",
"../..:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_main",
"//testing/gtest",
@@ -1872,7 +1872,7 @@
":isac_fix",
":neteq",
":neteq_quality_test_support",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_main",
"//testing/gtest",
"//third_party/gflags",
@@ -1890,7 +1890,7 @@
":g711",
":neteq",
":neteq_quality_test_support",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_main",
"//testing/gtest",
"//third_party/gflags",
@@ -1950,7 +1950,7 @@
deps = [
":isac",
":isac_test_util",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
]
configs += [ ":isac_test_warnings_config" ]
@@ -1991,7 +1991,7 @@
deps = [
":isac",
":isac_test_util",
- "../../rtc_base:rtc_base_approved",
+ "../../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/modules/audio_conference_mixer/BUILD.gn b/modules/audio_conference_mixer/BUILD.gn
index 56b2019..8939da2 100644
--- a/modules/audio_conference_mixer/BUILD.gn
+++ b/modules/audio_conference_mixer/BUILD.gn
@@ -42,7 +42,7 @@
"..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../audio_processing",
]
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index a60087e..c565165 100644
--- a/modules/audio_device/BUILD.gn
+++ b/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 = [
- "../../rtc_base:gtest_prod",
- "../../rtc_base:rtc_base",
+ "../../base:gtest_prod",
+ "../../base:rtc_base",
"../../sdk:objc_audio",
"../../sdk:objc_common",
]
@@ -281,7 +281,7 @@
deps = [
":audio_device",
":mock_audio_device",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
"../utility:utility",
@@ -331,7 +331,7 @@
deps = [
":audio_device",
"../..:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_main",
"../../test:test_support",
diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn
index 0410e1f..cd3b768 100644
--- a/modules/audio_mixer/BUILD.gn
+++ b/modules/audio_mixer/BUILD.gn
@@ -41,7 +41,7 @@
"..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../audio_processing",
]
@@ -61,7 +61,7 @@
deps = [
"..:module_api",
"../../audio/utility",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
]
}
@@ -90,8 +90,8 @@
"..:module_api",
"../../api:audio_mixer_api",
"../../audio/utility:audio_frame_operations",
- "../../rtc_base:rtc_base",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base",
+ "../../base:rtc_base_approved",
"../../test:test_support",
"//testing/gmock",
]
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index fe83596..3af019e 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -238,8 +238,8 @@
"..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
- "../../rtc_base:gtest_prod",
- "../../rtc_base:protobuf_utils",
+ "../../base:gtest_prod",
+ "../../base:protobuf_utils",
"../audio_coding:isac",
]
public_deps = [
@@ -303,8 +303,8 @@
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 = [
- "../../rtc_base:rtc_base_approved",
+ "../../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 = [
- "../../rtc_base:rtc_base_approved",
+ "../../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",
- "../../rtc_base:rtc_task_queue",
+ "../../base:rtc_task_queue",
"aec_dump",
"aec_dump:aec_dump_unittests",
]
@@ -696,7 +696,7 @@
deps = [
":audio_processing",
":audioproc_test_utils",
- "../../rtc_base:protobuf_utils",
+ "../../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",
- "../../rtc_base:protobuf_utils",
- "../../rtc_base:rtc_base_approved",
+ "../../base:protobuf_utils",
+ "../../base:rtc_base_approved",
]
}
}
diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn
index 818a9bf..950dd68 100644
--- a/modules/audio_processing/aec_dump/BUILD.gn
+++ b/modules/audio_processing/aec_dump/BUILD.gn
@@ -18,7 +18,7 @@
]
deps = [
- "../../../rtc_base:rtc_base_approved",
+ "../../../base:rtc_base_approved",
]
}
@@ -49,7 +49,7 @@
deps = [
":mock_aec_dump",
"..:audio_processing",
- "../../../rtc_base:rtc_base_approved",
+ "../../../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/modules/audio_processing/test/conversational_speech/BUILD.gn b/modules/audio_processing/test/conversational_speech/BUILD.gn
index 587663b..af24f8a 100644
--- a/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -22,8 +22,8 @@
]
deps = [
":lib",
- "../../../../rtc_base:rtc_base_approved",
- "../../../../test:test_support",
+ "../../../../../webrtc/base:rtc_base_approved",
+ "../../../../../webrtc/test:test_support",
"//third_party/gflags",
]
}
@@ -45,9 +45,9 @@
"wavreader_interface.h",
]
deps = [
- "../../../..:webrtc_common",
- "../../../../common_audio",
- "../../../../rtc_base:rtc_base_approved",
+ "../../../../../webrtc:webrtc_common",
+ "../../../../../webrtc/base:rtc_base_approved",
+ "../../../../../webrtc/common_audio",
]
visibility = [ ":*" ] # Only targets in this file can depend on this.
}
@@ -63,11 +63,14 @@
]
deps = [
":lib",
- "../../../..:webrtc_common",
- "../../../../common_audio",
- "../../../../rtc_base:rtc_base_approved",
- "../../../../test:test_support",
+ "../../../../../webrtc:webrtc_common",
+ "../../../../../webrtc/base:rtc_base_approved",
+ "../../../../../webrtc/common_audio",
+ "../../../../../webrtc/test:test_support",
"//testing/gmock",
"//testing/gtest",
+ "//webrtc:webrtc_common",
+ "//webrtc/base:rtc_base_approved",
+ "//webrtc/test:test_support",
]
}
diff --git a/modules/audio_processing/test/py_quality_assessment/BUILD.gn b/modules/audio_processing/test/py_quality_assessment/BUILD.gn
index 74d5eee..154219b 100644
--- a/modules/audio_processing/test/py_quality_assessment/BUILD.gn
+++ b/modules/audio_processing/test/py_quality_assessment/BUILD.gn
@@ -105,7 +105,7 @@
output_name = "py_quality_assessment/quality_assessment/fake_polqa"
deps = [
"//webrtc:webrtc_common",
- "//webrtc/rtc_base:rtc_base_approved",
+ "//webrtc/base:rtc_base_approved",
]
}
diff --git a/modules/bitrate_controller/BUILD.gn b/modules/bitrate_controller/BUILD.gn
index f31025b..33a2886 100644
--- a/modules/bitrate_controller/BUILD.gn
+++ b/modules/bitrate_controller/BUILD.gn
@@ -37,7 +37,7 @@
}
deps = [
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../rtp_rtcp",
]
diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn
index 0212372..ef45297 100644
--- a/modules/congestion_controller/BUILD.gn
+++ b/modules/congestion_controller/BUILD.gn
@@ -49,10 +49,10 @@
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",
@@ -88,8 +88,8 @@
deps = [
":congestion_controller",
":mock_congestion_controller",
- "../../rtc_base:rtc_base",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base",
+ "../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:field_trial",
"../../test:test_support",
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index 47b186d..93ceb75 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -28,7 +28,7 @@
deps = [
"../..:webrtc_common",
- "../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
+ "../../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
]
}
@@ -49,8 +49,8 @@
":desktop_capture_mock",
":primitives",
":screen_drawer",
- "../../rtc_base:rtc_base",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
"../../test:video_test_support",
@@ -94,7 +94,7 @@
":desktop_capture_mock",
":primitives",
"../..:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
"//testing/gmock",
@@ -131,7 +131,7 @@
deps = [
":primitives",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
]
}
@@ -155,7 +155,7 @@
deps = [
":primitives",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_support",
]
}
@@ -290,7 +290,7 @@
deps = [
":primitives",
"../..:webrtc_common",
- "../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
+ "../../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
"../../system_wrappers",
"//third_party/libyuv",
]
diff --git a/modules/media_file/BUILD.gn b/modules/media_file/BUILD.gn
index 989305c..7ab897f 100644
--- a/modules/media_file/BUILD.gn
+++ b/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/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn
index 3d1d495..57126d7 100644
--- a/modules/pacing/BUILD.gn
+++ b/modules/pacing/BUILD.gn
@@ -28,8 +28,8 @@
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",
@@ -55,8 +55,8 @@
]
deps = [
":pacing",
- "../../rtc_base:rtc_base_approved",
- "../../rtc_base:rtc_base_tests_utils",
+ "../../base:rtc_base_approved",
+ "../../base:rtc_base_tests_utils",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
"../rtp_rtcp",
diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
index 1ecf630..5a3afc6 100644
--- a/modules/remote_bitrate_estimator/BUILD.gn
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -51,8 +51,8 @@
deps = [
"../..:webrtc_common",
- "../../rtc_base:rtc_base",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
]
}
@@ -117,9 +117,9 @@
":remote_bitrate_estimator",
"..:module_api",
"../..:webrtc_common",
- "../../rtc_base:gtest_prod",
- "../../rtc_base:rtc_base",
- "../../rtc_base:rtc_base_approved",
+ "../../base:gtest_prod",
+ "../../base:rtc_base",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
"../../voice_engine",
@@ -147,7 +147,7 @@
deps = [
":bwe_simulator_lib",
":remote_bitrate_estimator",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_support",
]
if (!build_with_chromium && is_clang) {
@@ -185,8 +185,8 @@
":mock_remote_bitrate_observer",
":remote_bitrate_estimator",
"../..:webrtc_common",
- "../../rtc_base:rtc_base",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base",
+ "../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:field_trial",
"../../test:test_support",
@@ -227,7 +227,7 @@
":bwe_simulator_lib",
":remote_bitrate_estimator",
"../..:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_main",
"//testing/gmock",
"//testing/gtest",
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 3b0bc35..dc623ce 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/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",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
]
# TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
@@ -221,7 +221,7 @@
deps = [
":rtp_rtcp",
"..:module_api",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_support",
]
}
@@ -256,7 +256,7 @@
]
deps = [
":rtp_rtcp",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../test:test_support",
]
if (!build_with_chromium && is_clang) {
@@ -342,8 +342,8 @@
"..: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/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
index e98b30d..7123890 100644
--- a/modules/utility/BUILD.gn
+++ b/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",
- "../../rtc_base:rtc_task_queue",
+ "../../base:rtc_task_queue",
"../../test:test_support",
"//testing/gmock",
]
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index b150dff..5865688 100644
--- a/modules/video_capture/BUILD.gn
+++ b/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",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
]
@@ -91,7 +91,7 @@
deps = [
":video_capture_module",
- "../../rtc_base:rtc_base_approved",
+ "../../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/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 94cb133..67c8ccd 100644
--- a/modules/video_coding/BUILD.gn
+++ b/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",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers:field_trial_default",
"../../system_wrappers:metrics_default",
"../../system_wrappers:system_wrappers",
@@ -354,8 +354,8 @@
":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 += [
- "../../rtc_base:rtc_base_approved",
+ "../../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 += [
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
# TODO(brandtr): Figure out if the java dep below could be moved into
# :video_coding_videoprocessor_integration_test, where it belongs.
@@ -575,10 +575,10 @@
"../..: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/modules/video_processing/BUILD.gn b/modules/video_processing/BUILD.gn
index 6afc5f7..c4c9c3b 100644
--- a/modules/video_processing/BUILD.gn
+++ b/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",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers",
]
diff --git a/ortc/BUILD.gn b/ortc/BUILD.gn
index 40c0b2a..b6a2cc9 100644
--- a/ortc/BUILD.gn
+++ b/ortc/BUILD.gn
@@ -35,6 +35,8 @@
deps = [
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
"../media:rtc_media",
@@ -43,8 +45,6 @@
"../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/p2p/BUILD.gn b/p2p/BUILD.gn
index 5963c92..f7d5905 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -87,7 +87,7 @@
deps = [
"../api:libjingle_peerconnection_api",
"../api:ortc_api",
- "../rtc_base:rtc_base",
+ "../base:rtc_base",
"../system_wrappers:field_trial_api",
]
@@ -155,9 +155,9 @@
deps = [
":rtc_p2p",
"../api:ortc_api",
- "../rtc_base:rtc_base",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_base_tests_utils",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
+ "../base:rtc_base_tests_utils",
"../test:test_support",
"//testing/gmock",
]
@@ -209,9 +209,9 @@
":rtc_p2p",
"../api:fakemetricsobserver",
"../api:ortc_api",
- "../rtc_base:rtc_base",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_base_tests_utils",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
+ "../base:rtc_base_tests_utils",
"../test:test_support",
"//testing/gmock",
"//testing/gtest",
@@ -238,7 +238,7 @@
deps = [
":rtc_p2p",
"..:webrtc_common",
- "../rtc_base:rtc_base",
+ "../base:rtc_base",
]
}
@@ -259,8 +259,8 @@
":libstunprober",
":p2p_test_utils",
":rtc_p2p",
- "../rtc_base:rtc_base",
- "../rtc_base:rtc_base_tests_utils",
+ "../base:rtc_base",
+ "../base:rtc_base_tests_utils",
"//testing/gmock",
"//testing/gtest",
]
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 768a25c..2ff1a0a 100644
--- a/pc/BUILD.gn
+++ b/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/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 98accaf..9e90582 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -203,8 +203,8 @@
# Dependency on chromium's logging (in //base).
deps += [ "//base:base" ]
sources += [
- "../../webrtc_overrides/webrtc/rtc_base/logging.cc",
- "../../webrtc_overrides/webrtc/rtc_base/logging.h",
+ "../../webrtc_overrides/webrtc/base/logging.cc",
+ "../../webrtc_overrides/webrtc/base/logging.h",
]
} else {
sources += [
@@ -301,8 +301,8 @@
if (build_with_chromium) {
sources = [
- "../../webrtc_overrides/webrtc/rtc_base/task_queue.cc",
- "../../webrtc_overrides/webrtc/rtc_base/task_queue.h",
+ "../../webrtc_overrides/webrtc/base/task_queue.cc",
+ "../../webrtc_overrides/webrtc/base/task_queue.h",
]
} else {
sources = [
@@ -517,7 +517,7 @@
if (build_with_chromium) {
if (is_win) {
- sources += [ "../../webrtc_overrides/webrtc/rtc_base/win32socketinit.cc" ]
+ sources += [ "../../webrtc_overrides/webrtc/base/win32socketinit.cc" ]
}
include_dirs = [ "../../boringssl/src/include" ]
public_configs += [ ":rtc_base_chromium_config" ]
diff --git a/rtc_base/callback.h.pump b/rtc_base/callback.h.pump
index cceddf7..2389952 100644
--- a/rtc_base/callback.h.pump
+++ b/rtc_base/callback.h.pump
@@ -57,8 +57,8 @@
#ifndef WEBRTC_RTC_BASE_CALLBACK_H_
#define WEBRTC_RTC_BASE_CALLBACK_H_
-#include "webrtc/rtc_base/refcount.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
namespace rtc {
diff --git a/rtc_base/sigslottester.h.pump b/rtc_base/sigslottester.h.pump
index 381b791..a88f0c6 100755
--- a/rtc_base/sigslottester.h.pump
+++ b/rtc_base/sigslottester.h.pump
@@ -35,8 +35,8 @@
// EXPECT_EQ("hello", capture);
// /* See unit-tests for more examples */
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/sigslot.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/sigslot.h"
namespace rtc {
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 21a4aa8..e224380 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -48,8 +48,8 @@
"simple_command_line_parser.h",
]
deps = [
- "../rtc_base:gtest_prod",
- "../rtc_base:rtc_base_approved",
+ "../base:gtest_prod",
+ "../base:rtc_base_approved",
]
}
@@ -206,13 +206,13 @@
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",
- "../rtc_base:rtc_base_approved",
+ "../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/rtc_tools/network_tester/BUILD.gn b/rtc_tools/network_tester/BUILD.gn
index 49a625d..bd069d6 100644
--- a/rtc_tools/network_tester/BUILD.gn
+++ b/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) {
@@ -84,7 +84,7 @@
deps = [
":network_tester",
"//testing/gtest",
- "//webrtc/rtc_base:rtc_base_tests_utils",
+ "//webrtc/base:rtc_base_tests_utils",
"//webrtc/test:test_support",
]
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index b2b396b..0a15fc5 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -63,7 +63,7 @@
]
deps = [
- "../rtc_base:rtc_base",
+ "../base:rtc_base",
]
configs += [ "..:common_objc" ]
@@ -98,7 +98,7 @@
deps = [
":objc_common",
- "../rtc_base:rtc_base_approved",
+ "../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",
- "../rtc_base:rtc_base_approved",
+ "../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) {
@@ -334,12 +334,12 @@
":objc_videotracksource",
"../api:video_frame_api",
"../api/video_codecs:video_codecs_api",
+ "../base:rtc_base",
"../media:rtc_audio_video",
"../media:rtc_media_base",
"../modules:module_api",
"../pc:create_pc_factory",
"../pc:peerconnection",
- "../rtc_base:rtc_base",
"../system_wrappers:field_trial_api",
]
}
@@ -371,7 +371,7 @@
deps = [
":objc_peerconnectionfactory_base",
"../api:libjingle_peerconnection_api",
- "../rtc_base:rtc_base",
+ "../base:rtc_base",
]
}
@@ -484,11 +484,11 @@
":objc_corevideoframebuffer",
":objc_videotracksource",
"../api:video_frame_api",
+ "../base:rtc_base",
"../common_video",
"../media:rtc_media_base",
"../modules:module_api",
"../pc:peerconnection",
- "../rtc_base:rtc_base",
]
}
@@ -530,7 +530,7 @@
deps = [
":objc_peerconnection",
"..//system_wrappers:system_wrappers_default",
- "../rtc_base:rtc_base_tests_utils",
+ "../base:rtc_base_tests_utils",
"../system_wrappers:system_wrappers_default",
"//third_party/ocmock",
]
@@ -632,7 +632,7 @@
":objc_audio",
":objc_peerconnection",
":objc_ui",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
]
@@ -673,8 +673,8 @@
]
deps = [
+ "../base:rtc_base_approved",
"../common_video",
- "../rtc_base:rtc_base_approved",
]
if (!build_with_chromium && is_clang) {
@@ -705,13 +705,13 @@
":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/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index d9cf609..689d6cf 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -49,8 +49,8 @@
deps = [
"//webrtc/api:libjingle_peerconnection_api",
- "//webrtc/rtc_base:rtc_base",
- "//webrtc/rtc_base:rtc_base_approved",
+ "//webrtc/base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
"//webrtc/system_wrappers:metrics_api",
]
@@ -139,16 +139,16 @@
"//webrtc/api:libjingle_peerconnection_api",
"//webrtc/api:video_frame_api",
"//webrtc/api/video_codecs:video_codecs_api",
+ "//webrtc/base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
+ "//webrtc/base:rtc_task_queue",
+ "//webrtc/base:sequenced_task_checker",
+ "//webrtc/base:weak_ptr",
"//webrtc/common_video:common_video",
"//webrtc/media:rtc_audio_video",
"//webrtc/media:rtc_media_base",
"//webrtc/modules/utility:utility",
"//webrtc/modules/video_coding:video_coding_utility",
- "//webrtc/rtc_base:rtc_base",
- "//webrtc/rtc_base:rtc_base_approved",
- "//webrtc/rtc_base:rtc_task_queue",
- "//webrtc/rtc_base:sequenced_task_checker",
- "//webrtc/rtc_base:weak_ptr",
"//webrtc/system_wrappers:system_wrappers",
]
@@ -237,13 +237,13 @@
deps = [
":base_jni",
"../..:webrtc_common",
+ "//webrtc/base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
+ "//webrtc/base:rtc_task_queue",
"//webrtc/media:rtc_data",
"//webrtc/media:rtc_media_base",
"//webrtc/modules/utility:utility",
"//webrtc/pc:peerconnection",
- "//webrtc/rtc_base:rtc_base",
- "//webrtc/rtc_base:rtc_base_approved",
- "//webrtc/rtc_base:rtc_task_queue",
"//webrtc/system_wrappers:system_wrappers",
]
}
@@ -294,9 +294,9 @@
":null_media_jni",
":null_video_jni",
":peerconnection_jni",
+ "//webrtc/base:rtc_base",
+ "//webrtc/base:rtc_base_approved",
"//webrtc/pc:peerconnection",
- "//webrtc/rtc_base:rtc_base",
- "//webrtc/rtc_base:rtc_base_approved",
]
output_extension = "so"
}
@@ -312,8 +312,8 @@
deps = [
":libjingle_peerconnection_jni",
":libjingle_peerconnection_metrics_default_jni",
+ "//webrtc/base:rtc_base",
"//webrtc/pc:libjingle_peerconnection",
- "//webrtc/rtc_base:rtc_base",
]
output_extension = "so"
}
diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm
index 5d6b6c6..4131a45 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm
+++ b/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm
@@ -17,7 +17,7 @@
#import "WebRTC/RTCVideoFrame.h"
#import "WebRTC/RTCVideoFrameBuffer.h"
-#include "webrtc/rtc_base/timeutils.h"
+#include "webrtc/base/timeutils.h"
#include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
#include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.h"
#include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.h"
diff --git a/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm b/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm
index c4d9bd1..61c8032 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm
+++ b/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm
@@ -18,11 +18,11 @@
#import "WebRTC/RTCVideoFrameBuffer.h"
#include "webrtc/api/video_codecs/video_decoder.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/modules/video_coding/include/video_error_codes.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/timeutils.h"
#include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
namespace webrtc {
diff --git a/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm b/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm
index a063237..e63c527 100644
--- a/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm
+++ b/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm
@@ -19,11 +19,11 @@
#include "webrtc/api/video/video_frame.h"
#include "webrtc/api/video_codecs/video_encoder.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/modules/video_coding/include/video_error_codes.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/timeutils.h"
#include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h"
#include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
diff --git a/sdk/objc/Framework/UnitTests/RTCTracingTest.mm b/sdk/objc/Framework/UnitTests/RTCTracingTest.mm
index 49cc812..ec3e226 100644
--- a/sdk/objc/Framework/UnitTests/RTCTracingTest.mm
+++ b/sdk/objc/Framework/UnitTests/RTCTracingTest.mm
@@ -12,7 +12,7 @@
#include <vector>
-#include "webrtc/rtc_base/gunit.h"
+#include "webrtc/base/gunit.h"
#import "NSString+StdString.h"
#import "WebRTC/RTCTracing.h"
diff --git a/stats/BUILD.gn b/stats/BUILD.gn
index eaa6f5d..4a2f578 100644
--- a/stats/BUILD.gn
+++ b/stats/BUILD.gn
@@ -24,7 +24,7 @@
deps = [
"../api:rtc_stats_api",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
}
@@ -58,9 +58,9 @@
":rtc_stats",
":rtc_stats_test_utils",
"../api:rtc_stats_api",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_base_tests_main",
- "../rtc_base:rtc_base_tests_utils",
+ "../base:rtc_base_approved",
+ "../base:rtc_base_tests_main",
+ "../base:rtc_base_tests_utils",
"../system_wrappers:metrics_default",
"//testing/gmock",
]
diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn
index 7dfcff7..1cf1b6f 100644
--- a/system_wrappers/BUILD.gn
+++ b/system_wrappers/BUILD.gn
@@ -107,10 +107,10 @@
cflags = [ "/wd4334" ] # Ignore warning on shift operator promotion.
- # Windows needs //webrtc/rtc_base:rtc_base due to include of
- # webrtc/rtc_base/win32.h in source/clock.cc.
+ # Windows needs //webrtc/base:rtc_base due to include of webrtc/base/win32.h
+ # in source/clock.cc.
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
- deps += [ "../rtc_base:rtc_base" ]
+ deps += [ "../base:rtc_base" ]
}
if (is_win && is_clang) {
@@ -118,7 +118,7 @@
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
- deps += [ "../rtc_base:rtc_base_approved" ]
+ deps += [ "../base:rtc_base_approved" ]
}
rtc_source_set("cpu_features_api") {
@@ -148,7 +148,7 @@
]
deps = [
"..:webrtc_common",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
}
@@ -169,7 +169,7 @@
]
deps = [
":metrics_api",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
}
@@ -228,7 +228,7 @@
":metrics_default",
":system_wrappers",
"..:webrtc_common",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"../test:test_main",
"//testing/gtest",
]
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 29d02e7..f54a622 100644
--- a/test/BUILD.gn
+++ b/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",
- "../rtc_base:rtc_base_approved",
+ "../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",
- "../rtc_base:rtc_base_approved",
+ "../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",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
if (is_ios) {
sources += [ "testsupport/iosfileutils.mm" ]
deps += [ "../sdk:objc_common" ]
}
if (is_win) {
- deps += [ "../rtc_base:rtc_base" ]
+ deps += [ "../base:rtc_base" ]
}
visibility = [ ":*" ]
}
@@ -375,7 +375,7 @@
deps = [
":fileutils",
":test_support",
- "../rtc_base:rtc_base_approved",
+ "../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",
]
}
@@ -478,6 +478,9 @@
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../audio",
+ "../base:rtc_base_approved",
+ "../base:rtc_task_queue",
+ "../base:sequenced_task_checker",
"../call",
"../common_video",
"../logging:rtc_event_log_api",
@@ -489,9 +492,6 @@
"../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",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"//testing/gmock",
]
}
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index b206c72..3e68470 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -15,7 +15,7 @@
"webrtc_fuzzer_main.cc",
]
deps = [
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
"../../system_wrappers:field_trial_default",
"../../system_wrappers:metrics_default",
"//testing/libfuzzer:libfuzzer_main",
@@ -95,8 +95,8 @@
"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/video/BUILD.gn b/video/BUILD.gn
index 529a73c..5faf280 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -58,6 +58,11 @@
"..:webrtc_common",
"../api:transport_api",
"../api/video_codecs:video_codecs_api",
+ "../base:rtc_base_approved",
+ "../base:rtc_numerics",
+ "../base:rtc_task_queue",
+ "../base:sequenced_task_checker",
+ "../base:weak_ptr",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_video",
@@ -74,11 +79,6 @@
"../modules/video_coding:video_coding_utility",
"../modules/video_coding:webrtc_vp8",
"../modules/video_processing",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_numerics",
- "../rtc_base:rtc_task_queue",
- "../rtc_base:sequenced_task_checker",
- "../rtc_base:weak_ptr",
"../system_wrappers",
"../voice_engine",
]
@@ -93,6 +93,8 @@
"video_quality_test.h",
]
deps = [
+ "../base:rtc_base_tests_utils",
+ "../base:rtc_task_queue",
"../call:call_interfaces",
"../common_video",
"../logging:rtc_event_log_api",
@@ -103,8 +105,6 @@
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8",
"../modules/video_coding:webrtc_vp9",
- "../rtc_base:rtc_base_tests_utils",
- "../rtc_base:rtc_task_queue",
"../system_wrappers",
"../test:test_common",
"../test:test_support",
@@ -155,7 +155,7 @@
]
deps = [
":video_quality_test",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"../system_wrappers:metrics_default",
"../test:field_trial",
"../test:run_test",
@@ -180,7 +180,7 @@
deps = [
":video_quality_test",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"../system_wrappers:metrics_default",
"../test:field_trial",
"../test:run_test",
@@ -203,11 +203,11 @@
deps = [
"..:webrtc_common",
"../api/video_codecs:video_codecs_api",
+ "../base:rtc_base_approved",
"../call:call_interfaces",
"../common_video",
"../logging:rtc_event_log_api",
"../modules/rtp_rtcp",
- "../rtc_base:rtc_base_approved",
"../system_wrappers",
"../system_wrappers:metrics_default",
"../test:field_trial",
@@ -260,6 +260,8 @@
"..:video_stream_api",
"../api:video_frame_api",
"../api/video_codecs:video_codecs_api",
+ "../base:rtc_base_approved",
+ "../base:rtc_base_tests_utils",
"../call:call_interfaces",
"../call:rtp_receiver",
"../common_video",
@@ -277,8 +279,6 @@
"../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/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 78c92f6..e16b176 100644
--- a/voice_engine/BUILD.gn
+++ b/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",
]
@@ -143,6 +143,8 @@
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../audio/utility:audio_frame_operations",
+ "../base:rtc_base_approved",
+ "../base:rtc_task_queue",
"../call:rtp_interfaces",
"../common_audio",
"../logging:rtc_event_log_api",
@@ -157,8 +159,6 @@
"../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",
]
}
@@ -182,9 +182,9 @@
deps = [
":file_player",
":voice_engine",
+ "../base:rtc_base_approved",
+ "../base:rtc_base_tests_utils",
"../modules:module_api",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_base_tests_utils",
"../test:test_common",
"//testing/gmock",
"//testing/gtest",
@@ -247,11 +247,11 @@
deps = [
":voice_engine",
"..:webrtc_common",
+ "../base:rtc_base_approved",
"../modules:module_api",
"../modules/audio_device:audio_device",
"../modules/audio_processing:audio_processing",
"../modules/rtp_rtcp:rtp_rtcp",
- "../rtc_base:rtc_base_approved",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
