6cae2d5513
This reverts commit 3f87250a4f0e6c69002fbcdfb995b0dfcd7bf710. Reason for revert: Downstream is fixed Original change's description: > Revert "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely" > > This reverts commit 5f0eb93d2a44cec2102fc8c3757d5bb814bd145f. > > Reason for revert: Breaks downstream project. I'm going to fix that one and create a reland of this CL after. > > Original change's description: > > Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely > > > > Bug: webrtc:13555, webrtc:13082 > > Change-Id: Iff2cda6f516739419e97e975e03f77a98f74be03 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249260 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Artem Titov <titovartem@webrtc.org> > > Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> > > Cr-Commit-Position: refs/heads/main@{#35805} > > TBR=hta@webrtc.org,titovartem@webrtc.org,daniel.l@hpcnt.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I33d497f1132adfe6d151023195a388d9b7d548f9 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:13555, webrtc:13082 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249364 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35807} # Not skipping CQ checks because this is a reland. Bug: webrtc:13555, webrtc:13082 Change-Id: I7ef1ef3b6e3c41b1a96014aa75f003c0fcf33949 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249365 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35814}
331 lines
6.7 KiB
Python
331 lines
6.7 KiB
Python
# This is supposed to be a complete list of top-level directories,
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# excepting only api/ itself.
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include_rules = [
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"-audio",
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"-base",
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"-build",
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"-buildtools",
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"-build_overrides",
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"-call",
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"-common_audio",
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"-common_video",
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"-data",
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"-examples",
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"-g3doc",
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"-ios",
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"-infra",
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"-logging",
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"-media",
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"-net",
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"-modules",
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"-out",
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"-p2p",
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"-pc",
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"-resources",
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"-rtc_base",
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"-rtc_tools",
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"-sdk",
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"-stats",
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"-style-guide",
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"-system_wrappers",
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"-test",
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"-testing",
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"-third_party",
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"-tools",
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"-tools_webrtc",
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"-video",
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"-external/webrtc/webrtc", # Android platform build.
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"-libyuv",
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"-common_types.h",
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"-WebRTC",
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]
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specific_include_rules = {
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# Some internal headers are allowed even in API headers:
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"call_factory_interface\.h": [
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"+call/rtp_transport_controller_send_factory_interface.h",
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],
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".*\.h": [
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"+rtc_base/checks.h",
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"+rtc_base/system/rtc_export.h",
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"+rtc_base/system/rtc_export_template.h",
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"+rtc_base/units/unit_base.h",
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],
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"array_view\.h": [
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"+rtc_base/type_traits.h",
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],
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# Needed because AudioEncoderOpus is in the wrong place for
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# backwards compatibilty reasons. See
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# https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
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"audio_encoder_opus\.h": [
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"+modules/audio_coding/codecs/opus/audio_encoder_opus.h",
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],
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"async_resolver_factory\.h": [
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"+rtc_base/async_resolver_interface.h",
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],
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"async_dns_resolver\.h": [
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"+rtc_base/socket_address.h",
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],
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"candidate\.h": [
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"+rtc_base/network_constants.h",
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"+rtc_base/socket_address.h",
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],
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"data_channel_interface\.h": [
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"+rtc_base/copy_on_write_buffer.h",
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"+rtc_base/ref_count.h",
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],
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"data_channel_transport_interface\.h": [
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"+rtc_base/copy_on_write_buffer.h",
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],
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"dtls_transport_interface\.h": [
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"+rtc_base/ref_count.h",
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"+rtc_base/ssl_certificate.h",
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],
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"dtmf_sender_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"fec_controller\.h": [
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"+modules/include/module_fec_types.h",
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],
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"frame_transformer_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"ice_transport_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"jsep\.h": [
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"+rtc_base/ref_count.h",
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],
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"media_stream_interface\.h": [
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"+modules/audio_processing/include/audio_processing_statistics.h",
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"+rtc_base/ref_count.h",
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],
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"packet_socket_factory\.h": [
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"+rtc_base/proxy_info.h",
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"+rtc_base/async_packet_socket.h",
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],
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"peer_connection_interface\.h": [
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"+call/rtp_transport_controller_send_factory_interface.h",
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"+media/base/media_config.h",
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"+media/base/media_engine.h",
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"+p2p/base/port.h",
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"+p2p/base/port_allocator.h",
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"+rtc_base/network.h",
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"+rtc_base/network_constants.h",
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"+rtc_base/network_monitor_factory.h",
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"+rtc_base/ref_count.h",
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"+rtc_base/rtc_certificate.h",
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"+rtc_base/rtc_certificate_generator.h",
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"+rtc_base/socket_address.h",
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"+rtc_base/ssl_certificate.h",
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"+rtc_base/ssl_stream_adapter.h",
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"+rtc_base/thread.h",
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],
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"proxy\.h": [
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"+rtc_base/event.h",
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"+rtc_base/message_handler.h", # Inherits from it.
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"+rtc_base/ref_counted_object.h",
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"+rtc_base/thread.h",
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],
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"ref_counted_base\.h": [
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"+rtc_base/ref_count.h",
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"+rtc_base/ref_counter.h",
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],
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"rtc_error\.h": [
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"+rtc_base/logging.h",
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],
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"rtc_event_log_output_file.h": [
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# For private member and constructor.
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"+rtc_base/system/file_wrapper.h",
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],
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"rtp_packet_infos\.h": [
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"+rtc_base/ref_counted_object.h",
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],
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"rtp_receiver_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"rtp_sender_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"rtp_transceiver_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"sctp_transport_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"set_local_description_observer_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"set_remote_description_observer_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"stats_types\.h": [
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"+rtc_base/ref_count.h",
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"+rtc_base/thread_checker.h",
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],
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"uma_metrics\.h": [
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"+rtc_base/ref_count.h",
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],
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"audio_mixer\.h": [
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"+rtc_base/ref_count.h",
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],
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"audio_decoder\.h": [
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"+rtc_base/buffer.h",
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],
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"audio_decoder_factory\.h": [
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"+rtc_base/ref_count.h",
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],
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"audio_decoder_factory_template\.h": [
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"+rtc_base/ref_counted_object.h",
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],
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"audio_encoder\.h": [
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"+rtc_base/buffer.h",
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],
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"audio_encoder_factory\.h": [
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"+rtc_base/ref_count.h",
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],
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"audio_encoder_factory_template\.h": [
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"+rtc_base/ref_counted_object.h",
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],
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"frame_decryptor_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"frame_encryptor_interface\.h": [
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"+rtc_base/ref_count.h",
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],
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"rtc_stats_collector_callback\.h": [
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"+rtc_base/ref_count.h",
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],
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"rtc_stats_report\.h": [
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"+rtc_base/ref_count.h",
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"+rtc_base/ref_counted_object.h",
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],
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"audioproc_float\.h": [
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"+modules/audio_processing/include/audio_processing.h",
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],
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"echo_detector_creator\.h": [
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"+modules/audio_processing/include/audio_processing.h",
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],
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"fake_frame_decryptor\.h": [
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"+rtc_base/ref_counted_object.h",
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],
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"fake_frame_encryptor\.h": [
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"+rtc_base/ref_counted_object.h",
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],
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"mock.*\.h": [
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"+test/gmock.h",
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],
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"mock_peerconnectioninterface\.h": [
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"+rtc_base/ref_counted_object.h",
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],
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"mock_video_track\.h": [
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"+rtc_base/ref_counted_object.h",
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],
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"simulated_network\.h": [
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"+rtc_base/random.h",
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"+rtc_base/thread_annotations.h",
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],
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"test_dependency_factory\.h": [
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"+rtc_base/thread_checker.h",
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],
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"time_controller\.h": [
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"+rtc_base/thread.h",
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],
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"videocodec_test_fixture\.h": [
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"+modules/video_coding/include/video_codec_interface.h"
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],
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"video_encoder_config\.h": [
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"+rtc_base/ref_count.h",
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],
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"sequence_checker\.h": [
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"+rtc_base/synchronization/sequence_checker_internal.h",
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"+rtc_base/thread_annotations.h",
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],
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"wrapping_async_dns_resolver\.h": [
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"+rtc_base/async_resolver.h",
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"+rtc_base/async_resolver_interface.h",
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"+rtc_base/socket_address.h",
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"+rtc_base/third_party/sigslot/sigslot.h",
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"+rtc_base/thread_annotations.h",
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],
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# .cc files in api/ should not be restricted in what they can #include,
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# so we re-add all the top-level directories here. (That's because .h
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# files leak their #includes to whoever's #including them, but .cc files
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# do not since no one #includes them.)
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".*\.cc": [
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"+audio",
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"+call",
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"+common_audio",
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"+common_video",
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"+examples",
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"+logging",
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"+media",
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"+modules",
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"+p2p",
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"+pc",
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"+rtc_base",
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"+rtc_tools",
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"+sdk",
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"+stats",
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"+system_wrappers",
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"+test",
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"+tools",
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"+tools_webrtc",
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"+video",
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"+third_party",
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],
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}
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