From 9483b49bafc681a8360dff7217e7651a74dea71d Mon Sep 17 00:00:00 2001 From: ehmaldonado Date: Mon, 10 Jul 2017 04:50:54 -0700 Subject: [PATCH] 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} --- PRESUBMIT.py | 3 +- webrtc/BUILD.gn | 25 ++-- webrtc/api/BUILD.gn | 16 +-- webrtc/api/audio_codecs/BUILD.gn | 6 +- webrtc/api/audio_codecs/g722/BUILD.gn | 4 +- webrtc/api/audio_codecs/ilbc/BUILD.gn | 4 +- webrtc/api/audio_codecs/opus/BUILD.gn | 8 +- webrtc/api/audio_codecs/test/BUILD.gn | 4 +- webrtc/api/video_codecs/BUILD.gn | 2 +- webrtc/audio/BUILD.gn | 8 +- webrtc/audio/utility/BUILD.gn | 4 +- webrtc/base/BUILD.gn | 135 ------------------ webrtc/base/Dummy.java | 9 -- webrtc/base/array_view.h | 19 --- webrtc/base/arraysize.h | 19 --- webrtc/base/asyncinvoker-inl.h | 19 --- webrtc/base/asyncinvoker.h | 19 --- webrtc/base/asyncpacketsocket.h | 19 --- webrtc/base/asyncresolverinterface.h | 19 --- webrtc/base/asyncsocket.h | 19 --- webrtc/base/asynctcpsocket.h | 19 --- webrtc/base/asyncudpsocket.h | 19 --- webrtc/base/atomicops.h | 19 --- webrtc/base/base64.h | 20 --- webrtc/base/basictypes.h | 19 --- webrtc/base/bind.h | 69 --------- webrtc/base/bitbuffer.h | 19 --- webrtc/base/buffer.h | 18 --- webrtc/base/bufferqueue.h | 19 --- webrtc/base/bytebuffer.h | 19 --- webrtc/base/byteorder.h | 19 --- webrtc/base/callback.h | 70 --------- webrtc/base/checks.h | 19 --- webrtc/base/compile_assert_c.h | 18 --- webrtc/base/constructormagic.h | 19 --- webrtc/base/copyonwritebuffer.h | 19 --- webrtc/base/cpu_time.h | 19 --- webrtc/base/crc32.h | 19 --- webrtc/base/criticalsection.h | 18 --- webrtc/base/cryptstring.h | 19 --- webrtc/base/deprecation.h | 19 --- webrtc/base/dscp.h | 19 --- webrtc/base/event.h | 19 --- webrtc/base/event_tracer.h | 34 ----- webrtc/base/fakeclock.h | 19 --- webrtc/base/fakenetwork.h | 19 --- webrtc/base/fakesslidentity.h | 19 --- webrtc/base/file.h | 19 --- webrtc/base/filerotatingstream.h | 19 --- webrtc/base/fileutils.h | 20 --- webrtc/base/firewallsocketserver.h | 19 --- webrtc/base/flags.h | 31 ---- webrtc/base/format_macros.h | 19 --- webrtc/base/function_view.h | 19 --- webrtc/base/gtest_prod_util.h | 19 --- webrtc/base/gunit.h | 19 --- webrtc/base/gunit_prod.h | 18 --- webrtc/base/helpers.h | 19 --- webrtc/base/httpbase.h | 20 --- webrtc/base/httpcommon-inl.h | 19 --- webrtc/base/httpcommon.h | 19 --- webrtc/base/httpserver.h | 19 --- webrtc/base/ifaddrs-android.h | 19 --- webrtc/base/ifaddrs_converter.h | 19 --- webrtc/base/ignore_wundef.h | 19 --- webrtc/base/ipaddress.h | 19 --- webrtc/base/json.h | 19 --- webrtc/base/keep_ref_until_done.h | 19 --- webrtc/base/location.h | 19 --- webrtc/base/logging.h | 54 ------- webrtc/base/logsinks.h | 19 --- webrtc/base/macutils.h | 19 --- webrtc/base/mathutils.h | 19 --- webrtc/base/md5.h | 31 ---- webrtc/base/md5digest.h | 19 --- webrtc/base/memory_usage.h | 18 --- webrtc/base/messagedigest.h | 19 --- webrtc/base/messagehandler.h | 19 --- webrtc/base/messagequeue.h | 19 --- webrtc/base/mod_ops.h | 19 --- webrtc/base/natserver.h | 19 --- webrtc/base/natsocketfactory.h | 19 --- webrtc/base/nattypes.h | 19 --- webrtc/base/nethelpers.h | 19 --- webrtc/base/network.h | 19 --- webrtc/base/networkmonitor.h | 19 --- webrtc/base/networkroute.h | 19 --- webrtc/base/nullsocketserver.h | 19 --- webrtc/base/numerics/exp_filter.h | 19 --- webrtc/base/numerics/percentile_filter.h | 19 --- webrtc/base/onetimeevent.h | 19 --- webrtc/base/openssl.h | 19 --- webrtc/base/openssladapter.h | 19 --- webrtc/base/openssldigest.h | 19 --- webrtc/base/opensslidentity.h | 19 --- webrtc/base/opensslstreamadapter.h | 19 --- webrtc/base/optional.h | 19 --- webrtc/base/optionsfile.h | 19 --- webrtc/base/pathutils.h | 19 --- webrtc/base/physicalsocketserver.h | 19 --- webrtc/base/platform_file.h | 19 --- webrtc/base/platform_thread.h | 19 --- webrtc/base/platform_thread_types.h | 19 --- webrtc/base/protobuf_utils.h | 21 --- webrtc/base/proxyinfo.h | 19 --- webrtc/base/proxyserver.h | 19 --- webrtc/base/ptr_util.h | 21 --- webrtc/base/race_checker.h | 19 --- webrtc/base/random.h | 19 --- webrtc/base/rate_limiter.h | 19 --- webrtc/base/rate_statistics.h | 19 --- webrtc/base/ratelimiter.h | 19 --- webrtc/base/ratetracker.h | 19 --- webrtc/base/refcount.h | 18 --- webrtc/base/refcountedobject.h | 18 --- webrtc/base/rollingaccumulator.h | 19 --- webrtc/base/rtccertificate.h | 19 --- webrtc/base/rtccertificategenerator.h | 19 --- webrtc/base/safe_compare.h | 39 ----- webrtc/base/safe_conversions.h | 21 --- webrtc/base/safe_conversions_impl.h | 21 --- webrtc/base/safe_minmax.h | 18 --- webrtc/base/sanitizer.h | 19 --- webrtc/base/scoped_ref_ptr.h | 71 --------- webrtc/base/sequenced_task_checker.h | 19 --- webrtc/base/sequenced_task_checker_impl.h | 19 --- webrtc/base/sha1.h | 18 --- webrtc/base/sha1digest.h | 19 --- webrtc/base/sigslot.h | 104 -------------- webrtc/base/sigslottester.h | 23 --- webrtc/base/socket.h | 19 --- webrtc/base/socket_unittest.h | 19 --- webrtc/base/socketadapters.h | 19 --- webrtc/base/socketaddress.h | 19 --- webrtc/base/socketaddresspair.h | 19 --- webrtc/base/socketfactory.h | 19 --- webrtc/base/socketserver.h | 19 --- webrtc/base/socketstream.h | 19 --- webrtc/base/ssladapter.h | 19 --- webrtc/base/sslfingerprint.h | 19 --- webrtc/base/sslidentity.h | 21 --- webrtc/base/sslroots.h | 19 --- webrtc/base/sslstreamadapter.h | 19 --- webrtc/base/stream.h | 19 --- webrtc/base/string_to_number.h | 19 --- webrtc/base/stringencode.h | 19 --- webrtc/base/stringize_macros.h | 26 ---- webrtc/base/stringutils.h | 19 --- webrtc/base/swap_queue.h | 19 --- webrtc/base/task_queue.h | 19 --- webrtc/base/task_queue_posix.h | 19 --- webrtc/base/template_util.h | 21 --- webrtc/base/testbase64.h | 19 --- webrtc/base/testclient.h | 19 --- webrtc/base/testechoserver.h | 19 --- webrtc/base/testutils.h | 19 --- webrtc/base/thread.h | 19 --- webrtc/base/thread_annotations.h | 27 ---- webrtc/base/thread_checker.h | 21 --- webrtc/base/thread_checker_impl.h | 21 --- webrtc/base/timedelta.h | 19 --- webrtc/base/timestampaligner.h | 19 --- webrtc/base/timeutils.h | 19 --- webrtc/base/trace_event.h | 14 -- webrtc/base/transformadapter.h | 19 --- webrtc/base/type_traits.h | 19 --- webrtc/base/unixfilesystem.h | 19 --- webrtc/base/virtualsocketserver.h | 19 --- webrtc/base/weak_ptr.h | 19 --- webrtc/base/win32.h | 19 --- webrtc/base/win32filesystem.h | 19 --- webrtc/base/win32socketinit.h | 19 --- webrtc/base/win32socketserver.h | 19 --- webrtc/base/win32window.h | 19 --- webrtc/base/window.h | 19 --- webrtc/call/BUILD.gn | 16 +-- webrtc/common_audio/BUILD.gn | 20 +-- webrtc/common_video/BUILD.gn | 8 +- webrtc/examples/BUILD.gn | 32 ++--- webrtc/logging/BUILD.gn | 20 +-- webrtc/media/BUILD.gn | 34 ++--- webrtc/modules/BUILD.gn | 2 +- webrtc/modules/audio_coding/BUILD.gn | 122 ++++++++-------- .../modules/audio_conference_mixer/BUILD.gn | 2 +- webrtc/modules/audio_device/BUILD.gn | 12 +- webrtc/modules/audio_mixer/BUILD.gn | 8 +- webrtc/modules/audio_processing/BUILD.gn | 42 +++--- .../audio_processing/aec_dump/BUILD.gn | 12 +- .../test/conversational_speech/BUILD.gn | 21 ++- .../test/py_quality_assessment/BUILD.gn | 2 +- webrtc/modules/bitrate_controller/BUILD.gn | 2 +- webrtc/modules/congestion_controller/BUILD.gn | 10 +- webrtc/modules/desktop_capture/BUILD.gn | 14 +- webrtc/modules/media_file/BUILD.gn | 2 +- webrtc/modules/pacing/BUILD.gn | 6 +- .../modules/remote_bitrate_estimator/BUILD.gn | 18 +-- webrtc/modules/rtp_rtcp/BUILD.gn | 14 +- webrtc/modules/utility/BUILD.gn | 4 +- webrtc/modules/video_capture/BUILD.gn | 8 +- webrtc/modules/video_coding/BUILD.gn | 48 +++---- webrtc/modules/video_processing/BUILD.gn | 4 +- webrtc/ortc/BUILD.gn | 12 +- webrtc/p2p/BUILD.gn | 20 +-- webrtc/pc/BUILD.gn | 30 ++-- webrtc/rtc_base/BUILD.gn | 10 +- webrtc/rtc_base/callback.h.pump | 4 +- webrtc/rtc_base/sigslottester.h.pump | 4 +- webrtc/rtc_tools/BUILD.gn | 10 +- webrtc/rtc_tools/network_tester/BUILD.gn | 8 +- webrtc/sdk/BUILD.gn | 26 ++-- webrtc/sdk/android/BUILD.gn | 26 ++-- .../PeerConnection/RTCVideoCodecH264.mm | 2 +- .../objc_video_decoder_factory.mm | 4 +- .../objc_video_encoder_factory.mm | 4 +- .../Framework/UnitTests/RTCTracingTest.mm | 2 +- webrtc/stats/BUILD.gn | 8 +- webrtc/system_wrappers/BUILD.gn | 14 +- webrtc/test/BUILD.gn | 38 ++--- webrtc/test/fuzzers/BUILD.gn | 26 ++-- webrtc/video/BUILD.gn | 24 ++-- webrtc/voice_engine/BUILD.gn | 16 +-- 221 files changed, 425 insertions(+), 4008 deletions(-) delete mode 100644 webrtc/base/BUILD.gn delete mode 100644 webrtc/base/Dummy.java delete mode 100644 webrtc/base/array_view.h delete mode 100644 webrtc/base/arraysize.h delete mode 100644 webrtc/base/asyncinvoker-inl.h delete mode 100644 webrtc/base/asyncinvoker.h delete mode 100644 webrtc/base/asyncpacketsocket.h delete mode 100644 webrtc/base/asyncresolverinterface.h delete mode 100644 webrtc/base/asyncsocket.h delete mode 100644 webrtc/base/asynctcpsocket.h delete mode 100644 webrtc/base/asyncudpsocket.h delete mode 100644 webrtc/base/atomicops.h delete mode 100644 webrtc/base/base64.h delete mode 100644 webrtc/base/basictypes.h delete mode 100644 webrtc/base/bind.h delete mode 100644 webrtc/base/bitbuffer.h delete mode 100644 webrtc/base/buffer.h delete mode 100644 webrtc/base/bufferqueue.h delete mode 100644 webrtc/base/bytebuffer.h delete mode 100644 webrtc/base/byteorder.h delete mode 100644 webrtc/base/callback.h delete mode 100644 webrtc/base/checks.h delete mode 100644 webrtc/base/compile_assert_c.h delete mode 100644 webrtc/base/constructormagic.h delete mode 100644 webrtc/base/copyonwritebuffer.h delete mode 100644 webrtc/base/cpu_time.h delete mode 100644 webrtc/base/crc32.h delete mode 100644 webrtc/base/criticalsection.h delete mode 100644 webrtc/base/cryptstring.h delete mode 100644 webrtc/base/deprecation.h delete mode 100644 webrtc/base/dscp.h delete mode 100644 webrtc/base/event.h delete mode 100644 webrtc/base/event_tracer.h delete mode 100644 webrtc/base/fakeclock.h delete mode 100644 webrtc/base/fakenetwork.h delete mode 100644 webrtc/base/fakesslidentity.h delete mode 100644 webrtc/base/file.h delete mode 100644 webrtc/base/filerotatingstream.h delete mode 100644 webrtc/base/fileutils.h delete mode 100644 webrtc/base/firewallsocketserver.h delete mode 100644 webrtc/base/flags.h delete mode 100644 webrtc/base/format_macros.h delete mode 100644 webrtc/base/function_view.h delete mode 100644 webrtc/base/gtest_prod_util.h delete mode 100644 webrtc/base/gunit.h delete mode 100644 webrtc/base/gunit_prod.h delete mode 100644 webrtc/base/helpers.h delete mode 100644 webrtc/base/httpbase.h delete mode 100644 webrtc/base/httpcommon-inl.h delete mode 100644 webrtc/base/httpcommon.h delete mode 100644 webrtc/base/httpserver.h delete mode 100644 webrtc/base/ifaddrs-android.h delete mode 100644 webrtc/base/ifaddrs_converter.h delete mode 100644 webrtc/base/ignore_wundef.h delete mode 100644 webrtc/base/ipaddress.h delete mode 100644 webrtc/base/json.h delete mode 100644 webrtc/base/keep_ref_until_done.h delete mode 100644 webrtc/base/location.h delete mode 100644 webrtc/base/logging.h delete mode 100644 webrtc/base/logsinks.h delete mode 100644 webrtc/base/macutils.h delete mode 100644 webrtc/base/mathutils.h delete mode 100644 webrtc/base/md5.h delete mode 100644 webrtc/base/md5digest.h delete mode 100644 webrtc/base/memory_usage.h delete mode 100644 webrtc/base/messagedigest.h delete mode 100644 webrtc/base/messagehandler.h delete mode 100644 webrtc/base/messagequeue.h delete mode 100644 webrtc/base/mod_ops.h delete mode 100644 webrtc/base/natserver.h delete mode 100644 webrtc/base/natsocketfactory.h delete mode 100644 webrtc/base/nattypes.h delete mode 100644 webrtc/base/nethelpers.h delete mode 100644 webrtc/base/network.h delete mode 100644 webrtc/base/networkmonitor.h delete mode 100644 webrtc/base/networkroute.h delete mode 100644 webrtc/base/nullsocketserver.h delete mode 100644 webrtc/base/numerics/exp_filter.h delete mode 100644 webrtc/base/numerics/percentile_filter.h delete mode 100644 webrtc/base/onetimeevent.h delete mode 100644 webrtc/base/openssl.h delete mode 100644 webrtc/base/openssladapter.h delete mode 100644 webrtc/base/openssldigest.h delete mode 100644 webrtc/base/opensslidentity.h delete mode 100644 webrtc/base/opensslstreamadapter.h delete mode 100644 webrtc/base/optional.h delete mode 100644 webrtc/base/optionsfile.h delete mode 100644 webrtc/base/pathutils.h delete mode 100644 webrtc/base/physicalsocketserver.h delete mode 100644 webrtc/base/platform_file.h delete mode 100644 webrtc/base/platform_thread.h delete mode 100644 webrtc/base/platform_thread_types.h delete mode 100644 webrtc/base/protobuf_utils.h delete mode 100644 webrtc/base/proxyinfo.h delete mode 100644 webrtc/base/proxyserver.h delete mode 100644 webrtc/base/ptr_util.h delete mode 100644 webrtc/base/race_checker.h delete mode 100644 webrtc/base/random.h delete mode 100644 webrtc/base/rate_limiter.h delete mode 100644 webrtc/base/rate_statistics.h delete mode 100644 webrtc/base/ratelimiter.h delete mode 100644 webrtc/base/ratetracker.h delete mode 100644 webrtc/base/refcount.h delete mode 100644 webrtc/base/refcountedobject.h delete mode 100644 webrtc/base/rollingaccumulator.h delete mode 100644 webrtc/base/rtccertificate.h delete mode 100644 webrtc/base/rtccertificategenerator.h delete mode 100644 webrtc/base/safe_compare.h delete mode 100644 webrtc/base/safe_conversions.h delete mode 100644 webrtc/base/safe_conversions_impl.h delete mode 100644 webrtc/base/safe_minmax.h delete mode 100644 webrtc/base/sanitizer.h delete mode 100644 webrtc/base/scoped_ref_ptr.h delete mode 100644 webrtc/base/sequenced_task_checker.h delete mode 100644 webrtc/base/sequenced_task_checker_impl.h delete mode 100644 webrtc/base/sha1.h delete mode 100644 webrtc/base/sha1digest.h delete mode 100644 webrtc/base/sigslot.h delete mode 100644 webrtc/base/sigslottester.h delete mode 100644 webrtc/base/socket.h delete mode 100644 webrtc/base/socket_unittest.h delete mode 100644 webrtc/base/socketadapters.h delete mode 100644 webrtc/base/socketaddress.h delete mode 100644 webrtc/base/socketaddresspair.h delete mode 100644 webrtc/base/socketfactory.h delete mode 100644 webrtc/base/socketserver.h delete mode 100644 webrtc/base/socketstream.h delete mode 100644 webrtc/base/ssladapter.h delete mode 100644 webrtc/base/sslfingerprint.h delete mode 100644 webrtc/base/sslidentity.h delete mode 100644 webrtc/base/sslroots.h delete mode 100644 webrtc/base/sslstreamadapter.h delete mode 100644 webrtc/base/stream.h delete mode 100644 webrtc/base/string_to_number.h delete mode 100644 webrtc/base/stringencode.h delete mode 100644 webrtc/base/stringize_macros.h delete mode 100644 webrtc/base/stringutils.h delete mode 100644 webrtc/base/swap_queue.h delete mode 100644 webrtc/base/task_queue.h delete mode 100644 webrtc/base/task_queue_posix.h delete mode 100644 webrtc/base/template_util.h delete mode 100644 webrtc/base/testbase64.h delete mode 100644 webrtc/base/testclient.h delete mode 100644 webrtc/base/testechoserver.h delete mode 100644 webrtc/base/testutils.h delete mode 100644 webrtc/base/thread.h delete mode 100644 webrtc/base/thread_annotations.h delete mode 100644 webrtc/base/thread_checker.h delete mode 100644 webrtc/base/thread_checker_impl.h delete mode 100644 webrtc/base/timedelta.h delete mode 100644 webrtc/base/timestampaligner.h delete mode 100644 webrtc/base/timeutils.h delete mode 100644 webrtc/base/trace_event.h delete mode 100644 webrtc/base/transformadapter.h delete mode 100644 webrtc/base/type_traits.h delete mode 100644 webrtc/base/unixfilesystem.h delete mode 100644 webrtc/base/virtualsocketserver.h delete mode 100644 webrtc/base/weak_ptr.h delete mode 100644 webrtc/base/win32.h delete mode 100644 webrtc/base/win32filesystem.h delete mode 100644 webrtc/base/win32socketinit.h delete mode 100644 webrtc/base/win32socketserver.h delete mode 100644 webrtc/base/win32window.h delete mode 100644 webrtc/base/window.h diff --git a/PRESUBMIT.py b/PRESUBMIT.py index 3cc90cc0f9..2a22d73f5d 100755 --- a/PRESUBMIT.py +++ b/PRESUBMIT.py @@ -18,7 +18,6 @@ CPPLINT_BLACKLIST = [ 'tools_webrtc', 'webrtc/api/video_codecs/video_decoder.h', 'webrtc/api/video_codecs/video_encoder.h', - 'webrtc/base', 'webrtc/examples/objc', 'webrtc/media', 'webrtc/modules/audio_coding', @@ -74,7 +73,6 @@ NATIVE_API_DIRS = ( # These directories should not be used but are maintained only to avoid breaking # some legacy downstream code. LEGACY_API_DIRS = ( - 'webrtc/base', 'webrtc/common_audio/include', 'webrtc/modules/audio_coding/include', 'webrtc/modules/audio_conference_mixer/include', @@ -91,6 +89,7 @@ LEGACY_API_DIRS = ( 'webrtc/modules/video_coding/codecs/vp8/include', 'webrtc/modules/video_coding/codecs/vp9/include', 'webrtc/modules/video_coding/include', + 'webrtc/rtc_base', 'webrtc/system_wrappers/include', 'webrtc/voice_engine/include', ) diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index 9280d540bc..12ee93c872 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -232,8 +232,8 @@ rtc_source_set("video_stream_api") { deps = [ ":webrtc_common", "api:transport_api", - "base:rtc_base_approved", "common_video:common_video", + "rtc_base:rtc_base_approved", ] } @@ -252,7 +252,6 @@ if (!build_with_chromium) { "api", "api:transport_api", "audio", - "base", "call", "common_audio", "common_video", @@ -291,7 +290,6 @@ if (!build_with_chromium) { ":video_engine_tests", ":webrtc_nonparallel_tests", ":webrtc_perf_tests", - "base:rtc_base_tests_utils", "common_audio:common_audio_unittests", "common_video:common_video_unittests", "media:rtc_media_unittests", @@ -306,6 +304,7 @@ if (!build_with_chromium) { "ortc:ortc_unittests", "pc:peerconnection_unittests", "pc:rtc_pc_unittests", + "rtc_base:rtc_base_tests_utils", "stats:rtc_stats_unittests", "system_wrappers:system_wrappers_unittests", "test", @@ -393,16 +392,16 @@ if (rtc_include_tests) { ":webrtc_common", "api:rtc_api_unittests", "api/audio_codecs/test:audio_codecs_api_unittests", - "base:rtc_base_approved_unittests", - "base:rtc_base_tests_main", - "base:rtc_base_tests_utils", - "base:rtc_base_unittests", - "base:rtc_numerics_unittests", - "base:rtc_task_queue_unittests", - "base:sequenced_task_checker_unittests", - "base:weak_ptr_unittests", "p2p:libstunprober_unittests", "p2p:rtc_p2p_unittests", + "rtc_base:rtc_base_approved_unittests", + "rtc_base:rtc_base_tests_main", + "rtc_base:rtc_base_tests_utils", + "rtc_base:rtc_base_unittests", + "rtc_base:rtc_numerics_unittests", + "rtc_base:rtc_task_queue_unittests", + "rtc_base:sequenced_task_checker_unittests", + "rtc_base:weak_ptr_unittests", "system_wrappers:metrics_default", ] @@ -440,12 +439,12 @@ if (rtc_include_tests) { testonly = true deps = [ "audio:audio_tests", - "base:rtc_base_tests_utils", # TODO(eladalon): call_tests aren't actually video-specific, so we # should move them to a more appropriate test suite. "call:call_tests", "modules/video_capture", + "rtc_base:rtc_base_tests_utils", "test:test_common", "test:test_main", "test:video_test_common", @@ -517,7 +516,7 @@ if (rtc_include_tests) { rtc_test("webrtc_nonparallel_tests") { testonly = true deps = [ - "base:rtc_base_nonparallel_tests", + "rtc_base:rtc_base_nonparallel_tests", ] if (is_android) { deps += [ "//testing/android/native_test:native_test_support" ] diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn index 249411b1c2..c413f2874a 100644 --- a/webrtc/api/BUILD.gn +++ b/webrtc/api/BUILD.gn @@ -28,7 +28,7 @@ rtc_source_set("call_api") { ":audio_mixer_api", ":transport_api", "..:webrtc_common", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "audio_codecs:audio_codecs_api", ] } @@ -83,8 +83,8 @@ rtc_static_library("libjingle_peerconnection_api") { deps = [ ":rtc_stats_api", "..:webrtc_common", - "../base:rtc_base", - "../base:rtc_base_approved", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", "audio_codecs:audio_codecs_api", ] @@ -143,7 +143,7 @@ rtc_source_set("rtc_stats_api") { ] deps = [ - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } @@ -153,8 +153,8 @@ rtc_source_set("audio_mixer_api") { ] deps = [ - "../base:rtc_base_approved", "../modules:module_api", + "../rtc_base:rtc_base_approved", ] } @@ -178,7 +178,7 @@ rtc_source_set("video_frame_api") { ] deps = [ - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] @@ -206,7 +206,7 @@ rtc_source_set("libjingle_peerconnection_test_api") { ] deps = [ - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } @@ -235,7 +235,7 @@ if (rtc_include_tests) { ] deps = [ ":libjingle_peerconnection_api", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] if (!build_with_chromium && is_clang) { # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). diff --git a/webrtc/api/audio_codecs/BUILD.gn b/webrtc/api/audio_codecs/BUILD.gn index 416ccbbf3f..2174fb1063 100644 --- a/webrtc/api/audio_codecs/BUILD.gn +++ b/webrtc/api/audio_codecs/BUILD.gn @@ -27,7 +27,7 @@ rtc_source_set("audio_codecs_api") { ] deps = [ "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } @@ -38,8 +38,8 @@ rtc_static_library("builtin_audio_decoder_factory") { ] deps = [ ":audio_codecs_api", - "../../base:rtc_base_approved", "../../modules/audio_coding:builtin_audio_decoder_factory_internal", + "../../rtc_base:rtc_base_approved", ] } @@ -50,7 +50,7 @@ rtc_static_library("builtin_audio_encoder_factory") { ] deps = [ ":audio_codecs_api", - "../../base:rtc_base_approved", "../../modules/audio_coding:builtin_audio_encoder_factory_internal", + "../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/api/audio_codecs/g722/BUILD.gn b/webrtc/api/audio_codecs/g722/BUILD.gn index d2470a26e4..2c1349a7c5 100644 --- a/webrtc/api/audio_codecs/g722/BUILD.gn +++ b/webrtc/api/audio_codecs/g722/BUILD.gn @@ -26,8 +26,8 @@ rtc_static_library("audio_encoder_g722") { deps = [ ":audio_encoder_g722_config", "..:audio_codecs_api", - "../../../base:rtc_base_approved", "../../../modules/audio_coding:g722", + "../../../rtc_base:rtc_base_approved", ] } @@ -39,7 +39,7 @@ rtc_static_library("audio_decoder_g722") { deps = [ "..:audio_codecs_api", "../../..:webrtc_common", - "../../../base:rtc_base_approved", "../../../modules/audio_coding:g722", + "../../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/api/audio_codecs/ilbc/BUILD.gn b/webrtc/api/audio_codecs/ilbc/BUILD.gn index bba2662ad6..6ef8856639 100644 --- a/webrtc/api/audio_codecs/ilbc/BUILD.gn +++ b/webrtc/api/audio_codecs/ilbc/BUILD.gn @@ -26,8 +26,8 @@ rtc_static_library("audio_encoder_ilbc") { deps = [ ":audio_encoder_ilbc_config", "..:audio_codecs_api", - "../../../base:rtc_base_approved", "../../../modules/audio_coding:ilbc", + "../../../rtc_base:rtc_base_approved", ] } @@ -39,7 +39,7 @@ rtc_static_library("audio_decoder_ilbc") { deps = [ "..:audio_codecs_api", "../../..:webrtc_common", - "../../../base:rtc_base_approved", "../../../modules/audio_coding:ilbc", + "../../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/api/audio_codecs/opus/BUILD.gn b/webrtc/api/audio_codecs/opus/BUILD.gn index c7f7ac8201..29a68ff74e 100644 --- a/webrtc/api/audio_codecs/opus/BUILD.gn +++ b/webrtc/api/audio_codecs/opus/BUILD.gn @@ -18,7 +18,7 @@ rtc_static_library("audio_encoder_opus_config") { "audio_encoder_opus_config.h", ] deps = [ - "../../../base:rtc_base_approved", + "../../../rtc_base:rtc_base_approved", ] defines = [] if (rtc_opus_variable_complexity) { @@ -35,9 +35,9 @@ rtc_source_set("audio_encoder_opus") { deps = [ ":audio_encoder_opus_config", "..:audio_codecs_api", - "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed? - "../../../base:rtc_base_approved", "../../../modules/audio_coding:webrtc_opus", + "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed? + "../../../rtc_base:rtc_base_approved", ] } @@ -49,7 +49,7 @@ rtc_static_library("audio_decoder_opus") { deps = [ "..:audio_codecs_api", "../../..:webrtc_common", - "../../../base:rtc_base_approved", "../../../modules/audio_coding:webrtc_opus", + "../../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/api/audio_codecs/test/BUILD.gn b/webrtc/api/audio_codecs/test/BUILD.gn index 32cef2d5cb..4a0c878920 100644 --- a/webrtc/api/audio_codecs/test/BUILD.gn +++ b/webrtc/api/audio_codecs/test/BUILD.gn @@ -21,8 +21,8 @@ if (rtc_include_tests) { ] deps = [ "..:audio_codecs_api", - "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed? - "../../../base:rtc_base_approved", + "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed? + "../../../rtc_base:rtc_base_approved", "../../../test:audio_codec_mocks", "../../../test:test_support", "../g722:audio_decoder_g722", diff --git a/webrtc/api/video_codecs/BUILD.gn b/webrtc/api/video_codecs/BUILD.gn index d435534d83..5e27c78059 100644 --- a/webrtc/api/video_codecs/BUILD.gn +++ b/webrtc/api/video_codecs/BUILD.gn @@ -21,7 +21,7 @@ rtc_source_set("video_codecs_api") { deps = [ "..:video_frame_api", "../..:webrtc_common", - "../../base:rtc_base_approved", "../../common_video", + "../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn index 1577316cd4..2b7d06fcfa 100644 --- a/webrtc/audio/BUILD.gn +++ b/webrtc/audio/BUILD.gn @@ -37,8 +37,6 @@ rtc_static_library("audio") { "../api:call_api", "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_encoder_factory", - "../base:rtc_base_approved", - "../base:rtc_task_queue", "../call:call_interfaces", "../call:rtp_interfaces", "../common_audio", @@ -50,6 +48,8 @@ rtc_static_library("audio") { "../modules/pacing:pacing", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp:rtp_rtcp", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue", "../system_wrappers", "../voice_engine", ] @@ -77,14 +77,14 @@ if (rtc_include_tests) { deps = [ ":audio", "../api:mock_audio_mixer", - "../base:rtc_base_approved", - "../base:rtc_task_queue", "../call:rtp_receiver", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", "../modules/congestion_controller:congestion_controller", "../modules/congestion_controller:mock_congestion_controller", "../modules/pacing:pacing", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue", "../test:test_common", "../test:test_support", "utility:utility_tests", diff --git a/webrtc/audio/utility/BUILD.gn b/webrtc/audio/utility/BUILD.gn index ac477e4f25..65f9cb0da6 100644 --- a/webrtc/audio/utility/BUILD.gn +++ b/webrtc/audio/utility/BUILD.gn @@ -21,9 +21,9 @@ rtc_static_library("audio_frame_operations") { deps = [ "../..:webrtc_common", - "../../base:rtc_base_approved", "../../modules:module_api", "../../modules/audio_coding:audio_format_conversion", + "../../rtc_base:rtc_base_approved", ] } @@ -35,8 +35,8 @@ if (rtc_include_tests) { ] deps = [ ":audio_frame_operations", - "../../base:rtc_base_approved", "../../modules:module_api", + "../../rtc_base:rtc_base_approved", "../../test:test_support", "//testing/gtest", ] diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn deleted file mode 100644 index c786f15916..0000000000 --- a/webrtc/base/BUILD.gn +++ /dev/null @@ -1,135 +0,0 @@ -# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -import("//build/config/crypto.gni") -import("//build/config/ui.gni") -import("../webrtc.gni") - -if (is_android) { - import("//build/config/android/config.gni") - import("//build/config/android/rules.gni") -} -if (is_win) { - import("//build/config/clang/clang.gni") -} - -group("base") { - public_deps = [ - ":rtc_base", - ":rtc_base_approved", - ":rtc_task_queue", - ":sequenced_task_checker", - ":weak_ptr", - ] -} - -if (!rtc_build_ssl) { - config("external_ssl_library") { - assert(rtc_ssl_root != "", - "You must specify rtc_ssl_root when rtc_build_ssl==0.") - include_dirs = [ rtc_ssl_root ] - } -} - -# The targets below are deprecated and only exist here temporarily during -# refactoring. See https://bugs.webrtc.org/7634 for more details. - -group("protobuf_utils") { - public_deps = [ "../rtc_base:protobuf_utils" ] -} - -group("compile_assert_c") { - public_deps = [ "../rtc_base:compile_assert_c" ] -} - -group("rtc_base_approved") { - public_deps = [ "../rtc_base:rtc_base_approved" ] -} - -group("rtc_task_queue") { - public_deps = [ "../rtc_base:rtc_task_queue" ] -} - -group("sequenced_task_checker") { - public_deps = [ "../rtc_base:sequenced_task_checker" ] -} - -group("weak_ptr") { - public_deps = [ "../rtc_base:weak_ptr" ] -} - -group("rtc_numerics") { - public_deps = [ "../rtc_base:rtc_numerics" ] -} - -group("rtc_json") { - public_deps = [ "../rtc_base:rtc_json" ] -} - -group("rtc_base") { - public_deps = [ "../rtc_base:rtc_base" ] -} - -group("gtest_prod") { - public_deps = [ "../rtc_base:gtest_prod" ] -} - -group("rtc_base_tests_utils") { - testonly = true - public_deps = [ "../rtc_base:rtc_base_tests_utils" ] -} - -if (rtc_include_tests) { - group("rtc_base_tests_main") { - testonly = true - public_deps = [ "../rtc_base:rtc_base_tests_main" ] - } - - group("rtc_base_nonparallel_tests") { - testonly = true - public_deps = [ "../rtc_base:rtc_base_nonparallel_tests" ] - } - - group("rtc_base_approved_unittests") { - testonly = true - public_deps = [ "../rtc_base:rtc_base_approved_unittests" ] - } - - group("sequenced_task_checker_unittests") { - testonly = true - public_deps = [ "../rtc_base:sequenced_task_checker_unittests" ] - } - - group("weak_ptr_unittests") { - testonly = true - public_deps = [ "../rtc_base:weak_ptr_unittests" ] - } - - group("rtc_task_queue_unittests") { - testonly = true - public_deps = [ "../rtc_base:rtc_task_queue_unittests" ] - } - - - group("rtc_numerics_unittests") { - testonly = true - public_deps = [ "../rtc_base:rtc_numerics_unittests" ] - } - - group("rtc_base_unittests") { - testonly = true - public_deps = [ "../rtc_base:rtc_base_unittests" ] - } -} - -if (is_android) { - android_library("base_java") { - java_files = [ "Dummy.java" ] # Need one file to avoid hitting an assert. - deps = [ "../rtc_base:base_java" ] - } -} diff --git a/webrtc/base/Dummy.java b/webrtc/base/Dummy.java deleted file mode 100644 index 60cd440fd4..0000000000 --- a/webrtc/base/Dummy.java +++ /dev/null @@ -1,9 +0,0 @@ -/** - * This class only exists as glue in a transition. - * TODO(kjellander): Remove. - * See https://bugs.webrtc.org/7634 for more details. - */ -class Dummy { - Dummy() { - } -} diff --git a/webrtc/base/array_view.h b/webrtc/base/array_view.h deleted file mode 100644 index a451b59e2d..0000000000 --- a/webrtc/base/array_view.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ARRAY_VIEW_H_ -#define WEBRTC_BASE_ARRAY_VIEW_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/array_view.h" - -#endif // WEBRTC_BASE_ARRAY_VIEW_H_ diff --git a/webrtc/base/arraysize.h b/webrtc/base/arraysize.h deleted file mode 100644 index 8b37efa04b..0000000000 --- a/webrtc/base/arraysize.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ARRAYSIZE_H_ -#define WEBRTC_BASE_ARRAYSIZE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/arraysize.h" - -#endif // WEBRTC_BASE_ARRAYSIZE_H_ diff --git a/webrtc/base/asyncinvoker-inl.h b/webrtc/base/asyncinvoker-inl.h deleted file mode 100644 index cce42264ab..0000000000 --- a/webrtc/base/asyncinvoker-inl.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2014 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ASYNCINVOKER_INL_H_ -#define WEBRTC_BASE_ASYNCINVOKER_INL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/asyncinvoker-inl.h" - -#endif // WEBRTC_BASE_ASYNCINVOKER_INL_H_ diff --git a/webrtc/base/asyncinvoker.h b/webrtc/base/asyncinvoker.h deleted file mode 100644 index 0fcfc04947..0000000000 --- a/webrtc/base/asyncinvoker.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2014 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ASYNCINVOKER_H_ -#define WEBRTC_BASE_ASYNCINVOKER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/asyncinvoker.h" - -#endif // WEBRTC_BASE_ASYNCINVOKER_H_ diff --git a/webrtc/base/asyncpacketsocket.h b/webrtc/base/asyncpacketsocket.h deleted file mode 100644 index 809f1789af..0000000000 --- a/webrtc/base/asyncpacketsocket.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_ -#define WEBRTC_BASE_ASYNCPACKETSOCKET_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/asyncpacketsocket.h" - -#endif // WEBRTC_BASE_ASYNCPACKETSOCKET_H_ diff --git a/webrtc/base/asyncresolverinterface.h b/webrtc/base/asyncresolverinterface.h deleted file mode 100644 index b2a172fb17..0000000000 --- a/webrtc/base/asyncresolverinterface.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2013 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ASYNCRESOLVERINTERFACE_H_ -#define WEBRTC_BASE_ASYNCRESOLVERINTERFACE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/asyncresolverinterface.h" - -#endif diff --git a/webrtc/base/asyncsocket.h b/webrtc/base/asyncsocket.h deleted file mode 100644 index 9c971394d3..0000000000 --- a/webrtc/base/asyncsocket.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ASYNCSOCKET_H_ -#define WEBRTC_BASE_ASYNCSOCKET_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/asyncsocket.h" - -#endif // WEBRTC_BASE_ASYNCSOCKET_H_ diff --git a/webrtc/base/asynctcpsocket.h b/webrtc/base/asynctcpsocket.h deleted file mode 100644 index d64927bcd5..0000000000 --- a/webrtc/base/asynctcpsocket.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ASYNCTCPSOCKET_H_ -#define WEBRTC_BASE_ASYNCTCPSOCKET_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/asynctcpsocket.h" - -#endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_ diff --git a/webrtc/base/asyncudpsocket.h b/webrtc/base/asyncudpsocket.h deleted file mode 100644 index c3212c0cc6..0000000000 --- a/webrtc/base/asyncudpsocket.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ASYNCUDPSOCKET_H_ -#define WEBRTC_BASE_ASYNCUDPSOCKET_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/asyncudpsocket.h" - -#endif // WEBRTC_BASE_ASYNCUDPSOCKET_H_ diff --git a/webrtc/base/atomicops.h b/webrtc/base/atomicops.h deleted file mode 100644 index 3c3684814a..0000000000 --- a/webrtc/base/atomicops.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2011 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ATOMICOPS_H_ -#define WEBRTC_BASE_ATOMICOPS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/atomicops.h" - -#endif // WEBRTC_BASE_ATOMICOPS_H_ diff --git a/webrtc/base/base64.h b/webrtc/base/base64.h deleted file mode 100644 index 1e28357a67..0000000000 --- a/webrtc/base/base64.h +++ /dev/null @@ -1,20 +0,0 @@ - -//********************************************************************* -//* C_Base64 - a simple base64 encoder and decoder. -//* -//* Copyright (c) 1999, Bob Withers - bwit@pobox.com -//* -//* This code may be freely used for any purpose, either personal -//* or commercial, provided the authors copyright notice remains -//* intact. -//********************************************************************* - -#ifndef WEBRTC_BASE_BASE64_H_ -#define WEBRTC_BASE_BASE64_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/base64.h" - -#endif // WEBRTC_BASE_BASE64_H_ diff --git a/webrtc/base/basictypes.h b/webrtc/base/basictypes.h deleted file mode 100644 index 42ffa5a62e..0000000000 --- a/webrtc/base/basictypes.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_BASICTYPES_H_ -#define WEBRTC_BASE_BASICTYPES_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/basictypes.h" - -#endif // WEBRTC_BASE_BASICTYPES_H_ diff --git a/webrtc/base/bind.h b/webrtc/base/bind.h deleted file mode 100644 index 39d441f008..0000000000 --- a/webrtc/base/bind.h +++ /dev/null @@ -1,69 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Bind() is an overloaded function that converts method calls into function -// objects (aka functors). The method object is captured as a scoped_refptr<> if -// possible, and as a raw pointer otherwise. Any arguments to the method are -// captured by value. The return value of Bind is a stateful, nullary function -// object. Care should be taken about the lifetime of objects captured by -// Bind(); the returned functor knows nothing about the lifetime of a non -// ref-counted method object or any arguments passed by pointer, and calling the -// functor with a destroyed object will surely do bad things. -// -// To prevent the method object from being captured as a scoped_refptr<>, you -// can use Unretained. But this should only be done when absolutely necessary, -// and when the caller knows the extra reference isn't needed. -// -// Example usage: -// struct Foo { -// int Test1() { return 42; } -// int Test2() const { return 52; } -// int Test3(int x) { return x*x; } -// float Test4(int x, float y) { return x + y; } -// }; -// -// int main() { -// Foo foo; -// cout << rtc::Bind(&Foo::Test1, &foo)() << endl; -// cout << rtc::Bind(&Foo::Test2, &foo)() << endl; -// cout << rtc::Bind(&Foo::Test3, &foo, 3)() << endl; -// cout << rtc::Bind(&Foo::Test4, &foo, 7, 8.5f)() << endl; -// } -// -// Example usage of ref counted objects: -// struct Bar { -// int AddRef(); -// int Release(); -// -// void Test() {} -// void BindThis() { -// // The functor passed to AsyncInvoke() will keep this object alive. -// invoker.AsyncInvoke(RTC_FROM_HERE,rtc::Bind(&Bar::Test, this)); -// } -// }; -// -// int main() { -// rtc::scoped_refptr bar = new rtc::RefCountedObject(); -// auto functor = rtc::Bind(&Bar::Test, bar); -// bar = nullptr; -// // The functor stores an internal scoped_refptr, so this is safe. -// functor(); -// } -// - -#ifndef WEBRTC_BASE_BIND_H_ -#define WEBRTC_BASE_BIND_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/bind.h" - -#endif // WEBRTC_BASE_BIND_H_ diff --git a/webrtc/base/bitbuffer.h b/webrtc/base/bitbuffer.h deleted file mode 100644 index 09cba3c10b..0000000000 --- a/webrtc/base/bitbuffer.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_BITBUFFER_H_ -#define WEBRTC_BASE_BITBUFFER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/bitbuffer.h" - -#endif // WEBRTC_BASE_BITBUFFER_H_ diff --git a/webrtc/base/buffer.h b/webrtc/base/buffer.h deleted file mode 100644 index 92c85d9591..0000000000 --- a/webrtc/base/buffer.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_BUFFER_H_ -#define WEBRTC_BASE_BUFFER_H_ - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/buffer.h" - -#endif // WEBRTC_BASE_BUFFER_H_ diff --git a/webrtc/base/bufferqueue.h b/webrtc/base/bufferqueue.h deleted file mode 100644 index 3142ae3703..0000000000 --- a/webrtc/base/bufferqueue.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_BUFFERQUEUE_H_ -#define WEBRTC_BASE_BUFFERQUEUE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/bufferqueue.h" - -#endif // WEBRTC_BASE_BUFFERQUEUE_H_ diff --git a/webrtc/base/bytebuffer.h b/webrtc/base/bytebuffer.h deleted file mode 100644 index 0cc9a12d1c..0000000000 --- a/webrtc/base/bytebuffer.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_BYTEBUFFER_H_ -#define WEBRTC_BASE_BYTEBUFFER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/bytebuffer.h" - -#endif // WEBRTC_BASE_BYTEBUFFER_H_ diff --git a/webrtc/base/byteorder.h b/webrtc/base/byteorder.h deleted file mode 100644 index 28cbaa577b..0000000000 --- a/webrtc/base/byteorder.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_BYTEORDER_H_ -#define WEBRTC_BASE_BYTEORDER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/byteorder.h" - -#endif // WEBRTC_BASE_BYTEORDER_H_ diff --git a/webrtc/base/callback.h b/webrtc/base/callback.h deleted file mode 100644 index 4da1e6dfab..0000000000 --- a/webrtc/base/callback.h +++ /dev/null @@ -1,70 +0,0 @@ -// This file was GENERATED by command: -// pump.py callback.h.pump -// DO NOT EDIT BY HAND!!! - -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// To generate callback.h from callback.h.pump, execute: -// /home/build/google3/third_party/gtest/scripts/pump.py callback.h.pump - -// Callbacks are callable object containers. They can hold a function pointer -// or a function object and behave like a value type. Internally, data is -// reference-counted, making copies and pass-by-value inexpensive. -// -// Callbacks are typed using template arguments. The format is: -// CallbackN -// where N is the number of arguments supplied to the callable object. -// Callbacks are invoked using operator(), just like a function or a function -// object. Default-constructed callbacks are "empty," and executing an empty -// callback does nothing. A callback can be made empty by assigning it from -// a default-constructed callback. -// -// Callbacks are similar in purpose to std::function (which isn't available on -// all platforms we support) and a lightweight alternative to sigslots. Since -// they effectively hide the type of the object they call, they're useful in -// breaking dependencies between objects that need to interact with one another. -// Notably, they can hold the results of Bind(), std::bind*, etc, without -// needing -// to know the resulting object type of those calls. -// -// Sigslots, on the other hand, provide a fuller feature set, such as multiple -// subscriptions to a signal, optional thread-safety, and lifetime tracking of -// slots. When these features are needed, choose sigslots. -// -// Example: -// int sqr(int x) { return x * x; } -// struct AddK { -// int k; -// int operator()(int x) const { return x + k; } -// } add_k = {5}; -// -// Callback1 my_callback; -// cout << my_callback.empty() << endl; // true -// -// my_callback = Callback1(&sqr); -// cout << my_callback.empty() << endl; // false -// cout << my_callback(3) << endl; // 9 -// -// my_callback = Callback1(add_k); -// cout << my_callback(10) << endl; // 15 -// -// my_callback = Callback1(); -// cout << my_callback.empty() << endl; // true - -#ifndef WEBRTC_BASE_CALLBACK_H_ -#define WEBRTC_BASE_CALLBACK_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/callback.h" - -#endif // WEBRTC_BASE_CALLBACK_H_ diff --git a/webrtc/base/checks.h b/webrtc/base/checks.h deleted file mode 100644 index f56f157224..0000000000 --- a/webrtc/base/checks.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2006 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_CHECKS_H_ -#define WEBRTC_BASE_CHECKS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/checks.h" - -#endif // WEBRTC_BASE_CHECKS_H_ diff --git a/webrtc/base/compile_assert_c.h b/webrtc/base/compile_assert_c.h deleted file mode 100644 index 934cc9be7c..0000000000 --- a/webrtc/base/compile_assert_c.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_COMPILE_ASSERT_C_H_ -#define WEBRTC_BASE_COMPILE_ASSERT_C_H_ - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/compile_assert_c.h" - -#endif // WEBRTC_BASE_COMPILE_ASSERT_C_H_ diff --git a/webrtc/base/constructormagic.h b/webrtc/base/constructormagic.h deleted file mode 100644 index 21652c2d3d..0000000000 --- a/webrtc/base/constructormagic.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_CONSTRUCTORMAGIC_H_ -#define WEBRTC_BASE_CONSTRUCTORMAGIC_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/constructormagic.h" - -#endif // WEBRTC_BASE_CONSTRUCTORMAGIC_H_ diff --git a/webrtc/base/copyonwritebuffer.h b/webrtc/base/copyonwritebuffer.h deleted file mode 100644 index 6a95b31ced..0000000000 --- a/webrtc/base/copyonwritebuffer.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_COPYONWRITEBUFFER_H_ -#define WEBRTC_BASE_COPYONWRITEBUFFER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/copyonwritebuffer.h" - -#endif // WEBRTC_BASE_COPYONWRITEBUFFER_H_ diff --git a/webrtc/base/cpu_time.h b/webrtc/base/cpu_time.h deleted file mode 100644 index f627790822..0000000000 --- a/webrtc/base/cpu_time.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_CPU_TIME_H_ -#define WEBRTC_BASE_CPU_TIME_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/cpu_time.h" - -#endif // WEBRTC_BASE_CPU_TIME_H_ diff --git a/webrtc/base/crc32.h b/webrtc/base/crc32.h deleted file mode 100644 index 6854567cc6..0000000000 --- a/webrtc/base/crc32.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_CRC32_H_ -#define WEBRTC_BASE_CRC32_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/crc32.h" - -#endif // WEBRTC_BASE_CRC32_H_ diff --git a/webrtc/base/criticalsection.h b/webrtc/base/criticalsection.h deleted file mode 100644 index ab3f542244..0000000000 --- a/webrtc/base/criticalsection.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_CRITICALSECTION_H_ -#define WEBRTC_BASE_CRITICALSECTION_H_ - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/criticalsection.h" - -#endif // WEBRTC_BASE_CRITICALSECTION_H_ diff --git a/webrtc/base/cryptstring.h b/webrtc/base/cryptstring.h deleted file mode 100644 index 1a474b43f6..0000000000 --- a/webrtc/base/cryptstring.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_CRYPTSTRING_H_ -#define WEBRTC_BASE_CRYPTSTRING_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/cryptstring.h" - -#endif // WEBRTC_BASE_CRYPTSTRING_H_ diff --git a/webrtc/base/deprecation.h b/webrtc/base/deprecation.h deleted file mode 100644 index d6c5124c39..0000000000 --- a/webrtc/base/deprecation.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_DEPRECATION_H_ -#define WEBRTC_BASE_DEPRECATION_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/deprecation.h" - -#endif // WEBRTC_BASE_DEPRECATION_H_ diff --git a/webrtc/base/dscp.h b/webrtc/base/dscp.h deleted file mode 100644 index 1cf2756cdc..0000000000 --- a/webrtc/base/dscp.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2013 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_DSCP_H_ -#define WEBRTC_BASE_DSCP_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/dscp.h" - -#endif // WEBRTC_BASE_DSCP_H_ diff --git a/webrtc/base/event.h b/webrtc/base/event.h deleted file mode 100644 index 28ff7315e4..0000000000 --- a/webrtc/base/event.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_EVENT_H_ -#define WEBRTC_BASE_EVENT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/event.h" - -#endif // WEBRTC_BASE_EVENT_H_ diff --git a/webrtc/base/event_tracer.h b/webrtc/base/event_tracer.h deleted file mode 100644 index b6da14a47b..0000000000 --- a/webrtc/base/event_tracer.h +++ /dev/null @@ -1,34 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// This file defines the interface for event tracing in WebRTC. -// -// Event log handlers are set through SetupEventTracer(). User of this API will -// provide two function pointers to handle event tracing calls. -// -// * GetCategoryEnabledPtr -// Event tracing system calls this function to determine if a particular -// event category is enabled. -// -// * AddTraceEventPtr -// Adds a tracing event. It is the user's responsibility to log the data -// provided. -// -// Parameters for the above two functions are described in trace_event.h. - -#ifndef WEBRTC_BASE_EVENT_TRACER_H_ -#define WEBRTC_BASE_EVENT_TRACER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/event_tracer.h" - -#endif // WEBRTC_BASE_EVENT_TRACER_H_ diff --git a/webrtc/base/fakeclock.h b/webrtc/base/fakeclock.h deleted file mode 100644 index 22d640dbe5..0000000000 --- a/webrtc/base/fakeclock.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FAKECLOCK_H_ -#define WEBRTC_BASE_FAKECLOCK_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/fakeclock.h" - -#endif // WEBRTC_BASE_FAKECLOCK_H_ diff --git a/webrtc/base/fakenetwork.h b/webrtc/base/fakenetwork.h deleted file mode 100644 index c2c8e6dc40..0000000000 --- a/webrtc/base/fakenetwork.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2009 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FAKENETWORK_H_ -#define WEBRTC_BASE_FAKENETWORK_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/fakenetwork.h" - -#endif // WEBRTC_BASE_FAKENETWORK_H_ diff --git a/webrtc/base/fakesslidentity.h b/webrtc/base/fakesslidentity.h deleted file mode 100644 index da204b2ae6..0000000000 --- a/webrtc/base/fakesslidentity.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FAKESSLIDENTITY_H_ -#define WEBRTC_BASE_FAKESSLIDENTITY_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/fakesslidentity.h" - -#endif // WEBRTC_BASE_FAKESSLIDENTITY_H_ diff --git a/webrtc/base/file.h b/webrtc/base/file.h deleted file mode 100644 index 5a4465f6ac..0000000000 --- a/webrtc/base/file.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FILE_H_ -#define WEBRTC_BASE_FILE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/file.h" - -#endif // WEBRTC_BASE_FILE_H_ diff --git a/webrtc/base/filerotatingstream.h b/webrtc/base/filerotatingstream.h deleted file mode 100644 index 26306db6e0..0000000000 --- a/webrtc/base/filerotatingstream.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FILEROTATINGSTREAM_H_ -#define WEBRTC_BASE_FILEROTATINGSTREAM_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/filerotatingstream.h" - -#endif // WEBRTC_BASE_FILEROTATINGSTREAM_H_ diff --git a/webrtc/base/fileutils.h b/webrtc/base/fileutils.h deleted file mode 100644 index 18de30cf4d..0000000000 --- a/webrtc/base/fileutils.h +++ /dev/null @@ -1,20 +0,0 @@ - -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FILEUTILS_H_ -#define WEBRTC_BASE_FILEUTILS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/fileutils.h" - -#endif // WEBRTC_BASE_FILEUTILS_H_ diff --git a/webrtc/base/firewallsocketserver.h b/webrtc/base/firewallsocketserver.h deleted file mode 100644 index 18ad9bcdf3..0000000000 --- a/webrtc/base/firewallsocketserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FIREWALLSOCKETSERVER_H_ -#define WEBRTC_BASE_FIREWALLSOCKETSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/firewallsocketserver.h" - -#endif // WEBRTC_BASE_FIREWALLSOCKETSERVER_H_ diff --git a/webrtc/base/flags.h b/webrtc/base/flags.h deleted file mode 100644 index 9094466403..0000000000 --- a/webrtc/base/flags.h +++ /dev/null @@ -1,31 +0,0 @@ -/* - * Copyright 2006 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - - -// Originally comes from shared/commandlineflags/flags.h - -// Flags are defined and declared using DEFINE_xxx and DECLARE_xxx macros, -// where xxx is the flag type. Flags are referred to via FLAG_yyy, -// where yyy is the flag name. For intialization and iteration of flags, -// see the FlagList class. For full programmatic access to any -// flag, see the Flag class. -// -// The implementation only relies and basic C++ functionality -// and needs no special library or STL support. - -#ifndef WEBRTC_BASE_FLAGS_H_ -#define WEBRTC_BASE_FLAGS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/flags.h" - -#endif // SHARED_COMMANDLINEFLAGS_FLAGS_H_ diff --git a/webrtc/base/format_macros.h b/webrtc/base/format_macros.h deleted file mode 100644 index 844e71ebbb..0000000000 --- a/webrtc/base/format_macros.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FORMAT_MACROS_H_ -#define WEBRTC_BASE_FORMAT_MACROS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/format_macros.h" - -#endif // WEBRTC_BASE_FORMAT_MACROS_H_ diff --git a/webrtc/base/function_view.h b/webrtc/base/function_view.h deleted file mode 100644 index 12300268ef..0000000000 --- a/webrtc/base/function_view.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_FUNCTION_VIEW_H_ -#define WEBRTC_BASE_FUNCTION_VIEW_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/function_view.h" - -#endif // WEBRTC_BASE_FUNCTION_VIEW_H_ diff --git a/webrtc/base/gtest_prod_util.h b/webrtc/base/gtest_prod_util.h deleted file mode 100644 index 0c25943f2c..0000000000 --- a/webrtc/base/gtest_prod_util.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_GTEST_PROD_UTIL_H_ -#define WEBRTC_BASE_GTEST_PROD_UTIL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/gtest_prod_util.h" - -#endif // WEBRTC_BASE_GTEST_PROD_UTIL_H_ diff --git a/webrtc/base/gunit.h b/webrtc/base/gunit.h deleted file mode 100644 index d6c092e029..0000000000 --- a/webrtc/base/gunit.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_GUNIT_H_ -#define WEBRTC_BASE_GUNIT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/gunit.h" - -#endif // WEBRTC_BASE_GUNIT_H_ diff --git a/webrtc/base/gunit_prod.h b/webrtc/base/gunit_prod.h deleted file mode 100644 index 436abee92a..0000000000 --- a/webrtc/base/gunit_prod.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_GUNIT_PROD_H_ -#define WEBRTC_BASE_GUNIT_PROD_H_ - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/gunit_prod.h" - -#endif // WEBRTC_BASE_GUNIT_PROD_H_ diff --git a/webrtc/base/helpers.h b/webrtc/base/helpers.h deleted file mode 100644 index 86a388e8b0..0000000000 --- a/webrtc/base/helpers.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_HELPERS_H_ -#define WEBRTC_BASE_HELPERS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/helpers.h" - -#endif // WEBRTC_BASE_HELPERS_H_ diff --git a/webrtc/base/httpbase.h b/webrtc/base/httpbase.h deleted file mode 100644 index a66ce15a7f..0000000000 --- a/webrtc/base/httpbase.h +++ /dev/null @@ -1,20 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - - -#ifndef WEBRTC_BASE_HTTPBASE_H_ -#define WEBRTC_BASE_HTTPBASE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/httpbase.h" - -#endif // WEBRTC_BASE_HTTPBASE_H_ diff --git a/webrtc/base/httpcommon-inl.h b/webrtc/base/httpcommon-inl.h deleted file mode 100644 index 7dfe18242d..0000000000 --- a/webrtc/base/httpcommon-inl.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_HTTPCOMMON_INL_H_ -#define WEBRTC_BASE_HTTPCOMMON_INL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/httpcommon-inl.h" - -#endif // WEBRTC_BASE_HTTPCOMMON_INL_H_ diff --git a/webrtc/base/httpcommon.h b/webrtc/base/httpcommon.h deleted file mode 100644 index 3946dfcd77..0000000000 --- a/webrtc/base/httpcommon.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_HTTPCOMMON_H_ -#define WEBRTC_BASE_HTTPCOMMON_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/httpcommon.h" - -#endif // WEBRTC_BASE_HTTPCOMMON_H_ diff --git a/webrtc/base/httpserver.h b/webrtc/base/httpserver.h deleted file mode 100644 index 4fd75a2a05..0000000000 --- a/webrtc/base/httpserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_HTTPSERVER_H_ -#define WEBRTC_BASE_HTTPSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/httpserver.h" - -#endif // WEBRTC_BASE_HTTPSERVER_H_ diff --git a/webrtc/base/ifaddrs-android.h b/webrtc/base/ifaddrs-android.h deleted file mode 100644 index 9c49c9ffb0..0000000000 --- a/webrtc/base/ifaddrs-android.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2013 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_IFADDRS_ANDROID_H_ -#define WEBRTC_BASE_IFADDRS_ANDROID_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/ifaddrs-android.h" - -#endif // WEBRTC_BASE_IFADDRS_ANDROID_H_ diff --git a/webrtc/base/ifaddrs_converter.h b/webrtc/base/ifaddrs_converter.h deleted file mode 100644 index de7ad87eee..0000000000 --- a/webrtc/base/ifaddrs_converter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_IFADDRS_CONVERTER_H_ -#define WEBRTC_BASE_IFADDRS_CONVERTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/ifaddrs_converter.h" - -#endif // WEBRTC_BASE_IFADDRS_CONVERTER_H_ diff --git a/webrtc/base/ignore_wundef.h b/webrtc/base/ignore_wundef.h deleted file mode 100644 index fdfba9b84a..0000000000 --- a/webrtc/base/ignore_wundef.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_IGNORE_WUNDEF_H_ -#define WEBRTC_BASE_IGNORE_WUNDEF_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/ignore_wundef.h" - -#endif // WEBRTC_BASE_IGNORE_WUNDEF_H_ diff --git a/webrtc/base/ipaddress.h b/webrtc/base/ipaddress.h deleted file mode 100644 index 44e432d2c8..0000000000 --- a/webrtc/base/ipaddress.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2011 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_IPADDRESS_H_ -#define WEBRTC_BASE_IPADDRESS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/ipaddress.h" - -#endif // WEBRTC_BASE_IPADDRESS_H_ diff --git a/webrtc/base/json.h b/webrtc/base/json.h deleted file mode 100644 index 175028f607..0000000000 --- a/webrtc/base/json.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_JSON_H_ -#define WEBRTC_BASE_JSON_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/json.h" - -#endif // WEBRTC_BASE_JSON_H_ diff --git a/webrtc/base/keep_ref_until_done.h b/webrtc/base/keep_ref_until_done.h deleted file mode 100644 index 171e04886d..0000000000 --- a/webrtc/base/keep_ref_until_done.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_ -#define WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/keep_ref_until_done.h" - -#endif // WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_ diff --git a/webrtc/base/location.h b/webrtc/base/location.h deleted file mode 100644 index 432471c013..0000000000 --- a/webrtc/base/location.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_LOCATION_H_ -#define WEBRTC_BASE_LOCATION_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/location.h" - -#endif // WEBRTC_BASE_LOCATION_H_ diff --git a/webrtc/base/logging.h b/webrtc/base/logging.h deleted file mode 100644 index 594d9c992a..0000000000 --- a/webrtc/base/logging.h +++ /dev/null @@ -1,54 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// LOG(...) an ostream target that can be used to send formatted -// output to a variety of logging targets, such as debugger console, stderr, -// or any LogSink. -// The severity level passed as the first argument to the LOGging -// functions is used as a filter, to limit the verbosity of the logging. -// Static members of LogMessage documented below are used to control the -// verbosity and target of the output. -// There are several variations on the LOG macro which facilitate logging -// of common error conditions, detailed below. - -// LOG(sev) logs the given stream at severity "sev", which must be a -// compile-time constant of the LoggingSeverity type, without the namespace -// prefix. -// LOG_V(sev) Like LOG(), but sev is a run-time variable of the LoggingSeverity -// type (basically, it just doesn't prepend the namespace). -// LOG_F(sev) Like LOG(), but includes the name of the current function. -// LOG_T(sev) Like LOG(), but includes the this pointer. -// LOG_T_F(sev) Like LOG_F(), but includes the this pointer. -// LOG_GLE(M)(sev [, mod]) attempt to add a string description of the -// HRESULT returned by GetLastError. The "M" variant allows searching of a -// DLL's string table for the error description. -// LOG_ERRNO(sev) attempts to add a string description of an errno-derived -// error. errno and associated facilities exist on both Windows and POSIX, -// but on Windows they only apply to the C/C++ runtime. -// LOG_ERR(sev) is an alias for the platform's normal error system, i.e. _GLE on -// Windows and _ERRNO on POSIX. -// (The above three also all have _EX versions that let you specify the error -// code, rather than using the last one.) -// LOG_E(sev, ctx, err, ...) logs a detailed error interpreted using the -// specified context. -// LOG_CHECK_LEVEL(sev) (and LOG_CHECK_LEVEL_V(sev)) can be used as a test -// before performing expensive or sensitive operations whose sole purpose is -// to output logging data at the desired level. -// Lastly, PLOG(sev, err) is an alias for LOG_ERR_EX. - -#ifndef WEBRTC_BASE_LOGGING_H_ -#define WEBRTC_BASE_LOGGING_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/logging.h" - -#endif // WEBRTC_BASE_LOGGING_H_ diff --git a/webrtc/base/logsinks.h b/webrtc/base/logsinks.h deleted file mode 100644 index 95e6dc6154..0000000000 --- a/webrtc/base/logsinks.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_LOGSINKS_H_ -#define WEBRTC_BASE_LOGSINKS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/logsinks.h" - -#endif // WEBRTC_BASE_LOGSINKS_H_ diff --git a/webrtc/base/macutils.h b/webrtc/base/macutils.h deleted file mode 100644 index ed0c4f5473..0000000000 --- a/webrtc/base/macutils.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2007 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_MACUTILS_H_ -#define WEBRTC_BASE_MACUTILS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/macutils.h" - -#endif // WEBRTC_BASE_MACUTILS_H_ diff --git a/webrtc/base/mathutils.h b/webrtc/base/mathutils.h deleted file mode 100644 index 9e5c3cab78..0000000000 --- a/webrtc/base/mathutils.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2005 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_MATHUTILS_H_ -#define WEBRTC_BASE_MATHUTILS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/mathutils.h" - -#endif // WEBRTC_BASE_MATHUTILS_H_ diff --git a/webrtc/base/md5.h b/webrtc/base/md5.h deleted file mode 100644 index fd17541960..0000000000 --- a/webrtc/base/md5.h +++ /dev/null @@ -1,31 +0,0 @@ -/* - * This is the header file for the MD5 message-digest algorithm. - * The algorithm is due to Ron Rivest. This code was - * written by Colin Plumb in 1993, no copyright is claimed. - * This code is in the public domain; do with it what you wish. - * - * Equivalent code is available from RSA Data Security, Inc. - * This code has been tested against that, and is equivalent, - * except that you don't need to include two pages of legalese - * with every copy. - * To compute the message digest of a chunk of bytes, declare an - * MD5Context structure, pass it to MD5Init, call MD5Update as - * needed on buffers full of bytes, and then call MD5Final, which - * will fill a supplied 16-byte array with the digest. - * - */ - -// Changes(fbarchard): Ported to C++ and Google style guide. -// Made context first parameter in MD5Final for consistency with Sha1. -// Changes(hellner): added rtc namespace -// Changes(pbos): Reverted types back to uint32(8)_t with _t suffix. - -#ifndef WEBRTC_BASE_MD5_H_ -#define WEBRTC_BASE_MD5_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/md5.h" - -#endif // WEBRTC_BASE_MD5_H_ diff --git a/webrtc/base/md5digest.h b/webrtc/base/md5digest.h deleted file mode 100644 index 66d6ee1820..0000000000 --- a/webrtc/base/md5digest.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_MD5DIGEST_H_ -#define WEBRTC_BASE_MD5DIGEST_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/md5digest.h" - -#endif // WEBRTC_BASE_MD5DIGEST_H_ diff --git a/webrtc/base/memory_usage.h b/webrtc/base/memory_usage.h deleted file mode 100644 index 5c225597a7..0000000000 --- a/webrtc/base/memory_usage.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#ifndef WEBRTC_BASE_MEMORY_USAGE_H_ -#define WEBRTC_BASE_MEMORY_USAGE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/memory_usage.h" - -#endif // WEBRTC_BASE_MEMORY_USAGE_H_ diff --git a/webrtc/base/messagedigest.h b/webrtc/base/messagedigest.h deleted file mode 100644 index b73f9079c8..0000000000 --- a/webrtc/base/messagedigest.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_MESSAGEDIGEST_H_ -#define WEBRTC_BASE_MESSAGEDIGEST_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/messagedigest.h" - -#endif // WEBRTC_BASE_MESSAGEDIGEST_H_ diff --git a/webrtc/base/messagehandler.h b/webrtc/base/messagehandler.h deleted file mode 100644 index 943d0d7d9b..0000000000 --- a/webrtc/base/messagehandler.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_MESSAGEHANDLER_H_ -#define WEBRTC_BASE_MESSAGEHANDLER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/messagehandler.h" - -#endif // WEBRTC_BASE_MESSAGEHANDLER_H_ diff --git a/webrtc/base/messagequeue.h b/webrtc/base/messagequeue.h deleted file mode 100644 index 353a4b7725..0000000000 --- a/webrtc/base/messagequeue.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_MESSAGEQUEUE_H_ -#define WEBRTC_BASE_MESSAGEQUEUE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/messagequeue.h" - -#endif // WEBRTC_BASE_MESSAGEQUEUE_H_ diff --git a/webrtc/base/mod_ops.h b/webrtc/base/mod_ops.h deleted file mode 100644 index d61bd055e7..0000000000 --- a/webrtc/base/mod_ops.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_MOD_OPS_H_ -#define WEBRTC_BASE_MOD_OPS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/mod_ops.h" - -#endif // WEBRTC_BASE_MOD_OPS_H_ diff --git a/webrtc/base/natserver.h b/webrtc/base/natserver.h deleted file mode 100644 index b803ad8587..0000000000 --- a/webrtc/base/natserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NATSERVER_H_ -#define WEBRTC_BASE_NATSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/natserver.h" - -#endif // WEBRTC_BASE_NATSERVER_H_ diff --git a/webrtc/base/natsocketfactory.h b/webrtc/base/natsocketfactory.h deleted file mode 100644 index 31c29ab277..0000000000 --- a/webrtc/base/natsocketfactory.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NATSOCKETFACTORY_H_ -#define WEBRTC_BASE_NATSOCKETFACTORY_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/natsocketfactory.h" - -#endif // WEBRTC_BASE_NATSOCKETFACTORY_H_ diff --git a/webrtc/base/nattypes.h b/webrtc/base/nattypes.h deleted file mode 100644 index 001f57fe7d..0000000000 --- a/webrtc/base/nattypes.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NATTYPES_H_ -#define WEBRTC_BASE_NATTYPES_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/nattypes.h" - -#endif // WEBRTC_BASE_NATTYPES_H_ diff --git a/webrtc/base/nethelpers.h b/webrtc/base/nethelpers.h deleted file mode 100644 index 9a8e6073dd..0000000000 --- a/webrtc/base/nethelpers.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2008 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NETHELPERS_H_ -#define WEBRTC_BASE_NETHELPERS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/nethelpers.h" - -#endif // WEBRTC_BASE_NETHELPERS_H_ diff --git a/webrtc/base/network.h b/webrtc/base/network.h deleted file mode 100644 index 29530987c1..0000000000 --- a/webrtc/base/network.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NETWORK_H_ -#define WEBRTC_BASE_NETWORK_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/network.h" - -#endif // WEBRTC_BASE_NETWORK_H_ diff --git a/webrtc/base/networkmonitor.h b/webrtc/base/networkmonitor.h deleted file mode 100644 index 290da4f48e..0000000000 --- a/webrtc/base/networkmonitor.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NETWORKMONITOR_H_ -#define WEBRTC_BASE_NETWORKMONITOR_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/networkmonitor.h" - -#endif // WEBRTC_BASE_NETWORKMONITOR_H_ diff --git a/webrtc/base/networkroute.h b/webrtc/base/networkroute.h deleted file mode 100644 index b5e8c13842..0000000000 --- a/webrtc/base/networkroute.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NETWORKROUTE_H_ -#define WEBRTC_BASE_NETWORKROUTE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/networkroute.h" - -#endif // WEBRTC_BASE_NETWORKROUTE_H_ diff --git a/webrtc/base/nullsocketserver.h b/webrtc/base/nullsocketserver.h deleted file mode 100644 index 214c542b5f..0000000000 --- a/webrtc/base/nullsocketserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NULLSOCKETSERVER_H_ -#define WEBRTC_BASE_NULLSOCKETSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/nullsocketserver.h" - -#endif // WEBRTC_BASE_NULLSOCKETSERVER_H_ diff --git a/webrtc/base/numerics/exp_filter.h b/webrtc/base/numerics/exp_filter.h deleted file mode 100644 index a4eaea2c91..0000000000 --- a/webrtc/base/numerics/exp_filter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NUMERICS_EXP_FILTER_H_ -#define WEBRTC_BASE_NUMERICS_EXP_FILTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/numerics/exp_filter.h" - -#endif // WEBRTC_BASE_NUMERICS_EXP_FILTER_H_ diff --git a/webrtc/base/numerics/percentile_filter.h b/webrtc/base/numerics/percentile_filter.h deleted file mode 100644 index a9058a2b4c..0000000000 --- a/webrtc/base/numerics/percentile_filter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_ -#define WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/numerics/percentile_filter.h" - -#endif // WEBRTC_BASE_NUMERICS_PERCENTILE_FILTER_H_ diff --git a/webrtc/base/onetimeevent.h b/webrtc/base/onetimeevent.h deleted file mode 100644 index 6849bac581..0000000000 --- a/webrtc/base/onetimeevent.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ONETIMEEVENT_H_ -#define WEBRTC_BASE_ONETIMEEVENT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/onetimeevent.h" - -#endif // WEBRTC_BASE_ONETIMEEVENT_H_ diff --git a/webrtc/base/openssl.h b/webrtc/base/openssl.h deleted file mode 100644 index 795af70321..0000000000 --- a/webrtc/base/openssl.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2013 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_OPENSSL_H_ -#define WEBRTC_BASE_OPENSSL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/openssl.h" - -#endif // WEBRTC_BASE_OPENSSL_H_ diff --git a/webrtc/base/openssladapter.h b/webrtc/base/openssladapter.h deleted file mode 100644 index 6444215098..0000000000 --- a/webrtc/base/openssladapter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_OPENSSLADAPTER_H_ -#define WEBRTC_BASE_OPENSSLADAPTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/openssladapter.h" - -#endif // WEBRTC_BASE_OPENSSLADAPTER_H_ diff --git a/webrtc/base/openssldigest.h b/webrtc/base/openssldigest.h deleted file mode 100644 index 031c0b1cb0..0000000000 --- a/webrtc/base/openssldigest.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_OPENSSLDIGEST_H_ -#define WEBRTC_BASE_OPENSSLDIGEST_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/openssldigest.h" - -#endif // WEBRTC_BASE_OPENSSLDIGEST_H_ diff --git a/webrtc/base/opensslidentity.h b/webrtc/base/opensslidentity.h deleted file mode 100644 index 59fa571ce5..0000000000 --- a/webrtc/base/opensslidentity.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_OPENSSLIDENTITY_H_ -#define WEBRTC_BASE_OPENSSLIDENTITY_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/opensslidentity.h" - -#endif // WEBRTC_BASE_OPENSSLIDENTITY_H_ diff --git a/webrtc/base/opensslstreamadapter.h b/webrtc/base/opensslstreamadapter.h deleted file mode 100644 index e17e029ffe..0000000000 --- a/webrtc/base/opensslstreamadapter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_ -#define WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/opensslstreamadapter.h" - -#endif // WEBRTC_BASE_OPENSSLSTREAMADAPTER_H_ diff --git a/webrtc/base/optional.h b/webrtc/base/optional.h deleted file mode 100644 index 7657ec3366..0000000000 --- a/webrtc/base/optional.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_OPTIONAL_H_ -#define WEBRTC_BASE_OPTIONAL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/optional.h" - -#endif // WEBRTC_BASE_OPTIONAL_H_ diff --git a/webrtc/base/optionsfile.h b/webrtc/base/optionsfile.h deleted file mode 100644 index e77fd8adfc..0000000000 --- a/webrtc/base/optionsfile.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2008 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_OPTIONSFILE_H_ -#define WEBRTC_BASE_OPTIONSFILE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/optionsfile.h" - -#endif // WEBRTC_BASE_OPTIONSFILE_H_ diff --git a/webrtc/base/pathutils.h b/webrtc/base/pathutils.h deleted file mode 100644 index b45ca04f7c..0000000000 --- a/webrtc/base/pathutils.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_PATHUTILS_H_ -#define WEBRTC_BASE_PATHUTILS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/pathutils.h" - -#endif // WEBRTC_BASE_PATHUTILS_H_ diff --git a/webrtc/base/physicalsocketserver.h b/webrtc/base/physicalsocketserver.h deleted file mode 100644 index 63e6dfa5b9..0000000000 --- a/webrtc/base/physicalsocketserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_PHYSICALSOCKETSERVER_H_ -#define WEBRTC_BASE_PHYSICALSOCKETSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/physicalsocketserver.h" - -#endif // WEBRTC_BASE_PHYSICALSOCKETSERVER_H_ diff --git a/webrtc/base/platform_file.h b/webrtc/base/platform_file.h deleted file mode 100644 index c7396ec4c7..0000000000 --- a/webrtc/base/platform_file.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2014 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_PLATFORM_FILE_H_ -#define WEBRTC_BASE_PLATFORM_FILE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/platform_file.h" - -#endif // WEBRTC_BASE_PLATFORM_FILE_H_ diff --git a/webrtc/base/platform_thread.h b/webrtc/base/platform_thread.h deleted file mode 100644 index 626d66fc07..0000000000 --- a/webrtc/base/platform_thread.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_PLATFORM_THREAD_H_ -#define WEBRTC_BASE_PLATFORM_THREAD_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/platform_thread.h" - -#endif // WEBRTC_BASE_PLATFORM_THREAD_H_ diff --git a/webrtc/base/platform_thread_types.h b/webrtc/base/platform_thread_types.h deleted file mode 100644 index f2dbd58363..0000000000 --- a/webrtc/base/platform_thread_types.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_ -#define WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/platform_thread_types.h" - -#endif // WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_ diff --git a/webrtc/base/protobuf_utils.h b/webrtc/base/protobuf_utils.h deleted file mode 100644 index 3d2dd862ff..0000000000 --- a/webrtc/base/protobuf_utils.h +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include - -#ifndef WEBRTC_BASE_PROTOBUF_UTILS_H_ -#define WEBRTC_BASE_PROTOBUF_UTILS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/protobuf_utils.h" - -#endif // WEBRTC_BASE_PROTOBUF_UTILS_H_ diff --git a/webrtc/base/proxyinfo.h b/webrtc/base/proxyinfo.h deleted file mode 100644 index f0ae1825e4..0000000000 --- a/webrtc/base/proxyinfo.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_PROXYINFO_H_ -#define WEBRTC_BASE_PROXYINFO_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/proxyinfo.h" - -#endif // WEBRTC_BASE_PROXYINFO_H_ diff --git a/webrtc/base/proxyserver.h b/webrtc/base/proxyserver.h deleted file mode 100644 index 1bf580ad70..0000000000 --- a/webrtc/base/proxyserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_PROXYSERVER_H_ -#define WEBRTC_BASE_PROXYSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/proxyserver.h" - -#endif // WEBRTC_BASE_PROXYSERVER_H_ diff --git a/webrtc/base/ptr_util.h b/webrtc/base/ptr_util.h deleted file mode 100644 index aa6f3b4016..0000000000 --- a/webrtc/base/ptr_util.h +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright 2017 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// This implementation is borrowed from chromium. - -#ifndef WEBRTC_BASE_PTR_UTIL_H_ -#define WEBRTC_BASE_PTR_UTIL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/ptr_util.h" - -#endif // WEBRTC_BASE_PTR_UTIL_H_ diff --git a/webrtc/base/race_checker.h b/webrtc/base/race_checker.h deleted file mode 100644 index 474fdb59ba..0000000000 --- a/webrtc/base/race_checker.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_RACE_CHECKER_H_ -#define WEBRTC_BASE_RACE_CHECKER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/race_checker.h" - -#endif // WEBRTC_BASE_RACE_CHECKER_H_ diff --git a/webrtc/base/random.h b/webrtc/base/random.h deleted file mode 100644 index 12a490202b..0000000000 --- a/webrtc/base/random.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_RANDOM_H_ -#define WEBRTC_BASE_RANDOM_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/random.h" - -#endif // WEBRTC_BASE_RANDOM_H_ diff --git a/webrtc/base/rate_limiter.h b/webrtc/base/rate_limiter.h deleted file mode 100644 index 0cba5fb9a9..0000000000 --- a/webrtc/base/rate_limiter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_RATE_LIMITER_H_ -#define WEBRTC_BASE_RATE_LIMITER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/rate_limiter.h" - -#endif // WEBRTC_BASE_RATE_LIMITER_H_ diff --git a/webrtc/base/rate_statistics.h b/webrtc/base/rate_statistics.h deleted file mode 100644 index 1a17500727..0000000000 --- a/webrtc/base/rate_statistics.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_RATE_STATISTICS_H_ -#define WEBRTC_BASE_RATE_STATISTICS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/rate_statistics.h" - -#endif // WEBRTC_BASE_RATE_STATISTICS_H_ diff --git a/webrtc/base/ratelimiter.h b/webrtc/base/ratelimiter.h deleted file mode 100644 index 0e372db691..0000000000 --- a/webrtc/base/ratelimiter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_RATELIMITER_H_ -#define WEBRTC_BASE_RATELIMITER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/ratelimiter.h" - -#endif // WEBRTC_BASE_RATELIMITER_H_ diff --git a/webrtc/base/ratetracker.h b/webrtc/base/ratetracker.h deleted file mode 100644 index d1fd75d0ee..0000000000 --- a/webrtc/base/ratetracker.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_RATETRACKER_H_ -#define WEBRTC_BASE_RATETRACKER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/ratetracker.h" - -#endif // WEBRTC_BASE_RATETRACKER_H_ diff --git a/webrtc/base/refcount.h b/webrtc/base/refcount.h deleted file mode 100644 index 4a7cea313f..0000000000 --- a/webrtc/base/refcount.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * Copyright 2011 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#ifndef WEBRTC_BASE_REFCOUNT_H_ -#define WEBRTC_BASE_REFCOUNT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/refcount.h" - -#endif // WEBRTC_BASE_REFCOUNT_H_ diff --git a/webrtc/base/refcountedobject.h b/webrtc/base/refcountedobject.h deleted file mode 100644 index 78304fa5f5..0000000000 --- a/webrtc/base/refcountedobject.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#ifndef WEBRTC_BASE_REFCOUNTEDOBJECT_H_ -#define WEBRTC_BASE_REFCOUNTEDOBJECT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/refcountedobject.h" - -#endif // WEBRTC_BASE_REFCOUNTEDOBJECT_H_ diff --git a/webrtc/base/rollingaccumulator.h b/webrtc/base/rollingaccumulator.h deleted file mode 100644 index a7de4f19dd..0000000000 --- a/webrtc/base/rollingaccumulator.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2011 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_ROLLINGACCUMULATOR_H_ -#define WEBRTC_BASE_ROLLINGACCUMULATOR_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/rollingaccumulator.h" - -#endif // WEBRTC_BASE_ROLLINGACCUMULATOR_H_ diff --git a/webrtc/base/rtccertificate.h b/webrtc/base/rtccertificate.h deleted file mode 100644 index 22d8fe754b..0000000000 --- a/webrtc/base/rtccertificate.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2015 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_RTCCERTIFICATE_H_ -#define WEBRTC_BASE_RTCCERTIFICATE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/rtccertificate.h" - -#endif // WEBRTC_BASE_RTCCERTIFICATE_H_ diff --git a/webrtc/base/rtccertificategenerator.h b/webrtc/base/rtccertificategenerator.h deleted file mode 100644 index fac1cec9ef..0000000000 --- a/webrtc/base/rtccertificategenerator.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_ -#define WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/rtccertificategenerator.h" - -#endif // WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_ diff --git a/webrtc/base/safe_compare.h b/webrtc/base/safe_compare.h deleted file mode 100644 index acdd9cebd7..0000000000 --- a/webrtc/base/safe_compare.h +++ /dev/null @@ -1,39 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// This file defines six constexpr functions: -// -// rtc::SafeEq // == -// rtc::SafeNe // != -// rtc::SafeLt // < -// rtc::SafeLe // <= -// rtc::SafeGt // > -// rtc::SafeGe // >= -// -// They each accept two arguments of arbitrary types, and in almost all cases, -// they simply call the appropriate comparison operator. However, if both -// arguments are integers, they don't compare them using C++'s quirky rules, -// but instead adhere to the true mathematical definitions. It is as if the -// arguments were first converted to infinite-range signed integers, and then -// compared, although of course nothing expensive like that actually takes -// place. In practice, for signed/signed and unsigned/unsigned comparisons and -// some mixed-signed comparisons with a compile-time constant, the overhead is -// zero; in the remaining cases, it is just a few machine instructions (no -// branches). - -#ifndef WEBRTC_BASE_SAFE_COMPARE_H_ -#define WEBRTC_BASE_SAFE_COMPARE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/safe_compare.h" - -#endif // WEBRTC_BASE_SAFE_COMPARE_H_ diff --git a/webrtc/base/safe_conversions.h b/webrtc/base/safe_conversions.h deleted file mode 100644 index ac0bb651f3..0000000000 --- a/webrtc/base/safe_conversions.h +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright 2014 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Borrowed from Chromium's src/base/numerics/safe_conversions.h. - -#ifndef WEBRTC_BASE_SAFE_CONVERSIONS_H_ -#define WEBRTC_BASE_SAFE_CONVERSIONS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/safe_conversions.h" - -#endif // WEBRTC_BASE_SAFE_CONVERSIONS_H_ diff --git a/webrtc/base/safe_conversions_impl.h b/webrtc/base/safe_conversions_impl.h deleted file mode 100644 index 497e156dbb..0000000000 --- a/webrtc/base/safe_conversions_impl.h +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright 2014 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Borrowed from Chromium's src/base/numerics/safe_conversions_impl.h. - -#ifndef WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_ -#define WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/safe_conversions_impl.h" - -#endif // WEBRTC_BASE_SAFE_CONVERSIONS_IMPL_H_ diff --git a/webrtc/base/safe_minmax.h b/webrtc/base/safe_minmax.h deleted file mode 100644 index 54d99b720b..0000000000 --- a/webrtc/base/safe_minmax.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * Copyright 2017 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SAFE_MINMAX_H_ -#define WEBRTC_BASE_SAFE_MINMAX_H_ - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/safe_minmax.h" - -#endif // WEBRTC_BASE_SAFE_MINMAX_H_ diff --git a/webrtc/base/sanitizer.h b/webrtc/base/sanitizer.h deleted file mode 100644 index 56a5e103f7..0000000000 --- a/webrtc/base/sanitizer.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SANITIZER_H_ -#define WEBRTC_BASE_SANITIZER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sanitizer.h" - -#endif // WEBRTC_BASE_SANITIZER_H_ diff --git a/webrtc/base/scoped_ref_ptr.h b/webrtc/base/scoped_ref_ptr.h deleted file mode 100644 index 259956292f..0000000000 --- a/webrtc/base/scoped_ref_ptr.h +++ /dev/null @@ -1,71 +0,0 @@ -/* - * Copyright 2011 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Originally these classes are from Chromium. -// http://src.chromium.org/viewvc/chrome/trunk/src/base/memory/ref_counted.h?view=markup - -// -// A smart pointer class for reference counted objects. Use this class instead -// of calling AddRef and Release manually on a reference counted object to -// avoid common memory leaks caused by forgetting to Release an object -// reference. Sample usage: -// -// class MyFoo : public RefCounted { -// ... -// }; -// -// void some_function() { -// scoped_refptr foo = new MyFoo(); -// foo->Method(param); -// // |foo| is released when this function returns -// } -// -// void some_other_function() { -// scoped_refptr foo = new MyFoo(); -// ... -// foo = nullptr; // explicitly releases |foo| -// ... -// if (foo) -// foo->Method(param); -// } -// -// The above examples show how scoped_refptr acts like a pointer to T. -// Given two scoped_refptr classes, it is also possible to exchange -// references between the two objects, like so: -// -// { -// scoped_refptr a = new MyFoo(); -// scoped_refptr b; -// -// b.swap(a); -// // now, |b| references the MyFoo object, and |a| references null. -// } -// -// To make both |a| and |b| in the above example reference the same MyFoo -// object, simply use the assignment operator: -// -// { -// scoped_refptr a = new MyFoo(); -// scoped_refptr b; -// -// b = a; -// // now, |a| and |b| each own a reference to the same MyFoo object. -// } -// - -#ifndef WEBRTC_BASE_SCOPED_REF_PTR_H_ -#define WEBRTC_BASE_SCOPED_REF_PTR_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/scoped_ref_ptr.h" - -#endif // WEBRTC_BASE_SCOPED_REF_PTR_H_ diff --git a/webrtc/base/sequenced_task_checker.h b/webrtc/base/sequenced_task_checker.h deleted file mode 100644 index e586b8d6da..0000000000 --- a/webrtc/base/sequenced_task_checker.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_ -#define WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sequenced_task_checker.h" - -#endif // WEBRTC_BASE_SEQUENCED_TASK_CHECKER_H_ diff --git a/webrtc/base/sequenced_task_checker_impl.h b/webrtc/base/sequenced_task_checker_impl.h deleted file mode 100644 index 4972539e66..0000000000 --- a/webrtc/base/sequenced_task_checker_impl.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_ -#define WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sequenced_task_checker_impl.h" - -#endif // WEBRTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_ diff --git a/webrtc/base/sha1.h b/webrtc/base/sha1.h deleted file mode 100644 index fde3e598c3..0000000000 --- a/webrtc/base/sha1.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * SHA-1 in C - * By Steve Reid - * 100% Public Domain - * -*/ - -// Ported to C++, Google style, under namespace rtc. - -#ifndef WEBRTC_BASE_SHA1_H_ -#define WEBRTC_BASE_SHA1_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sha1.h" - -#endif // WEBRTC_BASE_SHA1_H_ diff --git a/webrtc/base/sha1digest.h b/webrtc/base/sha1digest.h deleted file mode 100644 index e3b4ef840b..0000000000 --- a/webrtc/base/sha1digest.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SHA1DIGEST_H_ -#define WEBRTC_BASE_SHA1DIGEST_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sha1digest.h" - -#endif // WEBRTC_BASE_SHA1DIGEST_H_ diff --git a/webrtc/base/sigslot.h b/webrtc/base/sigslot.h deleted file mode 100644 index 9d31441a49..0000000000 --- a/webrtc/base/sigslot.h +++ /dev/null @@ -1,104 +0,0 @@ -// sigslot.h: Signal/Slot classes -// -// Written by Sarah Thompson (sarah@telergy.com) 2002. -// -// License: Public domain. You are free to use this code however you like, with -// the proviso that the author takes on no responsibility or liability for any -// use. -// -// QUICK DOCUMENTATION -// -// (see also the full documentation at http://sigslot.sourceforge.net/) -// -// #define switches -// SIGSLOT_PURE_ISO: -// Define this to force ISO C++ compliance. This also disables all of -// the thread safety support on platforms where it is available. -// -// SIGSLOT_USE_POSIX_THREADS: -// Force use of Posix threads when using a C++ compiler other than gcc -// on a platform that supports Posix threads. (When using gcc, this is -// the default - use SIGSLOT_PURE_ISO to disable this if necessary) -// -// SIGSLOT_DEFAULT_MT_POLICY: -// Where thread support is enabled, this defaults to -// multi_threaded_global. Otherwise, the default is single_threaded. -// #define this yourself to override the default. In pure ISO mode, -// anything other than single_threaded will cause a compiler error. -// -// PLATFORM NOTES -// -// Win32: -// On Win32, the WEBRTC_WIN symbol must be #defined. Most mainstream -// compilers do this by default, but you may need to define it yourself -// if your build environment is less standard. This causes the Win32 -// thread support to be compiled in and used automatically. -// -// Unix/Linux/BSD, etc.: -// If you're using gcc, it is assumed that you have Posix threads -// available, so they are used automatically. You can override this (as -// under Windows) with the SIGSLOT_PURE_ISO switch. If you're using -// something other than gcc but still want to use Posix threads, you -// need to #define SIGSLOT_USE_POSIX_THREADS. -// -// ISO C++: -// If none of the supported platforms are detected, or if -// SIGSLOT_PURE_ISO is defined, all multithreading support is turned -// off, along with any code that might cause a pure ISO C++ environment -// to complain. Before you ask, gcc -ansi -pedantic won't compile this -// library, but gcc -ansi is fine. Pedantic mode seems to throw a lot of -// errors that aren't really there. If you feel like investigating this, -// please contact the author. -// -// -// THREADING MODES -// -// single_threaded: -// Your program is assumed to be single threaded from the point of view -// of signal/slot usage (i.e. all objects using signals and slots are -// created and destroyed from a single thread). Behaviour if objects are -// destroyed concurrently is undefined (i.e. you'll get the occasional -// segmentation fault/memory exception). -// -// multi_threaded_global: -// Your program is assumed to be multi threaded. Objects using signals -// and slots can be safely created and destroyed from any thread, even -// when connections exist. In multi_threaded_global mode, this is -// achieved by a single global mutex (actually a critical section on -// Windows because they are faster). This option uses less OS resources, -// but results in more opportunities for contention, possibly resulting -// in more context switches than are strictly necessary. -// -// multi_threaded_local: -// Behaviour in this mode is essentially the same as -// multi_threaded_global, except that each signal, and each object that -// inherits has_slots, all have their own mutex/critical section. In -// practice, this means that mutex collisions (and hence context -// switches) only happen if they are absolutely essential. However, on -// some platforms, creating a lot of mutexes can slow down the whole OS, -// so use this option with care. -// -// USING THE LIBRARY -// -// See the full documentation at http://sigslot.sourceforge.net/ -// -// Libjingle specific: -// -// This file has been modified such that has_slots and signalx do not have to be -// using the same threading requirements. E.g. it is possible to connect a -// has_slots and signal0 or -// has_slots and signal0. -// If has_slots is single threaded the user must ensure that it is not trying -// to connect or disconnect to signalx concurrently or data race may occur. -// If signalx is single threaded the user must ensure that disconnect, connect -// or signal is not happening concurrently or data race may occur. - -#ifndef WEBRTC_BASE_SIGSLOT_H_ -#define WEBRTC_BASE_SIGSLOT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sigslot.h" - -#endif // WEBRTC_BASE_SIGSLOT_H_ diff --git a/webrtc/base/sigslottester.h b/webrtc/base/sigslottester.h deleted file mode 100644 index 545bf9e235..0000000000 --- a/webrtc/base/sigslottester.h +++ /dev/null @@ -1,23 +0,0 @@ -// This file was GENERATED by command: -// pump.py sigslottester.h.pump -// DO NOT EDIT BY HAND!!! - -/* - * Copyright 2014 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SIGSLOTTESTER_H_ -#define WEBRTC_BASE_SIGSLOTTESTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sigslottester.h" - -#endif // WEBRTC_BASE_SIGSLOTTESTER_H_ diff --git a/webrtc/base/socket.h b/webrtc/base/socket.h deleted file mode 100644 index 19ea7a032f..0000000000 --- a/webrtc/base/socket.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SOCKET_H_ -#define WEBRTC_BASE_SOCKET_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/socket.h" - -#endif // WEBRTC_BASE_SOCKET_H_ diff --git a/webrtc/base/socket_unittest.h b/webrtc/base/socket_unittest.h deleted file mode 100644 index f6769f9470..0000000000 --- a/webrtc/base/socket_unittest.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2009 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SOCKET_UNITTEST_H_ -#define WEBRTC_BASE_SOCKET_UNITTEST_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/socket_unittest.h" - -#endif // WEBRTC_BASE_SOCKET_UNITTEST_H_ diff --git a/webrtc/base/socketadapters.h b/webrtc/base/socketadapters.h deleted file mode 100644 index 7df0f3ae2f..0000000000 --- a/webrtc/base/socketadapters.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SOCKETADAPTERS_H_ -#define WEBRTC_BASE_SOCKETADAPTERS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/socketadapters.h" - -#endif // WEBRTC_BASE_SOCKETADAPTERS_H_ diff --git a/webrtc/base/socketaddress.h b/webrtc/base/socketaddress.h deleted file mode 100644 index 20199ad96b..0000000000 --- a/webrtc/base/socketaddress.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SOCKETADDRESS_H_ -#define WEBRTC_BASE_SOCKETADDRESS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/socketaddress.h" - -#endif // WEBRTC_BASE_SOCKETADDRESS_H_ diff --git a/webrtc/base/socketaddresspair.h b/webrtc/base/socketaddresspair.h deleted file mode 100644 index 3f53f10fee..0000000000 --- a/webrtc/base/socketaddresspair.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SOCKETADDRESSPAIR_H_ -#define WEBRTC_BASE_SOCKETADDRESSPAIR_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/socketaddresspair.h" - -#endif // WEBRTC_BASE_SOCKETADDRESSPAIR_H_ diff --git a/webrtc/base/socketfactory.h b/webrtc/base/socketfactory.h deleted file mode 100644 index 3a829ac10d..0000000000 --- a/webrtc/base/socketfactory.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SOCKETFACTORY_H_ -#define WEBRTC_BASE_SOCKETFACTORY_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/socketfactory.h" - -#endif // WEBRTC_BASE_SOCKETFACTORY_H_ diff --git a/webrtc/base/socketserver.h b/webrtc/base/socketserver.h deleted file mode 100644 index 55b427da7e..0000000000 --- a/webrtc/base/socketserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SOCKETSERVER_H_ -#define WEBRTC_BASE_SOCKETSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/socketserver.h" - -#endif // WEBRTC_BASE_SOCKETSERVER_H_ diff --git a/webrtc/base/socketstream.h b/webrtc/base/socketstream.h deleted file mode 100644 index a76ffb3814..0000000000 --- a/webrtc/base/socketstream.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2005 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SOCKETSTREAM_H_ -#define WEBRTC_BASE_SOCKETSTREAM_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/socketstream.h" - -#endif // WEBRTC_BASE_SOCKETSTREAM_H_ diff --git a/webrtc/base/ssladapter.h b/webrtc/base/ssladapter.h deleted file mode 100644 index 3d432ecd0c..0000000000 --- a/webrtc/base/ssladapter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SSLADAPTER_H_ -#define WEBRTC_BASE_SSLADAPTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/ssladapter.h" - -#endif // WEBRTC_BASE_SSLADAPTER_H_ diff --git a/webrtc/base/sslfingerprint.h b/webrtc/base/sslfingerprint.h deleted file mode 100644 index 6be82fd1b2..0000000000 --- a/webrtc/base/sslfingerprint.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2012 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SSLFINGERPRINT_H_ -#define WEBRTC_BASE_SSLFINGERPRINT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sslfingerprint.h" - -#endif // WEBRTC_BASE_SSLFINGERPRINT_H_ diff --git a/webrtc/base/sslidentity.h b/webrtc/base/sslidentity.h deleted file mode 100644 index 1cedfa09c1..0000000000 --- a/webrtc/base/sslidentity.h +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Handling of certificates and keypairs for SSLStreamAdapter's peer mode. - -#ifndef WEBRTC_BASE_SSLIDENTITY_H_ -#define WEBRTC_BASE_SSLIDENTITY_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sslidentity.h" - -#endif // WEBRTC_BASE_SSLIDENTITY_H_ diff --git a/webrtc/base/sslroots.h b/webrtc/base/sslroots.h deleted file mode 100644 index 9fa706b415..0000000000 --- a/webrtc/base/sslroots.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SSLROOTS_H_ -#define WEBRTC_BASE_SSLROOTS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sslroots.h" - -#endif // WEBRTC_BASE_SSLROOTS_H_ diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h deleted file mode 100644 index d7c062e4b8..0000000000 --- a/webrtc/base/sslstreamadapter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SSLSTREAMADAPTER_H_ -#define WEBRTC_BASE_SSLSTREAMADAPTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/sslstreamadapter.h" - -#endif // WEBRTC_BASE_SSLSTREAMADAPTER_H_ diff --git a/webrtc/base/stream.h b/webrtc/base/stream.h deleted file mode 100644 index 18dd865414..0000000000 --- a/webrtc/base/stream.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_STREAM_H_ -#define WEBRTC_BASE_STREAM_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/stream.h" - -#endif // WEBRTC_BASE_STREAM_H_ diff --git a/webrtc/base/string_to_number.h b/webrtc/base/string_to_number.h deleted file mode 100644 index fa88ba4da3..0000000000 --- a/webrtc/base/string_to_number.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2017 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_STRING_TO_NUMBER_H_ -#define WEBRTC_BASE_STRING_TO_NUMBER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/string_to_number.h" - -#endif // WEBRTC_BASE_STRING_TO_NUMBER_H_ diff --git a/webrtc/base/stringencode.h b/webrtc/base/stringencode.h deleted file mode 100644 index 27b810ea3c..0000000000 --- a/webrtc/base/stringencode.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_STRINGENCODE_H_ -#define WEBRTC_BASE_STRINGENCODE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/stringencode.h" - -#endif // WEBRTC_BASE_STRINGENCODE_H__ diff --git a/webrtc/base/stringize_macros.h b/webrtc/base/stringize_macros.h deleted file mode 100644 index 5f8a5b1b86..0000000000 --- a/webrtc/base/stringize_macros.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Modified from the Chromium original: -// src/base/strings/stringize_macros.h - -// This file defines preprocessor macros for stringizing preprocessor -// symbols (or their output) and manipulating preprocessor symbols -// that define strings. - -#ifndef WEBRTC_BASE_STRINGIZE_MACROS_H_ -#define WEBRTC_BASE_STRINGIZE_MACROS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/stringize_macros.h" - -#endif // WEBRTC_BASE_STRINGIZE_MACROS_H_ diff --git a/webrtc/base/stringutils.h b/webrtc/base/stringutils.h deleted file mode 100644 index e3b5e07822..0000000000 --- a/webrtc/base/stringutils.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_STRINGUTILS_H_ -#define WEBRTC_BASE_STRINGUTILS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/stringutils.h" - -#endif // WEBRTC_BASE_STRINGUTILS_H_ diff --git a/webrtc/base/swap_queue.h b/webrtc/base/swap_queue.h deleted file mode 100644 index 711114748f..0000000000 --- a/webrtc/base/swap_queue.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_SWAP_QUEUE_H_ -#define WEBRTC_BASE_SWAP_QUEUE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/swap_queue.h" - -#endif // WEBRTC_BASE_SWAP_QUEUE_H_ diff --git a/webrtc/base/task_queue.h b/webrtc/base/task_queue.h deleted file mode 100644 index 12f5cbbf9f..0000000000 --- a/webrtc/base/task_queue.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TASK_QUEUE_H_ -#define WEBRTC_BASE_TASK_QUEUE_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/task_queue.h" - -#endif // WEBRTC_BASE_TASK_QUEUE_H_ diff --git a/webrtc/base/task_queue_posix.h b/webrtc/base/task_queue_posix.h deleted file mode 100644 index 6cb51a03c6..0000000000 --- a/webrtc/base/task_queue_posix.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TASK_QUEUE_POSIX_H_ -#define WEBRTC_BASE_TASK_QUEUE_POSIX_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/task_queue_posix.h" - -#endif // WEBRTC_BASE_TASK_QUEUE_POSIX_H_ diff --git a/webrtc/base/template_util.h b/webrtc/base/template_util.h deleted file mode 100644 index 9a05643ddc..0000000000 --- a/webrtc/base/template_util.h +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Borrowed from Chromium's src/base/template_util.h. - -#ifndef WEBRTC_BASE_TEMPLATE_UTIL_H_ -#define WEBRTC_BASE_TEMPLATE_UTIL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/template_util.h" - -#endif // WEBRTC_BASE_TEMPLATE_UTIL_H_ diff --git a/webrtc/base/testbase64.h b/webrtc/base/testbase64.h deleted file mode 100644 index fc9846f1d8..0000000000 --- a/webrtc/base/testbase64.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TESTBASE64_H_ -#define WEBRTC_BASE_TESTBASE64_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/testbase64.h" - -#endif // WEBRTC_BASE_TESTBASE64_H_ diff --git a/webrtc/base/testclient.h b/webrtc/base/testclient.h deleted file mode 100644 index 378e2b81d4..0000000000 --- a/webrtc/base/testclient.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TESTCLIENT_H_ -#define WEBRTC_BASE_TESTCLIENT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/testclient.h" - -#endif // WEBRTC_BASE_TESTCLIENT_H_ diff --git a/webrtc/base/testechoserver.h b/webrtc/base/testechoserver.h deleted file mode 100644 index 21365e2a82..0000000000 --- a/webrtc/base/testechoserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TESTECHOSERVER_H_ -#define WEBRTC_BASE_TESTECHOSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/testechoserver.h" - -#endif // WEBRTC_BASE_TESTECHOSERVER_H_ diff --git a/webrtc/base/testutils.h b/webrtc/base/testutils.h deleted file mode 100644 index 74f216066e..0000000000 --- a/webrtc/base/testutils.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TESTUTILS_H_ -#define WEBRTC_BASE_TESTUTILS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/testutils.h" - -#endif // WEBRTC_BASE_TESTUTILS_H_ diff --git a/webrtc/base/thread.h b/webrtc/base/thread.h deleted file mode 100644 index 6a6887aa0c..0000000000 --- a/webrtc/base/thread.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_THREAD_H_ -#define WEBRTC_BASE_THREAD_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/thread.h" - -#endif // WEBRTC_BASE_THREAD_H_ diff --git a/webrtc/base/thread_annotations.h b/webrtc/base/thread_annotations.h deleted file mode 100644 index 5b94ffed85..0000000000 --- a/webrtc/base/thread_annotations.h +++ /dev/null @@ -1,27 +0,0 @@ -// -// Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. -// -// Use of this source code is governed by a BSD-style license -// that can be found in the LICENSE file in the root of the source -// tree. An additional intellectual property rights grant can be found -// in the file PATENTS. All contributing project authors may -// be found in the AUTHORS file in the root of the source tree. -// -// Borrowed from -// https://code.google.com/p/gperftools/source/browse/src/base/thread_annotations.h -// but adapted for clang attributes instead of the gcc. -// -// This header file contains the macro definitions for thread safety -// annotations that allow the developers to document the locking policies -// of their multi-threaded code. The annotations can also help program -// analysis tools to identify potential thread safety issues. - -#ifndef WEBRTC_BASE_THREAD_ANNOTATIONS_H_ -#define WEBRTC_BASE_THREAD_ANNOTATIONS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/thread_annotations.h" - -#endif // WEBRTC_BASE_THREAD_ANNOTATIONS_H_ diff --git a/webrtc/base/thread_checker.h b/webrtc/base/thread_checker.h deleted file mode 100644 index ade52564ec..0000000000 --- a/webrtc/base/thread_checker.h +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Borrowed from Chromium's src/base/threading/thread_checker.h. - -#ifndef WEBRTC_BASE_THREAD_CHECKER_H_ -#define WEBRTC_BASE_THREAD_CHECKER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/thread_checker.h" - -#endif // WEBRTC_BASE_THREAD_CHECKER_H_ diff --git a/webrtc/base/thread_checker_impl.h b/webrtc/base/thread_checker_impl.h deleted file mode 100644 index 3a0a6c7315..0000000000 --- a/webrtc/base/thread_checker_impl.h +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// Borrowed from Chromium's src/base/threading/thread_checker_impl.h. - -#ifndef WEBRTC_BASE_THREAD_CHECKER_IMPL_H_ -#define WEBRTC_BASE_THREAD_CHECKER_IMPL_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/thread_checker_impl.h" - -#endif // WEBRTC_BASE_THREAD_CHECKER_IMPL_H_ diff --git a/webrtc/base/timedelta.h b/webrtc/base/timedelta.h deleted file mode 100644 index f2e98a8cc2..0000000000 --- a/webrtc/base/timedelta.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TIMEDELTA_H_ -#define WEBRTC_BASE_TIMEDELTA_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/timedelta.h" - -#endif // WEBRTC_BASE_TIMEDELTA_H_ diff --git a/webrtc/base/timestampaligner.h b/webrtc/base/timestampaligner.h deleted file mode 100644 index 60c36311df..0000000000 --- a/webrtc/base/timestampaligner.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TIMESTAMPALIGNER_H_ -#define WEBRTC_BASE_TIMESTAMPALIGNER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/timestampaligner.h" - -#endif // WEBRTC_BASE_TIMESTAMPALIGNER_H_ diff --git a/webrtc/base/timeutils.h b/webrtc/base/timeutils.h deleted file mode 100644 index 1569b58f48..0000000000 --- a/webrtc/base/timeutils.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2005 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TIMEUTILS_H_ -#define WEBRTC_BASE_TIMEUTILS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/timeutils.h" - -#endif // WEBRTC_BASE_TIMEUTILS_H_ diff --git a/webrtc/base/trace_event.h b/webrtc/base/trace_event.h deleted file mode 100644 index 1bea5f4db8..0000000000 --- a/webrtc/base/trace_event.h +++ /dev/null @@ -1,14 +0,0 @@ -// Copyright (c) 2012 The Chromium Authors. All rights reserved. -// Use of this source code is governed by a BSD-style license that can be -// found in the LICENSE file under third_party_mods/chromium or at: -// http://src.chromium.org/svn/trunk/src/LICENSE - -#ifndef WEBRTC_BASE_TRACE_EVENT_H_ -#define WEBRTC_BASE_TRACE_EVENT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/trace_event.h" - -#endif // WEBRTC_BASE_TRACE_EVENT_H_ diff --git a/webrtc/base/transformadapter.h b/webrtc/base/transformadapter.h deleted file mode 100644 index 3d9c86bb26..0000000000 --- a/webrtc/base/transformadapter.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TRANSFORMADAPTER_H_ -#define WEBRTC_BASE_TRANSFORMADAPTER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/transformadapter.h" - -#endif // WEBRTC_BASE_TRANSFORMADAPTER_H_ diff --git a/webrtc/base/type_traits.h b/webrtc/base/type_traits.h deleted file mode 100644 index 6a4ac8d24e..0000000000 --- a/webrtc/base/type_traits.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_TYPE_TRAITS_H_ -#define WEBRTC_BASE_TYPE_TRAITS_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/type_traits.h" - -#endif // WEBRTC_BASE_TYPE_TRAITS_H_ diff --git a/webrtc/base/unixfilesystem.h b/webrtc/base/unixfilesystem.h deleted file mode 100644 index 7a182055af..0000000000 --- a/webrtc/base/unixfilesystem.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_UNIXFILESYSTEM_H_ -#define WEBRTC_BASE_UNIXFILESYSTEM_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/unixfilesystem.h" - -#endif // WEBRTC_BASE_UNIXFILESYSTEM_H_ diff --git a/webrtc/base/virtualsocketserver.h b/webrtc/base/virtualsocketserver.h deleted file mode 100644 index 31ce96d2e0..0000000000 --- a/webrtc/base/virtualsocketserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_VIRTUALSOCKETSERVER_H_ -#define WEBRTC_BASE_VIRTUALSOCKETSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/virtualsocketserver.h" - -#endif // WEBRTC_BASE_VIRTUALSOCKETSERVER_H_ diff --git a/webrtc/base/weak_ptr.h b/webrtc/base/weak_ptr.h deleted file mode 100644 index 282a551628..0000000000 --- a/webrtc/base/weak_ptr.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2016 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_WEAK_PTR_H_ -#define WEBRTC_BASE_WEAK_PTR_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/weak_ptr.h" - -#endif // WEBRTC_BASE_WEAK_PTR_H_ diff --git a/webrtc/base/win32.h b/webrtc/base/win32.h deleted file mode 100644 index 413bd11cab..0000000000 --- a/webrtc/base/win32.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_WIN32_H_ -#define WEBRTC_BASE_WIN32_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/win32.h" - -#endif // WEBRTC_BASE_WIN32_H_ diff --git a/webrtc/base/win32filesystem.h b/webrtc/base/win32filesystem.h deleted file mode 100644 index d647c440f0..0000000000 --- a/webrtc/base/win32filesystem.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_WIN32FILESYSTEM_H_ -#define WEBRTC_BASE_WIN32FILESYSTEM_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/win32filesystem.h" - -#endif // WEBRTC_BASE_WIN32FILESYSTEM_H_ diff --git a/webrtc/base/win32socketinit.h b/webrtc/base/win32socketinit.h deleted file mode 100644 index d7017e1387..0000000000 --- a/webrtc/base/win32socketinit.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2009 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_WIN32SOCKETINIT_H_ -#define WEBRTC_BASE_WIN32SOCKETINIT_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/win32socketinit.h" - -#endif // WEBRTC_BASE_WIN32SOCKETINIT_H_ diff --git a/webrtc/base/win32socketserver.h b/webrtc/base/win32socketserver.h deleted file mode 100644 index c14369295b..0000000000 --- a/webrtc/base/win32socketserver.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_WIN32SOCKETSERVER_H_ -#define WEBRTC_BASE_WIN32SOCKETSERVER_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/win32socketserver.h" - -#endif // WEBRTC_BASE_WIN32SOCKETSERVER_H_ diff --git a/webrtc/base/win32window.h b/webrtc/base/win32window.h deleted file mode 100644 index ffffdf9aa7..0000000000 --- a/webrtc/base/win32window.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_WIN32WINDOW_H_ -#define WEBRTC_BASE_WIN32WINDOW_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/win32window.h" - -#endif // WEBRTC_BASE_WIN32WINDOW_H_ diff --git a/webrtc/base/window.h b/webrtc/base/window.h deleted file mode 100644 index d515f7c829..0000000000 --- a/webrtc/base/window.h +++ /dev/null @@ -1,19 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_BASE_WINDOW_H_ -#define WEBRTC_BASE_WINDOW_H_ - - -// This header is deprecated and is just left here temporarily during -// refactoring. See https://bugs.webrtc.org/7634 for more details. -#include "webrtc/rtc_base/window.h" - -#endif // WEBRTC_BASE_WINDOW_H_ diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn index 6a50ea09a2..2d1853063a 100644 --- a/webrtc/call/BUILD.gn +++ b/webrtc/call/BUILD.gn @@ -28,8 +28,8 @@ rtc_source_set("call_interfaces") { "../api:libjingle_peerconnection_api", "../api:transport_api", "../api/audio_codecs:audio_codecs_api", - "../base:rtc_base", - "../base:rtc_base_approved", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", ] } @@ -43,7 +43,7 @@ rtc_source_set("rtp_interfaces") { "rtp_transport_controller_send_interface.h", ] deps = [ - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } @@ -64,8 +64,8 @@ rtc_source_set("rtp_receiver") { deps = [ ":rtp_interfaces", "..:webrtc_common", - "../base:rtc_base_approved", "../modules/rtp_rtcp", + "../rtc_base:rtc_base_approved", ] } @@ -76,8 +76,8 @@ rtc_source_set("rtp_sender") { ] deps = [ ":rtp_interfaces", - "../base:rtc_base_approved", "../modules/congestion_controller", + "../rtc_base:rtc_base_approved", ] } @@ -109,7 +109,6 @@ rtc_static_library("call") { "..:webrtc_common", "../api:transport_api", "../audio", - "../base:rtc_task_queue", "../logging:rtc_event_log_api", "../logging:rtc_event_log_impl", "../modules/bitrate_controller", @@ -117,6 +116,7 @@ rtc_static_library("call") { "../modules/pacing", "../modules/rtp_rtcp", "../modules/utility", + "../rtc_base:rtc_task_queue", "../system_wrappers", "../video", ] @@ -149,7 +149,6 @@ if (rtc_include_tests) { ":rtp_sender", "..:webrtc_common", "../api:mock_audio_mixer", - "../base:rtc_base_approved", "../logging:rtc_event_log_api", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer", @@ -158,6 +157,7 @@ if (rtc_include_tests) { "../modules/pacing", "../modules/rtp_rtcp", "../modules/rtp_rtcp:mock_rtp_rtcp", + "../rtc_base:rtc_base_approved", "../system_wrappers", "../test:audio_codec_mocks", "../test:direct_transport", @@ -191,11 +191,11 @@ if (rtc_include_tests) { ":call_interfaces", "..:webrtc_common", "../api/audio_codecs:builtin_audio_encoder_factory", - "../base:rtc_base_approved", "../logging:rtc_event_log_api", "../modules/audio_coding", "../modules/audio_mixer:audio_mixer_impl", "../modules/rtp_rtcp", + "../rtc_base:rtc_base_approved", "../system_wrappers", "../system_wrappers:metrics_default", "../test:direct_transport", diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn index ff0aa267fd..f7f3efbf36 100644 --- a/webrtc/common_audio/BUILD.gn +++ b/webrtc/common_audio/BUILD.gn @@ -63,8 +63,8 @@ rtc_static_library("common_audio") { deps = [ ":sinc_resampler", "..:webrtc_common", - "../base:gtest_prod", - "../base:rtc_base_approved", + "../rtc_base:gtest_prod", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] public_deps = [ @@ -209,8 +209,8 @@ rtc_source_set("common_audio_c") { ":common_audio_c_arm_asm", ":common_audio_cc", "..:webrtc_common", - "../base:compile_assert_c", - "../base:rtc_base_approved", + "../rtc_base:compile_assert_c", + "../rtc_base:rtc_base_approved", "../system_wrappers:system_wrappers", ] } @@ -225,7 +225,7 @@ rtc_source_set("common_audio_cc") { public_configs = [ ":common_audio_config" ] deps = [ "..:webrtc_common", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "../system_wrappers:system_wrappers", ] } @@ -236,8 +236,8 @@ rtc_source_set("sinc_resampler") { ] deps = [ "..:webrtc_common", - "../base:gtest_prod", - "../base:rtc_base_approved", + "../rtc_base:gtest_prod", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] } @@ -344,7 +344,7 @@ if (rtc_build_with_neon) { } deps = [ ":common_audio_c", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } } @@ -401,8 +401,8 @@ if (rtc_include_tests) { ":common_audio", ":sinc_resampler", "..:webrtc_common", - "../base:rtc_base_approved", - "../base:rtc_base_tests_utils", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers", "../test:test_main", "//testing/gmock", diff --git a/webrtc/common_video/BUILD.gn b/webrtc/common_video/BUILD.gn index 68b4934317..20953c802c 100644 --- a/webrtc/common_video/BUILD.gn +++ b/webrtc/common_video/BUILD.gn @@ -57,10 +57,10 @@ rtc_static_library("common_video") { deps = [ "..:webrtc_common", - "../base:rtc_base", - "../base:rtc_task_queue", "../media:rtc_h264_profile_id", "../modules:module_api", + "../rtc_base:rtc_base", + "../rtc_base:rtc_task_queue", "../system_wrappers", ] public_deps = [ @@ -114,9 +114,9 @@ if (rtc_include_tests) { deps = [ ":common_video", - "../base:rtc_base", - "../base:rtc_base_approved", "../modules/video_capture:video_capture", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", "../system_wrappers:system_wrappers", "../test:test_main", "../test:video_test_common", diff --git a/webrtc/examples/BUILD.gn b/webrtc/examples/BUILD.gn index 85813fa222..55b72ddf16 100644 --- a/webrtc/examples/BUILD.gn +++ b/webrtc/examples/BUILD.gn @@ -422,7 +422,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) { "objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm", ] deps = [ - "//webrtc/base:rtc_base", + "//webrtc/rtc_base:rtc_base", ] public_deps = [ ":AppRTCMobile_ios_frameworks", @@ -524,12 +524,12 @@ if (is_linux || is_win) { "//third_party/libyuv", "//webrtc/api:libjingle_peerconnection_test_api", "//webrtc/api:video_frame_api", - "//webrtc/base:rtc_base", - "//webrtc/base:rtc_base_approved", - "//webrtc/base:rtc_json", "//webrtc/media:rtc_media", "//webrtc/modules/video_capture:video_capture_module", "//webrtc/pc:libjingle_peerconnection", + "//webrtc/rtc_base:rtc_base", + "//webrtc/rtc_base:rtc_base_approved", + "//webrtc/rtc_base:rtc_json", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -548,7 +548,7 @@ if (is_linux || is_win) { ] deps = [ "//webrtc:webrtc_common", - "//webrtc/base:rtc_base_approved", + "//webrtc/rtc_base:rtc_base_approved", "//webrtc/rtc_tools:command_line_parser", ] if (!build_with_chromium && is_clang) { @@ -562,10 +562,10 @@ if (is_linux || is_win) { "relayserver/relayserver_main.cc", ] deps = [ - "../base:rtc_base", - "//webrtc/base:rtc_base_approved", + "../rtc_base:rtc_base", "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", + "//webrtc/rtc_base:rtc_base_approved", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -580,10 +580,10 @@ if (is_linux || is_win) { "turnserver/turnserver_main.cc", ] deps = [ - "../base:rtc_base", - "//webrtc/base:rtc_base_approved", + "../rtc_base:rtc_base", "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", + "//webrtc/rtc_base:rtc_base_approved", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -598,10 +598,10 @@ if (is_linux || is_win) { "stunserver/stunserver_main.cc", ] deps = [ - "../base:rtc_base", - "//webrtc/base:rtc_base_approved", + "../rtc_base:rtc_base", "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", + "//webrtc/rtc_base:rtc_base_approved", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -633,13 +633,13 @@ if (is_win) { deps = [ "//webrtc/api:libjingle_peerconnection_test_api", "//webrtc/api:video_frame_api", - "//webrtc/base:rtc_base", - "//webrtc/base:rtc_base_approved", - "//webrtc/base:rtc_json", "//webrtc/media:rtc_media", "//webrtc/media:rtc_media_base", "//webrtc/modules/video_capture:video_capture_module", "//webrtc/pc:libjingle_peerconnection", + "//webrtc/rtc_base:rtc_base", + "//webrtc/rtc_base:rtc_base_approved", + "//webrtc/rtc_base:rtc_json", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", ] @@ -661,10 +661,10 @@ if (!build_with_chromium) { } deps = [ - "../base:rtc_base", - "../base:rtc_base_approved", "../p2p:libstunprober", "../p2p:rtc_p2p", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", "../system_wrappers:field_trial_default", ] } diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn index f3c3469807..6a70324e5b 100644 --- a/webrtc/logging/BUILD.gn +++ b/webrtc/logging/BUILD.gn @@ -30,7 +30,7 @@ rtc_source_set("rtc_event_log_api") { deps = [ "..:video_stream_api", "..:webrtc_common", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } @@ -48,11 +48,11 @@ rtc_static_library("rtc_event_log_impl") { deps = [ ":rtc_event_log_api", "..:webrtc_common", - "../base:protobuf_utils", - "../base:rtc_base_approved", "../modules/audio_coding:audio_network_adaptor", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp", + "../rtc_base:protobuf_utils", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] @@ -96,8 +96,8 @@ if (rtc_enable_protobuf) { } deps = [ "..:video_stream_api", - "../base:protobuf_utils", - "../base:rtc_base_approved", + "../rtc_base:protobuf_utils", + "../rtc_base:rtc_base_approved", ] } @@ -111,12 +111,12 @@ if (rtc_enable_protobuf) { deps = [ ":rtc_event_log_impl", ":rtc_event_log_parser", - "../base:rtc_base_approved", - "../base:rtc_base_tests_utils", "../call", "../modules/audio_coding:audio_network_adaptor", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", "../test:test_support", "//testing/gmock", @@ -136,8 +136,8 @@ if (rtc_enable_protobuf) { ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_parser", - "../base:rtc_base_approved", "../modules/rtp_rtcp:rtp_rtcp", + "../rtc_base:rtc_base_approved", "../system_wrappers:field_trial_default", "../system_wrappers:metrics_default", "../test:rtp_test_utils", @@ -159,7 +159,7 @@ if (rtc_enable_protobuf) { ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_parser", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", # TODO(kwiberg): Remove this dependency. "../api/audio_codecs:audio_codecs_api", @@ -182,7 +182,7 @@ if (rtc_enable_protobuf) { ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_proto", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "//third_party/gflags", ] if (!build_with_chromium && is_clang) { diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn index fad410b2dd..ef9b79c7e2 100644 --- a/webrtc/media/BUILD.gn +++ b/webrtc/media/BUILD.gn @@ -45,8 +45,8 @@ rtc_source_set("rtc_h264_profile_id") { deps = [ "..:webrtc_common", - "../base:rtc_base", - "../base:rtc_base_approved", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", ] } @@ -115,9 +115,9 @@ rtc_static_library("rtc_media_base") { ":rtc_h264_profile_id", "..:webrtc_common", "../api:libjingle_peerconnection_api", - "../base:rtc_base", - "../base:rtc_base_approved", "../p2p", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", ] if (is_nacl) { @@ -227,10 +227,6 @@ rtc_static_library("rtc_audio_video") { "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", "../api/video_codecs:video_codecs_api", - "../base:rtc_base", - "../base:rtc_base_approved", - "../base:rtc_task_queue", - "../base:sequenced_task_checker", "../call", "../common_video:common_video", "../modules/audio_coding:rent_a_codec", @@ -245,6 +241,10 @@ rtc_static_library("rtc_audio_video") { "../modules/video_coding:webrtc_vp9", "../p2p:rtc_p2p", "../pc:rtc_pc_base", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue", + "../rtc_base:sequenced_task_checker", "../system_wrappers", "../video", "../voice_engine", @@ -292,9 +292,9 @@ rtc_static_library("rtc_data") { "..:webrtc_common", "../api:call_api", "../api:transport_api", - "../base:rtc_base", - "../base:rtc_base_approved", "../p2p:rtc_p2p", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] } @@ -368,10 +368,10 @@ if (rtc_include_tests) { "../api:call_api", "../api:video_frame_api", "../api/video_codecs:video_codecs_api", - "../base:rtc_base", - "../base:rtc_base_approved", - "../base:rtc_base_tests_utils", "../call:call_interfaces", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", "../test:test_support", "//testing/gtest", ] @@ -508,10 +508,6 @@ if (rtc_include_tests) { "../api/audio_codecs:builtin_audio_encoder_factory", "../api/video_codecs:video_codecs_api", "../audio", - "../base:rtc_base", - "../base:rtc_base_approved", - "../base:rtc_base_tests_main", - "../base:rtc_base_tests_utils", "../call:call_interfaces", "../common_video:common_video", "../logging:rtc_event_log_api", @@ -521,6 +517,10 @@ if (rtc_include_tests) { "../modules/video_coding:video_coding_utility", "../modules/video_coding:webrtc_vp8", "../p2p:p2p_test_utils", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_main", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", "../test:audio_codec_mocks", "../test:test_support", diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn index 258683165d..0ae50414fc 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn @@ -37,7 +37,7 @@ rtc_source_set("module_api") { deps = [ "..:webrtc_common", "../api:video_frame_api", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index d5b669c0eb..f751963ca4 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -47,7 +47,7 @@ rtc_static_library("audio_format_conversion") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } @@ -58,8 +58,8 @@ rtc_static_library("builtin_audio_decoder_factory_internal") { ] deps = [ "../..:webrtc_common", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", "../../api/audio_codecs:audio_codecs_api", ] + audio_codec_deps defines = audio_codec_defines @@ -72,8 +72,8 @@ rtc_static_library("builtin_audio_encoder_factory_internal") { ] deps = [ "../..:webrtc_common", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", "../../api/audio_codecs:audio_codecs_api", ] + audio_codec_deps defines = audio_codec_defines @@ -89,8 +89,8 @@ rtc_static_library("rent_a_codec") { deps = [ "../../api/audio_codecs:audio_codecs_api", "../..:webrtc_common", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ":audio_coding_module_typedefs", ":isac_common", @@ -156,7 +156,7 @@ rtc_static_library("audio_coding") { ":audio_coding_module_typedefs", ":neteq", ":rent_a_codec", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../logging:rtc_event_log_api", ] defines = audio_coding_defines @@ -169,7 +169,7 @@ rtc_static_library("legacy_encoded_audio_frame") { ] deps = [ "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } @@ -193,8 +193,8 @@ rtc_static_library("cng") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", ] } @@ -212,8 +212,8 @@ rtc_static_library("red") { deps = [ "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", ] } @@ -238,7 +238,7 @@ rtc_static_library("g711") { ":legacy_encoded_audio_frame", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] public_deps = [ ":g711_c", @@ -280,7 +280,7 @@ rtc_static_library("g722") { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/g722:audio_encoder_g722_config", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] public_deps = [ ":g722_c", @@ -323,8 +323,8 @@ rtc_static_library("ilbc") { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/ilbc:audio_encoder_ilbc_config", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", ] public_deps = [ ":ilbc_c", @@ -480,8 +480,8 @@ rtc_source_set("ilbc_c") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", ] } @@ -495,7 +495,7 @@ rtc_static_library("isac_common") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } @@ -587,9 +587,9 @@ rtc_static_library("isac_c") { deps = [ ":isac_common", "../..:webrtc_common", - "../../base:compile_assert_c", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:compile_assert_c", + "../../rtc_base:rtc_base_approved", ] } @@ -697,9 +697,9 @@ rtc_source_set("isac_fix_c") { ":isac_common", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:compile_assert_c", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:compile_assert_c", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -773,8 +773,8 @@ if (rtc_build_with_neon) { deps = [ ":isac_fix_common", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", ] } } @@ -799,7 +799,7 @@ rtc_static_library("pcm16b") { ":legacy_encoded_audio_frame", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] public_deps = [ ":pcm16b_c", @@ -837,10 +837,10 @@ rtc_static_library("webrtc_opus") { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/opus:audio_encoder_opus_config", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", - "../../base:rtc_numerics", "../../common_audio", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_numerics", "../../system_wrappers", ] public_deps = [ @@ -876,7 +876,7 @@ rtc_source_set("webrtc_opus_c") { deps = [ "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } @@ -926,10 +926,10 @@ rtc_static_library("audio_network_adaptor") { deps = [ "../..:webrtc_common", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", "../../common_audio", "../../logging:rtc_event_log_api", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -953,7 +953,7 @@ rtc_source_set("neteq_decoder_enum") { ] deps = [ "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } @@ -1042,9 +1042,9 @@ rtc_static_library("neteq") { "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:gtest_prod", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:gtest_prod", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -1102,7 +1102,7 @@ rtc_source_set("neteq_tools_minimal") { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../rtp_rtcp", ] } @@ -1134,9 +1134,9 @@ rtc_source_set("neteq_test_tools") { ":pcm16b", "..:module_api", "../..:webrtc_common", - "../../base:rtc_base_approved", - "../../base:rtc_base_tests_utils", "../../common_audio", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_base_tests_utils", "../../test:rtp_test_utils", "../rtp_rtcp", ] @@ -1183,8 +1183,8 @@ rtc_source_set("neteq_tools") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", "../rtp_rtcp", ] @@ -1212,8 +1212,8 @@ if (rtc_enable_protobuf) { } deps = [ - "../../base:rtc_base_approved", "../../logging:rtc_event_log_parser", + "../../rtc_base:rtc_base_approved", ] public_deps = [ "../../logging:rtc_event_log_proto", @@ -1307,7 +1307,7 @@ if (rtc_include_tests) { "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", ] @@ -1342,8 +1342,8 @@ if (rtc_include_tests) { ":neteq_test_tools", ":webrtc_opus", "../..:webrtc_common", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", ] @@ -1369,7 +1369,7 @@ if (rtc_include_tests) { "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", ":neteq_tools", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_support", "//testing/gtest", ] @@ -1388,7 +1388,7 @@ if (rtc_include_tests) { ":audio_coding", ":neteq_tools", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_support", "//testing/gtest", ] @@ -1412,7 +1412,7 @@ if (rtc_include_tests) { ":audio_format_conversion", "..:module_api", "../../:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../../system_wrappers:system_wrappers_default", "../../test:test_support", @@ -1442,7 +1442,7 @@ if (rtc_include_tests) { ":audio_format_conversion", "..:module_api", "../../:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../../system_wrappers:system_wrappers_default", "../../test:test_support", @@ -1489,8 +1489,8 @@ if (rtc_include_tests) { ":neteq_tools", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/opus:audio_encoder_opus", - "../../base:protobuf_utils", "../../common_audio", + "../../rtc_base:protobuf_utils", "../../test:test_main", "//testing/gtest", ] @@ -1540,7 +1540,7 @@ if (rtc_include_tests) { ":neteq", ":neteq_test_tools", "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "../../test:test_support", "//third_party/gflags", @@ -1573,7 +1573,7 @@ if (rtc_include_tests) { ":isac_fix", ":webrtc_opus", "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "../../test:test_main", "../audio_processing", @@ -1603,7 +1603,7 @@ if (rtc_include_tests) { "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../../test:test_support", "//testing/gtest", @@ -1628,7 +1628,7 @@ if (rtc_include_tests) { "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:builtin_audio_decoder_factory", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_support", "//testing/gtest", "//third_party/gflags", @@ -1705,8 +1705,8 @@ if (rtc_include_tests) { ":pcm16b", ":webrtc_opus", "../..:webrtc_common", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", ] configs += [ ":RTPencode_config" ] @@ -1749,7 +1749,7 @@ if (rtc_include_tests) { ] deps = [ - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "../../test:rtp_test_utils", "//testing/gtest", @@ -1774,7 +1774,7 @@ if (rtc_include_tests) { testonly = true deps = [ "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_support", "//testing/gtest", ] @@ -1853,7 +1853,7 @@ if (rtc_include_tests) { ":neteq_quality_test_support", ":neteq_tools", "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "../../test:test_main", "//testing/gtest", @@ -1872,7 +1872,7 @@ if (rtc_include_tests) { ":isac_fix", ":neteq", ":neteq_quality_test_support", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_main", "//testing/gtest", "//third_party/gflags", @@ -1890,7 +1890,7 @@ if (rtc_include_tests) { ":g711", ":neteq", ":neteq_quality_test_support", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_main", "//testing/gtest", "//third_party/gflags", @@ -1950,7 +1950,7 @@ if (rtc_include_tests) { deps = [ ":isac", ":isac_test_util", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] configs += [ ":isac_test_warnings_config" ] @@ -1991,7 +1991,7 @@ if (rtc_include_tests) { deps = [ ":isac", ":isac_test_util", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] include_dirs = [ @@ -2042,8 +2042,8 @@ if (rtc_include_tests) { deps = [ ":webrtc_opus", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", "../../test:test_main", "//testing/gtest", ] @@ -2167,11 +2167,11 @@ if (rtc_include_tests) { "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", "../../api/audio_codecs:builtin_audio_encoder_factory", - "../../base:protobuf_utils", - "../../base:rtc_base", - "../../base:rtc_base_approved", - "../../base:rtc_base_tests_utils", "../../common_audio", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_base_tests_utils", "../../system_wrappers:system_wrappers", "../../test:audio_codec_mocks", "../../test:field_trial", diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn index 8939da222e..56b20193d0 100644 --- a/webrtc/modules/audio_conference_mixer/BUILD.gn +++ b/webrtc/modules/audio_conference_mixer/BUILD.gn @@ -42,7 +42,7 @@ rtc_static_library("audio_conference_mixer") { "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../audio_processing", ] diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn index c565165192..a60087efd6 100644 --- a/webrtc/modules/audio_device/BUILD.gn +++ b/webrtc/modules/audio_device/BUILD.gn @@ -51,9 +51,9 @@ rtc_static_library("audio_device") { deps = [ "..:module_api", "../..:webrtc_common", - "../../base:rtc_base_approved", - "../../base:rtc_task_queue", "../../common_audio", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_task_queue", "../../system_wrappers", "../utility", ] @@ -177,8 +177,8 @@ rtc_static_library("audio_device") { } if (is_ios) { public_deps = [ - "../../base:gtest_prod", - "../../base:rtc_base", + "../../rtc_base:gtest_prod", + "../../rtc_base:rtc_base", "../../sdk:objc_audio", "../../sdk:objc_common", ] @@ -281,7 +281,7 @@ if (rtc_include_tests) { deps = [ ":audio_device", ":mock_audio_device", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", "../utility:utility", @@ -331,7 +331,7 @@ if (rtc_include_tests) { deps = [ ":audio_device", "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../../test:test_main", "../../test:test_support", diff --git a/webrtc/modules/audio_mixer/BUILD.gn b/webrtc/modules/audio_mixer/BUILD.gn index cd3b768ee1..0410e1f0e9 100644 --- a/webrtc/modules/audio_mixer/BUILD.gn +++ b/webrtc/modules/audio_mixer/BUILD.gn @@ -41,7 +41,7 @@ rtc_static_library("audio_mixer_impl") { "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../audio_processing", ] @@ -61,7 +61,7 @@ rtc_static_library("audio_frame_manipulator") { deps = [ "..:module_api", "../../audio/utility", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } @@ -90,8 +90,8 @@ if (rtc_include_tests) { "..:module_api", "../../api:audio_mixer_api", "../../audio/utility:audio_frame_operations", - "../../base:rtc_base", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", "../../test:test_support", "//testing/gmock", ] diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn index 3af019e49e..fe83596277 100644 --- a/webrtc/modules/audio_processing/BUILD.gn +++ b/webrtc/modules/audio_processing/BUILD.gn @@ -238,8 +238,8 @@ rtc_static_library("audio_processing") { "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", - "../../base:gtest_prod", - "../../base:protobuf_utils", + "../../rtc_base:gtest_prod", + "../../rtc_base:protobuf_utils", "../audio_coding:isac", ] public_deps = [ @@ -303,8 +303,8 @@ rtc_static_library("audio_processing") { configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] deps += [ - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] } @@ -316,7 +316,7 @@ rtc_source_set("aec_dump_interface") { ] deps = [ - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } @@ -356,8 +356,8 @@ rtc_source_set("audio_processing_c") { deps = [ "../..:webrtc_common", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -470,7 +470,7 @@ if (rtc_build_with_neon) { ] } deps = [ - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] } } @@ -550,11 +550,11 @@ if (rtc_include_tests) { ":audioproc_test_utils", "..:module_api", "../..:webrtc_common", - "../../base:gtest_prod", - "../../base:protobuf_utils", - "../../base:rtc_base", - "../../base:rtc_base_approved", "../../common_audio:common_audio", + "../../rtc_base:gtest_prod", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", "../audio_coding:neteq_tools", @@ -594,7 +594,7 @@ if (rtc_include_tests) { ":audioproc_debug_proto", ":audioproc_protobuf_utils", ":audioproc_unittest_proto", - "../../base:rtc_task_queue", + "../../rtc_base:rtc_task_queue", "aec_dump", "aec_dump:aec_dump_unittests", ] @@ -696,7 +696,7 @@ if (rtc_include_tests) { deps = [ ":audio_processing", ":audioproc_test_utils", - "../../base:protobuf_utils", + "../../rtc_base:protobuf_utils", "//testing/gtest", ] @@ -720,9 +720,9 @@ if (rtc_include_tests) { ":audioproc_protobuf_utils", ":audioproc_test_utils", "../..:webrtc_common", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers_default", "//third_party/gflags:gflags", ] @@ -745,10 +745,10 @@ if (rtc_include_tests) { ":audioproc_debug_proto", ":audioproc_protobuf_utils", ":audioproc_test_utils", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", - "../../base:rtc_task_queue", "../../common_audio:common_audio", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_task_queue", "../../system_wrappers", "../../system_wrappers:system_wrappers_default", "../../test:test_support", @@ -776,8 +776,8 @@ if (rtc_include_tests) { deps = [ ":audio_processing", "..:module_api", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", ] } @@ -825,8 +825,8 @@ if (rtc_include_tests) { deps = [ ":audio_processing", ":audioproc_test_utils", - "../../base:rtc_base_approved", "../../common_audio:common_audio", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:metrics_default", "//third_party/gflags", ] @@ -866,8 +866,8 @@ if (rtc_include_tests) { deps = [ ":audioproc_debug_proto", "../..:webrtc_common", - "../../base:protobuf_utils", - "../../base:rtc_base_approved", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_base_approved", ] } } diff --git a/webrtc/modules/audio_processing/aec_dump/BUILD.gn b/webrtc/modules/audio_processing/aec_dump/BUILD.gn index 950dd68b17..818a9bf27d 100644 --- a/webrtc/modules/audio_processing/aec_dump/BUILD.gn +++ b/webrtc/modules/audio_processing/aec_dump/BUILD.gn @@ -18,7 +18,7 @@ rtc_source_set("aec_dump") { ] deps = [ - "../../../base:rtc_base_approved", + "../../../rtc_base:rtc_base_approved", ] } @@ -49,7 +49,7 @@ rtc_source_set("mock_aec_dump_unittests") { deps = [ ":mock_aec_dump", "..:audio_processing", - "../../../base:rtc_base_approved", + "../../../rtc_base:rtc_base_approved", "//testing/gtest", ] } @@ -73,10 +73,10 @@ if (rtc_enable_protobuf) { ] deps = [ - "../../../base:protobuf_utils", - "../../../base:rtc_base_approved", - "../../../base:rtc_task_queue", "../../../modules:module_api", + "../../../rtc_base:protobuf_utils", + "../../../rtc_base:rtc_base_approved", + "../../../rtc_base:rtc_task_queue", "../../../system_wrappers", ] @@ -90,8 +90,8 @@ if (rtc_enable_protobuf) { ":aec_dump_impl", "..:aec_dump_interface", "..:audioproc_debug_proto", - "../../../base:rtc_task_queue", "../../../modules:module_api", + "../../../rtc_base:rtc_task_queue", "../../../test:test_support", "//testing/gtest", ] diff --git a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn index af24f8ab5e..587663b7fc 100644 --- a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn +++ b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn @@ -22,8 +22,8 @@ rtc_executable("conversational_speech_generator") { ] deps = [ ":lib", - "../../../../../webrtc/base:rtc_base_approved", - "../../../../../webrtc/test:test_support", + "../../../../rtc_base:rtc_base_approved", + "../../../../test:test_support", "//third_party/gflags", ] } @@ -45,9 +45,9 @@ rtc_static_library("lib") { "wavreader_interface.h", ] deps = [ - "../../../../../webrtc:webrtc_common", - "../../../../../webrtc/base:rtc_base_approved", - "../../../../../webrtc/common_audio", + "../../../..:webrtc_common", + "../../../../common_audio", + "../../../../rtc_base:rtc_base_approved", ] visibility = [ ":*" ] # Only targets in this file can depend on this. } @@ -63,14 +63,11 @@ rtc_source_set("unittest") { ] deps = [ ":lib", - "../../../../../webrtc:webrtc_common", - "../../../../../webrtc/base:rtc_base_approved", - "../../../../../webrtc/common_audio", - "../../../../../webrtc/test:test_support", + "../../../..:webrtc_common", + "../../../../common_audio", + "../../../../rtc_base:rtc_base_approved", + "../../../../test:test_support", "//testing/gmock", "//testing/gtest", - "//webrtc:webrtc_common", - "//webrtc/base:rtc_base_approved", - "//webrtc/test:test_support", ] } diff --git a/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn b/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn index 154219bf53..74d5eee877 100644 --- a/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn +++ b/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn @@ -105,7 +105,7 @@ rtc_executable("fake_polqa") { output_name = "py_quality_assessment/quality_assessment/fake_polqa" deps = [ "//webrtc:webrtc_common", - "//webrtc/base:rtc_base_approved", + "//webrtc/rtc_base:rtc_base_approved", ] } diff --git a/webrtc/modules/bitrate_controller/BUILD.gn b/webrtc/modules/bitrate_controller/BUILD.gn index 33a28869ba..f31025bd05 100644 --- a/webrtc/modules/bitrate_controller/BUILD.gn +++ b/webrtc/modules/bitrate_controller/BUILD.gn @@ -37,7 +37,7 @@ rtc_static_library("bitrate_controller") { } deps = [ - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../rtp_rtcp", ] diff --git a/webrtc/modules/congestion_controller/BUILD.gn b/webrtc/modules/congestion_controller/BUILD.gn index ef45297031..02123724f9 100644 --- a/webrtc/modules/congestion_controller/BUILD.gn +++ b/webrtc/modules/congestion_controller/BUILD.gn @@ -49,10 +49,10 @@ rtc_static_library("congestion_controller") { deps = [ "..:module_api", "../..:webrtc_common", - "../../base:rtc_base", - "../../base:rtc_base_approved", - "../../base:rtc_numerics", "../../logging:rtc_event_log_api", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_numerics", "../../system_wrappers", "../bitrate_controller", "../pacing", @@ -88,8 +88,8 @@ if (rtc_include_tests) { deps = [ ":congestion_controller", ":mock_congestion_controller", - "../../base:rtc_base", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:field_trial", "../../test:test_support", diff --git a/webrtc/modules/desktop_capture/BUILD.gn b/webrtc/modules/desktop_capture/BUILD.gn index 93ceb752c2..47b186d835 100644 --- a/webrtc/modules/desktop_capture/BUILD.gn +++ b/webrtc/modules/desktop_capture/BUILD.gn @@ -28,7 +28,7 @@ rtc_static_library("primitives") { deps = [ "../..:webrtc_common", - "../../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806. + "../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806. ] } @@ -49,8 +49,8 @@ if (rtc_include_tests) { ":desktop_capture_mock", ":primitives", ":screen_drawer", - "../../base:rtc_base", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../../test:test_support", "../../test:video_test_support", @@ -94,7 +94,7 @@ if (rtc_include_tests) { ":desktop_capture_mock", ":primitives", "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../../test:test_support", "//testing/gmock", @@ -131,7 +131,7 @@ if (rtc_include_tests) { deps = [ ":primitives", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] } @@ -155,7 +155,7 @@ if (rtc_include_tests) { deps = [ ":primitives", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_support", ] } @@ -290,7 +290,7 @@ rtc_static_library("desktop_capture") { deps = [ ":primitives", "../..:webrtc_common", - "../../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806. + "../../rtc_base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806. "../../system_wrappers", "//third_party/libyuv", ] diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn index 7ab897f28f..989305c704 100644 --- a/webrtc/modules/media_file/BUILD.gn +++ b/webrtc/modules/media_file/BUILD.gn @@ -35,8 +35,8 @@ rtc_static_library("media_file") { deps = [ "..:module_api", "../..:webrtc_common", - "../../base:rtc_base_approved", "../../common_audio", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] } diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn index 57126d7143..3d1d495fa2 100644 --- a/webrtc/modules/pacing/BUILD.gn +++ b/webrtc/modules/pacing/BUILD.gn @@ -28,8 +28,8 @@ rtc_static_library("pacing") { deps = [ "..:module_api", "../../:webrtc_common", - "../../base:rtc_base_approved", "../../logging:rtc_event_log_api", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../remote_bitrate_estimator", "../rtp_rtcp", @@ -55,8 +55,8 @@ if (rtc_include_tests) { ] deps = [ ":pacing", - "../../base:rtc_base_approved", - "../../base:rtc_base_tests_utils", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_base_tests_utils", "../../system_wrappers:system_wrappers", "../../test:test_support", "../rtp_rtcp", diff --git a/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/webrtc/modules/remote_bitrate_estimator/BUILD.gn index 5a3afc65fd..1ecf630b1d 100644 --- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn +++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn @@ -51,8 +51,8 @@ rtc_static_library("remote_bitrate_estimator") { deps = [ "../..:webrtc_common", - "../../base:rtc_base", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] } @@ -117,9 +117,9 @@ if (rtc_include_tests) { ":remote_bitrate_estimator", "..:module_api", "../..:webrtc_common", - "../../base:gtest_prod", - "../../base:rtc_base", - "../../base:rtc_base_approved", + "../../rtc_base:gtest_prod", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../../test:test_support", "../../voice_engine", @@ -147,7 +147,7 @@ if (rtc_include_tests) { deps = [ ":bwe_simulator_lib", ":remote_bitrate_estimator", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_support", ] if (!build_with_chromium && is_clang) { @@ -185,8 +185,8 @@ if (rtc_include_tests) { ":mock_remote_bitrate_observer", ":remote_bitrate_estimator", "../..:webrtc_common", - "../../base:rtc_base", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:field_trial", "../../test:test_support", @@ -227,7 +227,7 @@ if (rtc_include_tests) { ":bwe_simulator_lib", ":remote_bitrate_estimator", "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_main", "//testing/gmock", "//testing/gtest", diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index dc623cefb4..3b0bc35b2b 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -170,11 +170,11 @@ rtc_static_library("rtp_rtcp") { "../../api:libjingle_peerconnection_api", "../../api:transport_api", "../../api/audio_codecs:audio_codecs_api", - "../../base:gtest_prod", - "../../base:rtc_base_approved", - "../../base:sequenced_task_checker", "../../common_video", "../../logging:rtc_event_log_api", + "../../rtc_base:gtest_prod", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:sequenced_task_checker", "../../system_wrappers", "../audio_coding:audio_format_conversion", "../remote_bitrate_estimator", @@ -200,7 +200,7 @@ rtc_source_set("fec_test_helper") { deps = [ ":rtp_rtcp", "..:module_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", ] # TODO(jschuh): bugs.webrtc.org/1348: fix this warning. @@ -221,7 +221,7 @@ rtc_source_set("mock_rtp_rtcp") { deps = [ ":rtp_rtcp", "..:module_api", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_support", ] } @@ -256,7 +256,7 @@ if (rtc_include_tests) { ] deps = [ ":rtp_rtcp", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../test:test_support", ] if (!build_with_chromium && is_clang) { @@ -342,8 +342,8 @@ if (rtc_include_tests) { "..:module_api", "../..:webrtc_common", "../../api:transport_api", - "../../base:rtc_base_approved", "../../common_video:common_video", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:field_trial", "../../test:rtp_test_utils", diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn index 71238905dc..e98b30dc53 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn @@ -33,8 +33,8 @@ rtc_static_library("utility") { "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", - "../../base:rtc_task_queue", "../../common_audio", + "../../rtc_base:rtc_task_queue", "../../system_wrappers", "../media_file", ] @@ -56,7 +56,7 @@ if (rtc_include_tests) { deps = [ ":utility", "..:module_api", - "../../base:rtc_task_queue", + "../../rtc_base:rtc_task_queue", "../../test:test_support", "//testing/gmock", ] diff --git a/webrtc/modules/video_capture/BUILD.gn b/webrtc/modules/video_capture/BUILD.gn index 5865688e20..b150dffe0b 100644 --- a/webrtc/modules/video_capture/BUILD.gn +++ b/webrtc/modules/video_capture/BUILD.gn @@ -28,8 +28,8 @@ rtc_static_library("video_capture_module") { deps = [ "..:module_api", "../..:webrtc_common", - "../../base:rtc_base_approved", "../../common_video", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -47,7 +47,7 @@ rtc_static_library("video_capture") { deps = [ ":video_capture_module", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -91,7 +91,7 @@ if (!build_with_chromium) { deps = [ ":video_capture_module", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -175,8 +175,8 @@ if (!build_with_chromium) { deps = [ ":video_capture_internal_impl", ":video_capture_module", - "../../base:rtc_base_approved", "../../common_video:common_video", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../system_wrappers:system_wrappers_default", "../../test:video_test_common", diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn index 67c8ccd239..94cb1334a8 100644 --- a/webrtc/modules/video_coding/BUILD.gn +++ b/webrtc/modules/video_coding/BUILD.gn @@ -97,12 +97,12 @@ rtc_static_library("video_coding") { "..:module_api", "../..:video_stream_api", "../..:webrtc_common", - "../../base:rtc_base", - "../../base:rtc_base_approved", - "../../base:rtc_numerics", - "../../base:rtc_task_queue", - "../../base:sequenced_task_checker", "../../common_video", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_numerics", + "../../rtc_base:rtc_task_queue", + "../../rtc_base:sequenced_task_checker", "../../system_wrappers", "../rtp_rtcp:rtp_rtcp", "../utility:utility", @@ -136,12 +136,12 @@ rtc_static_library("video_coding_utility") { "..:module_api", "../..:webrtc_common", "../../api/video_codecs:video_codecs_api", - "../../base:rtc_base_approved", - "../../base:rtc_numerics", - "../../base:rtc_task_queue", - "../../base:sequenced_task_checker", "../../common_video", "../../modules/rtp_rtcp:rtp_rtcp", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_numerics", + "../../rtc_base:rtc_task_queue", + "../../rtc_base:sequenced_task_checker", "../../system_wrappers", ] } @@ -160,8 +160,8 @@ rtc_static_library("webrtc_h264") { defines = [] deps = [ ":video_coding_utility", - "../../base:rtc_base_approved", "../../media:rtc_media_base", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] @@ -198,8 +198,8 @@ rtc_static_library("webrtc_i420") { deps = [ "../..:webrtc_common", - "../../base:rtc_base_approved", "../../common_video:common_video", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] } @@ -232,9 +232,9 @@ rtc_static_library("webrtc_vp8") { "..:module_api", "../..:webrtc_common", "../../api/video_codecs:video_codecs_api", - "../../base:rtc_base_approved", - "../../base:sequenced_task_checker", "../../common_video", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:sequenced_task_checker", "../../system_wrappers", ] if (rtc_build_libvpx) { @@ -267,8 +267,8 @@ rtc_static_library("webrtc_vp9") { deps = [ ":video_coding_utility", "..:module_api", - "../../base:rtc_base_approved", "../../common_video", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] if (rtc_build_libvpx) { @@ -292,8 +292,8 @@ if (rtc_include_tests) { ":video_coding", ":webrtc_vp8", "../../api:video_frame_api", - "../../base:rtc_base_approved", "../../common_video:common_video", + "../../rtc_base:rtc_base_approved", "../../test:test_support", ] } @@ -315,7 +315,7 @@ if (rtc_include_tests) { ":video_coding", ":webrtc_vp8", "../..:webrtc_common", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:field_trial_default", "../../system_wrappers:metrics_default", "../../system_wrappers:system_wrappers", @@ -354,8 +354,8 @@ if (rtc_include_tests) { ":webrtc_vp8", "../..:webrtc_common", "../../api/video_codecs:video_codecs_api", - "../../base:rtc_base_approved", "../../common_video:common_video", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", "../../test:test_support", "../../test:video_test_common", @@ -378,8 +378,8 @@ if (rtc_include_tests) { ":webrtc_vp8", ":webrtc_vp9", "../..:webrtc_common", - "../../base:rtc_base_approved", "../../media:rtc_media", + "../../rtc_base:rtc_base_approved", "../../test:test_support", "../../test:video_test_support", ] @@ -391,7 +391,7 @@ if (rtc_include_tests) { ] deps += [ - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../sdk/android:libjingle_peerconnection_jni", "//base", ] @@ -428,8 +428,8 @@ if (rtc_include_tests) { ":webrtc_vp8", ":webrtc_vp9", "../../api:video_frame_api", - "../../base:rtc_base_approved", "../../common_video:common_video", + "../../rtc_base:rtc_base_approved", "../../test:test_support", "../../test:video_test_common", "../video_capture", @@ -483,7 +483,7 @@ if (rtc_include_tests) { if (is_android) { deps += [ - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", # TODO(brandtr): Figure out if the java dep below could be moved into # :video_coding_videoprocessor_integration_test, where it belongs. @@ -575,10 +575,10 @@ if (rtc_include_tests) { "../..:webrtc_common", "../../api:video_frame_api", "../../api/video_codecs:video_codecs_api", - "../../base:rtc_base", - "../../base:rtc_base_approved", - "../../base:rtc_task_queue", "../../common_video:common_video", + "../../rtc_base:rtc_base", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_task_queue", "../../system_wrappers:metrics_default", "../../system_wrappers:system_wrappers", "../../test:field_trial", diff --git a/webrtc/modules/video_processing/BUILD.gn b/webrtc/modules/video_processing/BUILD.gn index c4c9c3b894..6afc5f7f35 100644 --- a/webrtc/modules/video_processing/BUILD.gn +++ b/webrtc/modules/video_processing/BUILD.gn @@ -27,10 +27,10 @@ rtc_static_library("video_processing") { deps = [ ":denoiser_filter", "..:module_api", - "../../base:rtc_base_approved", "../../common_audio", "../../common_video", "../../modules/utility", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] if (build_video_processing_sse2) { @@ -66,7 +66,7 @@ if (build_video_processing_sse2) { deps = [ ":denoiser_filter", - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers", ] diff --git a/webrtc/ortc/BUILD.gn b/webrtc/ortc/BUILD.gn index b6a2cc9459..40c0b2a22b 100644 --- a/webrtc/ortc/BUILD.gn +++ b/webrtc/ortc/BUILD.gn @@ -35,8 +35,6 @@ rtc_static_library("ortc") { deps = [ "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", - "../base:rtc_base", - "../base:rtc_base_approved", "../call:call_interfaces", "../logging:rtc_event_log_api", "../media:rtc_media", @@ -45,6 +43,8 @@ rtc_static_library("ortc") { "../p2p:rtc_p2p", "../pc:libjingle_peerconnection", "../pc:rtc_pc", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", ] public_deps = [ @@ -76,14 +76,14 @@ if (rtc_include_tests) { deps = [ ":ortc", - "../base:rtc_base", - "../base:rtc_base_approved", - "../base:rtc_base_tests_main", - "../base:rtc_base_tests_utils", "../media:rtc_media_tests_utils", "../p2p:p2p_test_utils", "../p2p:rtc_p2p", "../pc:pc_test_utils", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_main", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", ] diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn index f7d59058ba..5963c925b5 100644 --- a/webrtc/p2p/BUILD.gn +++ b/webrtc/p2p/BUILD.gn @@ -87,7 +87,7 @@ rtc_static_library("rtc_p2p") { deps = [ "../api:libjingle_peerconnection_api", "../api:ortc_api", - "../base:rtc_base", + "../rtc_base:rtc_base", "../system_wrappers:field_trial_api", ] @@ -155,9 +155,9 @@ if (rtc_include_tests) { deps = [ ":rtc_p2p", "../api:ortc_api", - "../base:rtc_base", - "../base:rtc_base_approved", - "../base:rtc_base_tests_utils", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", "../test:test_support", "//testing/gmock", ] @@ -209,9 +209,9 @@ if (rtc_include_tests) { ":rtc_p2p", "../api:fakemetricsobserver", "../api:ortc_api", - "../base:rtc_base", - "../base:rtc_base_approved", - "../base:rtc_base_tests_utils", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", "../test:test_support", "//testing/gmock", "//testing/gtest", @@ -238,7 +238,7 @@ rtc_static_library("libstunprober") { deps = [ ":rtc_p2p", "..:webrtc_common", - "../base:rtc_base", + "../rtc_base:rtc_base", ] } @@ -259,8 +259,8 @@ if (rtc_include_tests) { ":libstunprober", ":p2p_test_utils", ":rtc_p2p", - "../base:rtc_base", - "../base:rtc_base_tests_utils", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_tests_utils", "//testing/gmock", "//testing/gtest", ] diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn index 2ff1a0af50..768a25cf41 100644 --- a/webrtc/pc/BUILD.gn +++ b/webrtc/pc/BUILD.gn @@ -60,12 +60,12 @@ rtc_static_library("rtc_pc_base") { "../api:call_api", "../api:libjingle_peerconnection_api", "../api:ortc_api", - "../base:rtc_base", - "../base:rtc_task_queue", "../media:rtc_data", "../media:rtc_h264_profile_id", "../media:rtc_media_base", "../p2p:rtc_p2p", + "../rtc_base:rtc_base", + "../rtc_base:rtc_task_queue", ] if (rtc_build_libsrtp) { @@ -165,13 +165,13 @@ rtc_static_library("peerconnection") { "../api:call_api", "../api:rtc_stats_api", "../api/video_codecs:video_codecs_api", - "../base:rtc_base", - "../base:rtc_base_approved", "../call:call_interfaces", "../logging:rtc_event_log_api", "../media:rtc_data", "../media:rtc_media_base", "../p2p:rtc_p2p", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", "../stats", "../system_wrappers:system_wrappers", ] @@ -198,14 +198,14 @@ rtc_static_library("create_pc_factory") { "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", - "../base:rtc_base", - "../base:rtc_base_approved", "../call", "../call:call_interfaces", "../logging:rtc_event_log_api", "../media:rtc_audio_video", "../modules/audio_device:audio_device", "../modules/audio_processing:audio_processing", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", ] configs += [ ":libjingle_peerconnection_warnings_config" ] @@ -279,15 +279,15 @@ if (rtc_include_tests) { deps = [ ":libjingle_peerconnection", ":rtc_pc", - "../base:rtc_base", - "../base:rtc_base_approved", - "../base:rtc_base_tests_main", - "../base:rtc_base_tests_utils", "../logging:rtc_event_log_api", "../media:rtc_media_base", "../media:rtc_media_tests_utils", "../p2p:p2p_test_utils", "../p2p:rtc_p2p", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_main", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", ] @@ -325,15 +325,15 @@ if (rtc_include_tests) { "..:webrtc_common", "../api:libjingle_peerconnection_test_api", "../api:rtc_stats_api", - "../base:rtc_base", - "../base:rtc_base_approved", - "../base:rtc_base_tests_utils", "../call:call_interfaces", "../logging:rtc_event_log_api", "../media:rtc_media", "../media:rtc_media_tests_utils", "../modules/audio_device:audio_device", "../p2p:p2p_test_utils", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", "../test:test_support", "//testing/gmock", ] @@ -442,10 +442,10 @@ if (rtc_include_tests) { ":pc_test_utils", "..:webrtc_common", "../api:fakemetricsobserver", - "../base:rtc_base_tests_main", - "../base:rtc_base_tests_utils", "../media:rtc_media_tests_utils", "../pc:rtc_pc", + "../rtc_base:rtc_base_tests_main", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", "../test:audio_codec_mocks", "//testing/gmock", diff --git a/webrtc/rtc_base/BUILD.gn b/webrtc/rtc_base/BUILD.gn index 9e905820d2..98accaf131 100644 --- a/webrtc/rtc_base/BUILD.gn +++ b/webrtc/rtc_base/BUILD.gn @@ -203,8 +203,8 @@ rtc_static_library("rtc_base_approved") { # Dependency on chromium's logging (in //base). deps += [ "//base:base" ] sources += [ - "../../webrtc_overrides/webrtc/base/logging.cc", - "../../webrtc_overrides/webrtc/base/logging.h", + "../../webrtc_overrides/webrtc/rtc_base/logging.cc", + "../../webrtc_overrides/webrtc/rtc_base/logging.h", ] } else { sources += [ @@ -301,8 +301,8 @@ rtc_static_library("rtc_task_queue") { if (build_with_chromium) { sources = [ - "../../webrtc_overrides/webrtc/base/task_queue.cc", - "../../webrtc_overrides/webrtc/base/task_queue.h", + "../../webrtc_overrides/webrtc/rtc_base/task_queue.cc", + "../../webrtc_overrides/webrtc/rtc_base/task_queue.h", ] } else { sources = [ @@ -517,7 +517,7 @@ rtc_static_library("rtc_base") { if (build_with_chromium) { if (is_win) { - sources += [ "../../webrtc_overrides/webrtc/base/win32socketinit.cc" ] + sources += [ "../../webrtc_overrides/webrtc/rtc_base/win32socketinit.cc" ] } include_dirs = [ "../../boringssl/src/include" ] public_configs += [ ":rtc_base_chromium_config" ] diff --git a/webrtc/rtc_base/callback.h.pump b/webrtc/rtc_base/callback.h.pump index 23899526b9..cceddf7343 100644 --- a/webrtc/rtc_base/callback.h.pump +++ b/webrtc/rtc_base/callback.h.pump @@ -57,8 +57,8 @@ #ifndef WEBRTC_RTC_BASE_CALLBACK_H_ #define WEBRTC_RTC_BASE_CALLBACK_H_ -#include "webrtc/base/refcount.h" -#include "webrtc/base/scoped_ref_ptr.h" +#include "webrtc/rtc_base/refcount.h" +#include "webrtc/rtc_base/scoped_ref_ptr.h" namespace rtc { diff --git a/webrtc/rtc_base/sigslottester.h.pump b/webrtc/rtc_base/sigslottester.h.pump index a88f0c6616..381b7914fd 100755 --- a/webrtc/rtc_base/sigslottester.h.pump +++ b/webrtc/rtc_base/sigslottester.h.pump @@ -35,8 +35,8 @@ // EXPECT_EQ("hello", capture); // /* See unit-tests for more examples */ -#include "webrtc/base/constructormagic.h" -#include "webrtc/base/sigslot.h" +#include "webrtc/rtc_base/constructormagic.h" +#include "webrtc/rtc_base/sigslot.h" namespace rtc { diff --git a/webrtc/rtc_tools/BUILD.gn b/webrtc/rtc_tools/BUILD.gn index e224380cc7..21a4aa82b0 100644 --- a/webrtc/rtc_tools/BUILD.gn +++ b/webrtc/rtc_tools/BUILD.gn @@ -48,8 +48,8 @@ rtc_static_library("command_line_parser") { "simple_command_line_parser.h", ] deps = [ - "../base:gtest_prod", - "../base:rtc_base_approved", + "../rtc_base:gtest_prod", + "../rtc_base:rtc_base_approved", ] } @@ -206,13 +206,13 @@ if (rtc_enable_protobuf) { defines = [ "ENABLE_RTC_EVENT_LOG" ] deps = [ "..:video_stream_api", - "../base:rtc_base_approved", "../call:call_interfaces", "../logging:rtc_event_log_impl", "../logging:rtc_event_log_parser", "../modules:module_api", "../modules/audio_coding:ana_debug_dump_proto", "../modules/audio_coding:neteq_tools", + "../rtc_base:rtc_base_approved", # TODO(kwiberg): Remove this dependency. "../api/audio_codecs:audio_codecs_api", @@ -245,7 +245,7 @@ if (rtc_include_tests) { defines = [ "ENABLE_RTC_EVENT_LOG" ] deps = [ ":event_log_visualizer_utils", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "../test:field_trial", "../test:test_support", ] @@ -264,9 +264,9 @@ if (rtc_include_tests) { } deps = [ - "../base:rtc_base_approved", "../modules:module_api", "../modules/audio_processing", + "../rtc_base:rtc_base_approved", "../system_wrappers:metrics_default", "../test:test_support", "//build/win:default_exe_manifest", diff --git a/webrtc/rtc_tools/network_tester/BUILD.gn b/webrtc/rtc_tools/network_tester/BUILD.gn index bd069d6109..49a625d527 100644 --- a/webrtc/rtc_tools/network_tester/BUILD.gn +++ b/webrtc/rtc_tools/network_tester/BUILD.gn @@ -41,10 +41,10 @@ if (rtc_enable_protobuf) { deps = [ ":network_tester_config_proto", ":network_tester_packet_proto", - "../../base:protobuf_utils", - "../../base:rtc_task_queue", - "../../base:sequenced_task_checker", "../../p2p", + "../../rtc_base:protobuf_utils", + "../../rtc_base:rtc_task_queue", + "../../rtc_base:sequenced_task_checker", ] if (!build_with_chromium && is_clang) { @@ -84,7 +84,7 @@ if (rtc_enable_protobuf) { deps = [ ":network_tester", "//testing/gtest", - "//webrtc/base:rtc_base_tests_utils", + "//webrtc/rtc_base:rtc_base_tests_utils", "//webrtc/test:test_support", ] diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn index 0a15fc527b..b2b396bb11 100644 --- a/webrtc/sdk/BUILD.gn +++ b/webrtc/sdk/BUILD.gn @@ -63,7 +63,7 @@ if (is_ios || is_mac) { ] deps = [ - "../base:rtc_base", + "../rtc_base:rtc_base", ] configs += [ "..:common_objc" ] @@ -98,7 +98,7 @@ if (is_ios || is_mac) { deps = [ ":objc_common", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] if (is_clang) { @@ -127,9 +127,9 @@ if (is_ios || is_mac) { ":objc_common", "../api:libjingle_peerconnection_api", "../api:video_frame_api", - "../base:rtc_base", "../common_video", "../media:rtc_media_base", + "../rtc_base:rtc_base", ] configs += [ "..:common_objc" ] @@ -181,9 +181,9 @@ if (is_ios || is_mac) { ":objc_common", ":objc_videotracksource", "../api:libjingle_peerconnection_api", - "../base:rtc_base", "../common_video", "../media:rtc_media_base", + "../rtc_base:rtc_base", ] configs += [ "..:common_objc" ] @@ -247,7 +247,7 @@ if (is_ios || is_mac) { deps = [ ":objc_video", "../api:video_frame_api", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] configs += [ "..:common_objc" ] public_configs = [ ":objc_common_config" ] @@ -289,9 +289,9 @@ if (is_ios || is_mac) { ":objc_peerconnectionfactory", ":objc_video", "../api:video_frame_api", - "../base:rtc_base", "../media:rtc_media_base", "../pc:libjingle_peerconnection", + "../rtc_base:rtc_base", ] if (rtc_use_metal_rendering) { @@ -334,12 +334,12 @@ if (is_ios || is_mac) { ":objc_videotracksource", "../api:video_frame_api", "../api/video_codecs:video_codecs_api", - "../base:rtc_base", "../media:rtc_audio_video", "../media:rtc_media_base", "../modules:module_api", "../pc:create_pc_factory", "../pc:peerconnection", + "../rtc_base:rtc_base", "../system_wrappers:field_trial_api", ] } @@ -371,7 +371,7 @@ if (is_ios || is_mac) { deps = [ ":objc_peerconnectionfactory_base", "../api:libjingle_peerconnection_api", - "../base:rtc_base", + "../rtc_base:rtc_base", ] } @@ -484,11 +484,11 @@ if (is_ios || is_mac) { ":objc_corevideoframebuffer", ":objc_videotracksource", "../api:video_frame_api", - "../base:rtc_base", "../common_video", "../media:rtc_media_base", "../modules:module_api", "../pc:peerconnection", + "../rtc_base:rtc_base", ] } @@ -530,7 +530,7 @@ if (is_ios || is_mac) { deps = [ ":objc_peerconnection", "..//system_wrappers:system_wrappers_default", - "../base:rtc_base_tests_utils", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers:system_wrappers_default", "//third_party/ocmock", ] @@ -632,7 +632,7 @@ if (is_ios || is_mac) { ":objc_audio", ":objc_peerconnection", ":objc_ui", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "../system_wrappers:field_trial_default", "../system_wrappers:metrics_default", ] @@ -673,8 +673,8 @@ if (is_ios || is_mac) { ] deps = [ - "../base:rtc_base_approved", "../common_video", + "../rtc_base:rtc_base_approved", ] if (!build_with_chromium && is_clang) { @@ -705,13 +705,13 @@ if (is_ios || is_mac) { ":objc_common", ":objc_video", ":objc_videotracksource", - "../base:rtc_base_approved", "../common_video", "../media:rtc_media", "../media:rtc_media_base", "../modules:module_api", "../modules/video_coding:video_coding_utility", "../modules/video_coding:webrtc_h264", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] diff --git a/webrtc/sdk/android/BUILD.gn b/webrtc/sdk/android/BUILD.gn index 689d6cfb40..d9cf609db9 100644 --- a/webrtc/sdk/android/BUILD.gn +++ b/webrtc/sdk/android/BUILD.gn @@ -49,8 +49,8 @@ rtc_source_set("base_jni") { deps = [ "//webrtc/api:libjingle_peerconnection_api", - "//webrtc/base:rtc_base", - "//webrtc/base:rtc_base_approved", + "//webrtc/rtc_base:rtc_base", + "//webrtc/rtc_base:rtc_base_approved", "//webrtc/system_wrappers:metrics_api", ] @@ -139,16 +139,16 @@ rtc_static_library("video_jni") { "//webrtc/api:libjingle_peerconnection_api", "//webrtc/api:video_frame_api", "//webrtc/api/video_codecs:video_codecs_api", - "//webrtc/base:rtc_base", - "//webrtc/base:rtc_base_approved", - "//webrtc/base:rtc_task_queue", - "//webrtc/base:sequenced_task_checker", - "//webrtc/base:weak_ptr", "//webrtc/common_video:common_video", "//webrtc/media:rtc_audio_video", "//webrtc/media:rtc_media_base", "//webrtc/modules/utility:utility", "//webrtc/modules/video_coding:video_coding_utility", + "//webrtc/rtc_base:rtc_base", + "//webrtc/rtc_base:rtc_base_approved", + "//webrtc/rtc_base:rtc_task_queue", + "//webrtc/rtc_base:sequenced_task_checker", + "//webrtc/rtc_base:weak_ptr", "//webrtc/system_wrappers:system_wrappers", ] @@ -237,13 +237,13 @@ rtc_static_library("peerconnection_jni") { deps = [ ":base_jni", "../..:webrtc_common", - "//webrtc/base:rtc_base", - "//webrtc/base:rtc_base_approved", - "//webrtc/base:rtc_task_queue", "//webrtc/media:rtc_data", "//webrtc/media:rtc_media_base", "//webrtc/modules/utility:utility", "//webrtc/pc:peerconnection", + "//webrtc/rtc_base:rtc_base", + "//webrtc/rtc_base:rtc_base_approved", + "//webrtc/rtc_base:rtc_task_queue", "//webrtc/system_wrappers:system_wrappers", ] } @@ -294,9 +294,9 @@ rtc_shared_library("libjingle_peerconnection_datachannelonly_so") { ":null_media_jni", ":null_video_jni", ":peerconnection_jni", - "//webrtc/base:rtc_base", - "//webrtc/base:rtc_base_approved", "//webrtc/pc:peerconnection", + "//webrtc/rtc_base:rtc_base", + "//webrtc/rtc_base:rtc_base_approved", ] output_extension = "so" } @@ -312,8 +312,8 @@ rtc_shared_library("libjingle_peerconnection_so") { deps = [ ":libjingle_peerconnection_jni", ":libjingle_peerconnection_metrics_default_jni", - "//webrtc/base:rtc_base", "//webrtc/pc:libjingle_peerconnection", + "//webrtc/rtc_base:rtc_base", ] output_extension = "so" } diff --git a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm index 4131a45242..5d6b6c60b1 100644 --- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm +++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm @@ -17,7 +17,7 @@ #import "WebRTC/RTCVideoFrame.h" #import "WebRTC/RTCVideoFrameBuffer.h" -#include "webrtc/base/timeutils.h" +#include "webrtc/rtc_base/timeutils.h" #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h" #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.h" #include "webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.h" diff --git a/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm b/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm index 61c8032206..c4d9bd14fa 100644 --- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm +++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_decoder_factory.mm @@ -18,11 +18,11 @@ #import "WebRTC/RTCVideoFrameBuffer.h" #include "webrtc/api/video_codecs/video_decoder.h" -#include "webrtc/base/logging.h" -#include "webrtc/base/timeutils.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/include/video_codec_interface.h" #include "webrtc/modules/video_coding/include/video_error_codes.h" +#include "webrtc/rtc_base/logging.h" +#include "webrtc/rtc_base/timeutils.h" #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h" namespace webrtc { diff --git a/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm b/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm index e63c527cb3..a063237c7c 100644 --- a/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm +++ b/webrtc/sdk/objc/Framework/Classes/PeerConnection/objc_video_encoder_factory.mm @@ -19,11 +19,11 @@ #include "webrtc/api/video/video_frame.h" #include "webrtc/api/video_codecs/video_encoder.h" -#include "webrtc/base/logging.h" -#include "webrtc/base/timeutils.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/include/video_codec_interface.h" #include "webrtc/modules/video_coding/include/video_error_codes.h" +#include "webrtc/rtc_base/logging.h" +#include "webrtc/rtc_base/timeutils.h" #include "webrtc/sdk/objc/Framework/Classes/Common/helpers.h" #include "webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h" diff --git a/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm b/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm index ec3e226a03..49cc812b81 100644 --- a/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm +++ b/webrtc/sdk/objc/Framework/UnitTests/RTCTracingTest.mm @@ -12,7 +12,7 @@ #include -#include "webrtc/base/gunit.h" +#include "webrtc/rtc_base/gunit.h" #import "NSString+StdString.h" #import "WebRTC/RTCTracing.h" diff --git a/webrtc/stats/BUILD.gn b/webrtc/stats/BUILD.gn index 4a2f578e5b..eaa6f5d491 100644 --- a/webrtc/stats/BUILD.gn +++ b/webrtc/stats/BUILD.gn @@ -24,7 +24,7 @@ rtc_static_library("rtc_stats") { deps = [ "../api:rtc_stats_api", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } @@ -58,9 +58,9 @@ if (rtc_include_tests) { ":rtc_stats", ":rtc_stats_test_utils", "../api:rtc_stats_api", - "../base:rtc_base_approved", - "../base:rtc_base_tests_main", - "../base:rtc_base_tests_utils", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_main", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers:metrics_default", "//testing/gmock", ] diff --git a/webrtc/system_wrappers/BUILD.gn b/webrtc/system_wrappers/BUILD.gn index 1cf1b6f3b0..7dfcff7a3d 100644 --- a/webrtc/system_wrappers/BUILD.gn +++ b/webrtc/system_wrappers/BUILD.gn @@ -107,10 +107,10 @@ rtc_static_library("system_wrappers") { cflags = [ "/wd4334" ] # Ignore warning on shift operator promotion. - # Windows needs //webrtc/base:rtc_base due to include of webrtc/base/win32.h - # in source/clock.cc. + # Windows needs //webrtc/rtc_base:rtc_base due to include of + # webrtc/rtc_base/win32.h in source/clock.cc. # TODO(kjellander): Remove (bugs.webrtc.org/6828) - deps += [ "../base:rtc_base" ] + deps += [ "../rtc_base:rtc_base" ] } if (is_win && is_clang) { @@ -118,7 +118,7 @@ rtc_static_library("system_wrappers") { suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] } - deps += [ "../base:rtc_base_approved" ] + deps += [ "../rtc_base:rtc_base_approved" ] } rtc_source_set("cpu_features_api") { @@ -148,7 +148,7 @@ rtc_source_set("metrics_api") { ] deps = [ "..:webrtc_common", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } @@ -169,7 +169,7 @@ rtc_static_library("metrics_default") { ] deps = [ ":metrics_api", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] } @@ -228,7 +228,7 @@ if (rtc_include_tests) { ":metrics_default", ":system_wrappers", "..:webrtc_common", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "../test:test_main", "//testing/gtest", ] diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn index f54a62252d..29d02e7981 100644 --- a/webrtc/test/BUILD.gn +++ b/webrtc/test/BUILD.gn @@ -60,11 +60,11 @@ rtc_source_set("video_test_common") { deps = [ "..:video_stream_api", "..:webrtc_common", - "../base:rtc_base_approved", - "../base:rtc_task_queue", "../common_video", "../media:rtc_media_base", "../modules/video_capture:video_capture_module", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue", "../system_wrappers", ] } @@ -87,8 +87,8 @@ rtc_source_set("rtp_test_utils") { deps = [ "..:webrtc_common", - "../base:rtc_base_approved", "../modules/rtp_rtcp", + "../rtc_base:rtc_base_approved", "//testing/gtest", ] } @@ -131,9 +131,9 @@ rtc_source_set("test_support") { deps = [ "..:webrtc_common", - "../base:gtest_prod", - "../base:rtc_base_approved", "../common_video", + "../rtc_base:gtest_prod", + "../rtc_base:rtc_base_approved", "../system_wrappers", "//testing/gmock", "//testing/gtest", @@ -178,7 +178,7 @@ if (!build_with_chromium) { ] deps = [ ":field_trial", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "../system_wrappers:metrics_default", "//testing/gmock", "//testing/gtest", @@ -205,8 +205,8 @@ if (!build_with_chromium) { ":test_support", ":video_test_common", "..:webrtc_common", - "../base:rtc_base_approved", "../common_video", + "../rtc_base:rtc_base_approved", "../system_wrappers", "//testing/gmock", "//testing/gtest", @@ -243,7 +243,7 @@ if (!build_with_chromium) { ] deps = [ ":fileutils", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "//third_party/gflags", ] } @@ -273,10 +273,10 @@ if (!build_with_chromium) { ":fake_audio_device", ":rtp_test_utils", "../api:video_frame_api", - "../base:rtc_base_approved", "../call:call_interfaces", "../common_audio", "../modules/rtp_rtcp", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] sources = [ @@ -342,14 +342,14 @@ rtc_source_set("fileutils") { ] deps = [ "..:webrtc_common", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", ] if (is_ios) { sources += [ "testsupport/iosfileutils.mm" ] deps += [ "../sdk:objc_common" ] } if (is_win) { - deps += [ "../base:rtc_base" ] + deps += [ "../rtc_base:rtc_base" ] } visibility = [ ":*" ] } @@ -375,7 +375,7 @@ rtc_source_set("fileutils_unittests") { deps = [ ":fileutils", ":test_support", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "//testing/gmock", "//testing/gtest", ] @@ -396,9 +396,9 @@ rtc_source_set("direct_transport") { deps = [ "..:webrtc_common", "../api:transport_api", - "../base:rtc_base_approved", "../call", "../modules/rtp_rtcp", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] } @@ -415,9 +415,9 @@ rtc_source_set("fake_audio_device") { } deps = [ "..:webrtc_common", - "../base:rtc_base_approved", "../common_audio:common_audio", "../modules/audio_device:audio_device", + "../rtc_base:rtc_base_approved", "../system_wrappers:system_wrappers", ] } @@ -478,9 +478,6 @@ rtc_source_set("test_common") { "../api/audio_codecs:builtin_audio_encoder_factory", "../api/video_codecs:video_codecs_api", "../audio", - "../base:rtc_base_approved", - "../base:rtc_task_queue", - "../base:sequenced_task_checker", "../call", "../common_video", "../logging:rtc_event_log_api", @@ -492,6 +489,9 @@ rtc_source_set("test_common") { "../modules/video_coding:webrtc_h264", "../modules/video_coding:webrtc_vp8", "../modules/video_coding:webrtc_vp9", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue", + "../rtc_base:sequenced_task_checker", "../system_wrappers", "../video", "../voice_engine", @@ -571,9 +571,9 @@ rtc_source_set("test_renderer") { deps = [ ":test_support", "..:webrtc_common", - "../base:rtc_base_approved", "../common_video", "../modules/media_file", + "../rtc_base:rtc_base_approved", "//testing/gtest", ] } @@ -593,7 +593,7 @@ rtc_source_set("audio_codec_mocks") { ":test_support", "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_decoder_factory", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "//testing/gmock", ] } diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn index 3e68470749..b206c72d6c 100644 --- a/webrtc/test/fuzzers/BUILD.gn +++ b/webrtc/test/fuzzers/BUILD.gn @@ -15,7 +15,7 @@ rtc_static_library("webrtc_fuzzer_main") { "webrtc_fuzzer_main.cc", ] deps = [ - "../../base:rtc_base_approved", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:field_trial_default", "../../system_wrappers:metrics_default", "//testing/libfuzzer:libfuzzer_main", @@ -95,8 +95,8 @@ webrtc_fuzzer_test("flexfec_header_reader_fuzzer") { "flexfec_header_reader_fuzzer.cc", ] deps = [ - "../../base:rtc_base_approved", "../../modules/rtp_rtcp", + "../../rtc_base:rtc_base_approved", ] } @@ -116,9 +116,9 @@ webrtc_fuzzer_test("ulpfec_header_reader_fuzzer") { "ulpfec_header_reader_fuzzer.cc", ] deps = [ - "../../base:rtc_base_approved", "../../modules/rtp_rtcp", "../../modules/rtp_rtcp:fec_test_helper", + "../../rtc_base:rtc_base_approved", ] } @@ -127,9 +127,9 @@ webrtc_fuzzer_test("ulpfec_generator_fuzzer") { "ulpfec_generator_fuzzer.cc", ] deps = [ - "../../base:rtc_base_approved", "../../modules/rtp_rtcp", "../../modules/rtp_rtcp:fec_test_helper", + "../../rtc_base:rtc_base_approved", ] } @@ -138,8 +138,8 @@ webrtc_fuzzer_test("flexfec_receiver_fuzzer") { "flexfec_receiver_fuzzer.cc", ] deps = [ - "../../base:rtc_base_approved", "../../modules/rtp_rtcp", + "../../rtc_base:rtc_base_approved", ] libfuzzer_options = [ "max_len=2000" ] } @@ -160,8 +160,8 @@ webrtc_fuzzer_test("rtcp_receiver_fuzzer") { "rtcp_receiver_fuzzer.cc", ] deps = [ - "../../base:rtc_base_approved", "../../modules/rtp_rtcp", + "../../rtc_base:rtc_base_approved", "../../system_wrappers:system_wrappers", ] seed_corpus = "corpora/rtcp-corpus" @@ -207,8 +207,8 @@ rtc_static_library("audio_decoder_fuzzer") { deps = [ "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", - "../../base:rtc_base_approved", "../../modules/rtp_rtcp", + "../../rtc_base:rtc_base_approved", ] } @@ -286,13 +286,13 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") { "neteq_rtp_fuzzer.cc", ] deps = [ - "../../base:rtc_base_approved", - "../../base:rtc_base_tests_utils", "../../modules/audio_coding:neteq", "../../modules/audio_coding:neteq_test_tools", "../../modules/audio_coding:neteq_tools_minimal", "../../modules/audio_coding:pcm16b", "../../modules/rtp_rtcp", + "../../rtc_base:rtc_base_approved", + "../../rtc_base:rtc_base_tests_utils", ] } @@ -301,8 +301,8 @@ webrtc_fuzzer_test("residual_echo_detector_fuzzer") { "residual_echo_detector_fuzzer.cc", ] deps = [ - "../../base:rtc_base_approved", "../../modules/audio_processing:audio_processing", + "../../rtc_base:rtc_base_approved", ] } @@ -343,8 +343,8 @@ webrtc_fuzzer_test("pseudotcp_parser_fuzzer") { "pseudotcp_parser_fuzzer.cc", ] deps = [ - "../../base:rtc_base", "../../p2p:rtc_p2p", + "../../rtc_base:rtc_base", ] } @@ -353,8 +353,8 @@ webrtc_fuzzer_test("transport_feedback_packet_loss_tracker_fuzzer") { "transport_feedback_packet_loss_tracker_fuzzer.cc", ] deps = [ - "../../base:rtc_base_approved", "../../modules/rtp_rtcp", + "../../rtc_base:rtc_base_approved", "../../voice_engine", ] } @@ -366,8 +366,8 @@ webrtc_fuzzer_test("audio_processing_fuzzer") { "audio_processing_fuzzer_configs.cc", ] deps = [ - "../../base:rtc_base_approved", "../../modules:module_api", "../../modules/audio_processing", + "../../rtc_base:rtc_base_approved", ] } diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn index 5faf28048e..529a73c100 100644 --- a/webrtc/video/BUILD.gn +++ b/webrtc/video/BUILD.gn @@ -58,11 +58,6 @@ rtc_static_library("video") { "..:webrtc_common", "../api:transport_api", "../api/video_codecs:video_codecs_api", - "../base:rtc_base_approved", - "../base:rtc_numerics", - "../base:rtc_task_queue", - "../base:sequenced_task_checker", - "../base:weak_ptr", "../call:call_interfaces", "../call:rtp_interfaces", "../common_video", @@ -79,6 +74,11 @@ rtc_static_library("video") { "../modules/video_coding:video_coding_utility", "../modules/video_coding:webrtc_vp8", "../modules/video_processing", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_numerics", + "../rtc_base:rtc_task_queue", + "../rtc_base:sequenced_task_checker", + "../rtc_base:weak_ptr", "../system_wrappers", "../voice_engine", ] @@ -93,8 +93,6 @@ if (rtc_include_tests) { "video_quality_test.h", ] deps = [ - "../base:rtc_base_tests_utils", - "../base:rtc_task_queue", "../call:call_interfaces", "../common_video", "../logging:rtc_event_log_api", @@ -105,6 +103,8 @@ if (rtc_include_tests) { "../modules/video_coding:webrtc_h264", "../modules/video_coding:webrtc_vp8", "../modules/video_coding:webrtc_vp9", + "../rtc_base:rtc_base_tests_utils", + "../rtc_base:rtc_task_queue", "../system_wrappers", "../test:test_common", "../test:test_support", @@ -155,7 +155,7 @@ if (rtc_include_tests) { ] deps = [ ":video_quality_test", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "../system_wrappers:metrics_default", "../test:field_trial", "../test:run_test", @@ -180,7 +180,7 @@ if (rtc_include_tests) { deps = [ ":video_quality_test", - "../base:rtc_base_approved", + "../rtc_base:rtc_base_approved", "../system_wrappers:metrics_default", "../test:field_trial", "../test:run_test", @@ -203,11 +203,11 @@ if (rtc_include_tests) { deps = [ "..:webrtc_common", "../api/video_codecs:video_codecs_api", - "../base:rtc_base_approved", "../call:call_interfaces", "../common_video", "../logging:rtc_event_log_api", "../modules/rtp_rtcp", + "../rtc_base:rtc_base_approved", "../system_wrappers", "../system_wrappers:metrics_default", "../test:field_trial", @@ -260,8 +260,6 @@ if (rtc_include_tests) { "..:video_stream_api", "../api:video_frame_api", "../api/video_codecs:video_codecs_api", - "../base:rtc_base_approved", - "../base:rtc_base_tests_utils", "../call:call_interfaces", "../call:rtp_receiver", "../common_video", @@ -279,6 +277,8 @@ if (rtc_include_tests) { "../modules/video_coding:webrtc_h264", "../modules/video_coding:webrtc_vp8", "../modules/video_coding:webrtc_vp9", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", "../system_wrappers", "../system_wrappers:field_trial_default", "../system_wrappers:metrics_api", diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index e16b1762a7..78c92f6ec6 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -37,10 +37,10 @@ rtc_static_library("file_player") { deps = [ ":audio_coder", "..:webrtc_common", - "../base:rtc_base_approved", "../common_audio", "../modules:module_api", "../modules/media_file", + "../rtc_base:rtc_base_approved", ] if (!build_with_chromium && is_clang) { @@ -58,10 +58,10 @@ rtc_static_library("file_recorder") { ":audio_coder", "..:webrtc_common", "../audio/utility:audio_frame_operations", - "../base:rtc_base_approved", "../common_audio", "../modules:module_api", "../modules/media_file:media_file", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] @@ -143,8 +143,6 @@ rtc_static_library("voice_engine") { "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", "../audio/utility:audio_frame_operations", - "../base:rtc_base_approved", - "../base:rtc_task_queue", "../call:rtp_interfaces", "../common_audio", "../logging:rtc_event_log_api", @@ -159,6 +157,8 @@ rtc_static_library("voice_engine") { "../modules/pacing", "../modules/rtp_rtcp", "../modules/utility", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue", "../system_wrappers", ] } @@ -171,9 +171,9 @@ rtc_static_library("audio_level") { deps = [ "..:webrtc_common", - "../base:rtc_base_approved", "../common_audio", "../modules:module_api", + "../rtc_base:rtc_base_approved", ] } @@ -182,9 +182,9 @@ if (rtc_include_tests) { deps = [ ":file_player", ":voice_engine", - "../base:rtc_base_approved", - "../base:rtc_base_tests_utils", "../modules:module_api", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", "../test:test_common", "//testing/gmock", "//testing/gtest", @@ -247,11 +247,11 @@ if (rtc_include_tests) { deps = [ ":voice_engine", "..:webrtc_common", - "../base:rtc_base_approved", "../modules:module_api", "../modules/audio_device:audio_device", "../modules/audio_processing:audio_processing", "../modules/rtp_rtcp:rtp_rtcp", + "../rtc_base:rtc_base_approved", "//testing/gmock", "//testing/gtest", "//third_party/gflags",