blob: cadaaa2a7f03b2b35923d4c988a6a7bc7a84b4ff [file] [log] [blame]
# This is supposed to be a complete list of top-level directories,
# excepting only api/ itself.
include_rules = [
"-audio",
"-base",
"-build",
"-buildtools",
"-build_overrides",
"-call",
"-common_audio",
"-common_video",
"-data",
"-examples",
"-g3doc",
"-ios",
"-infra",
"-logging",
"-media",
"-net",
"-modules",
"-out",
"-p2p",
"-pc",
"-resources",
"-rtc_base",
"-rtc_tools",
"-sdk",
"-stats",
"-style-guide",
"-system_wrappers",
"-test",
"-testing",
"-third_party",
"-tools",
"-tools_webrtc",
"-video",
"-external/webrtc/webrtc", # Android platform build.
"-libyuv",
"-common_types.h",
"-WebRTC",
]
specific_include_rules = {
# Some internal headers are allowed even in API headers:
"call_factory_interface\.h": [
"+call/rtp_transport_controller_send_factory_interface.h",
],
".*\.h": [
"+rtc_base/checks.h",
"+rtc_base/system/rtc_export.h",
"+rtc_base/system/rtc_export_template.h",
"+rtc_base/units/unit_base.h",
],
"array_view\.h": [
"+rtc_base/type_traits.h",
],
# Needed because AudioEncoderOpus is in the wrong place for
# backwards compatibilty reasons. See
# https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
"audio_encoder_opus\.h": [
"+modules/audio_coding/codecs/opus/audio_encoder_opus.h",
],
"async_resolver_factory\.h": [
"+rtc_base/async_resolver_interface.h",
],
"async_dns_resolver\.h": [
"+rtc_base/socket_address.h",
],
"candidate\.h": [
"+rtc_base/network_constants.h",
"+rtc_base/socket_address.h",
],
"data_channel_interface\.h": [
"+rtc_base/copy_on_write_buffer.h",
"+rtc_base/ref_count.h",
],
"data_channel_transport_interface\.h": [
"+rtc_base/copy_on_write_buffer.h",
],
"dtls_transport_interface\.h": [
"+rtc_base/ref_count.h",
"+rtc_base/ssl_certificate.h",
],
"dtmf_sender_interface\.h": [
"+rtc_base/ref_count.h",
],
"fec_controller\.h": [
"+modules/include/module_fec_types.h",
],
"frame_transformer_interface\.h": [
"+rtc_base/ref_count.h",
],
"ice_transport_interface\.h": [
"+rtc_base/ref_count.h",
],
"jsep\.h": [
"+rtc_base/ref_count.h",
],
"jsep_ice_candidate\.h": [
"+rtc_base/constructor_magic.h",
],
"jsep_session_description\.h": [
"+rtc_base/constructor_magic.h",
],
"media_stream_interface\.h": [
"+modules/audio_processing/include/audio_processing_statistics.h",
"+rtc_base/ref_count.h",
],
"packet_socket_factory\.h": [
"+rtc_base/proxy_info.h",
"+rtc_base/async_packet_socket.h",
],
"peer_connection_interface\.h": [
"+call/rtp_transport_controller_send_factory_interface.h",
"+media/base/media_config.h",
"+media/base/media_engine.h",
"+p2p/base/port.h",
"+p2p/base/port_allocator.h",
"+rtc_base/network.h",
"+rtc_base/network_constants.h",
"+rtc_base/network_monitor_factory.h",
"+rtc_base/ref_count.h",
"+rtc_base/rtc_certificate.h",
"+rtc_base/rtc_certificate_generator.h",
"+rtc_base/socket_address.h",
"+rtc_base/ssl_certificate.h",
"+rtc_base/ssl_stream_adapter.h",
"+rtc_base/thread.h",
],
"proxy\.h": [
"+rtc_base/event.h",
"+rtc_base/message_handler.h", # Inherits from it.
"+rtc_base/ref_counted_object.h",
"+rtc_base/thread.h",
],
"ref_counted_base\.h": [
"+rtc_base/constructor_magic.h",
"+rtc_base/ref_count.h",
"+rtc_base/ref_counter.h",
],
"rtc_error\.h": [
"+rtc_base/logging.h",
],
"rtc_event_log_output_file.h": [
# For private member and constructor.
"+rtc_base/system/file_wrapper.h",
],
"rtp_receiver_interface\.h": [
"+rtc_base/ref_count.h",
],
"rtp_sender_interface\.h": [
"+rtc_base/ref_count.h",
],
"rtp_transceiver_interface\.h": [
"+rtc_base/ref_count.h",
],
"sctp_transport_interface\.h": [
"+rtc_base/ref_count.h",
],
"set_local_description_observer_interface\.h": [
"+rtc_base/ref_count.h",
],
"set_remote_description_observer_interface\.h": [
"+rtc_base/ref_count.h",
],
"stats_types\.h": [
"+rtc_base/constructor_magic.h",
"+rtc_base/ref_count.h",
"+rtc_base/thread_checker.h",
],
"uma_metrics\.h": [
"+rtc_base/ref_count.h",
],
"audio_frame\.h": [
"+rtc_base/constructor_magic.h",
],
"audio_mixer\.h": [
"+rtc_base/ref_count.h",
],
"audio_decoder\.h": [
"+rtc_base/buffer.h",
"+rtc_base/constructor_magic.h",
],
"audio_decoder_factory\.h": [
"+rtc_base/ref_count.h",
],
"audio_decoder_factory_template\.h": [
"+rtc_base/ref_counted_object.h",
],
"audio_encoder\.h": [
"+rtc_base/buffer.h",
],
"audio_encoder_factory\.h": [
"+rtc_base/ref_count.h",
],
"audio_encoder_factory_template\.h": [
"+rtc_base/ref_counted_object.h",
],
"frame_decryptor_interface\.h": [
"+rtc_base/ref_count.h",
],
"frame_encryptor_interface\.h": [
"+rtc_base/ref_count.h",
],
"rtc_stats_collector_callback\.h": [
"+rtc_base/ref_count.h",
],
"rtc_stats_report\.h": [
"+rtc_base/ref_count.h",
"+rtc_base/ref_counted_object.h",
],
"audioproc_float\.h": [
"+modules/audio_processing/include/audio_processing.h",
],
"echo_detector_creator\.h": [
"+modules/audio_processing/include/audio_processing.h",
],
"fake_frame_decryptor\.h": [
"+rtc_base/ref_counted_object.h",
],
"fake_frame_encryptor\.h": [
"+rtc_base/ref_counted_object.h",
],
"mock.*\.h": [
"+test/gmock.h",
],
"mock_peerconnectioninterface\.h": [
"+rtc_base/ref_counted_object.h",
],
"mock_video_track\.h": [
"+rtc_base/ref_counted_object.h",
],
"simulated_network\.h": [
"+rtc_base/random.h",
"+rtc_base/thread_annotations.h",
],
"test_dependency_factory\.h": [
"+rtc_base/thread_checker.h",
],
"time_controller\.h": [
"+rtc_base/thread.h",
],
"videocodec_test_fixture\.h": [
"+modules/video_coding/include/video_codec_interface.h"
],
"video_encoder_config\.h": [
"+rtc_base/ref_count.h",
],
"sequence_checker\.h": [
"+rtc_base/synchronization/sequence_checker_internal.h",
"+rtc_base/thread_annotations.h",
],
"wrapping_async_dns_resolver\.h": [
"+rtc_base/async_resolver.h",
"+rtc_base/async_resolver_interface.h",
"+rtc_base/socket_address.h",
"+rtc_base/third_party/sigslot/sigslot.h",
"+rtc_base/thread_annotations.h",
],
# .cc files in api/ should not be restricted in what they can #include,
# so we re-add all the top-level directories here. (That's because .h
# files leak their #includes to whoever's #including them, but .cc files
# do not since no one #includes them.)
".*\.cc": [
"+audio",
"+call",
"+common_audio",
"+common_video",
"+examples",
"+logging",
"+media",
"+modules",
"+p2p",
"+pc",
"+rtc_base",
"+rtc_tools",
"+sdk",
"+stats",
"+system_wrappers",
"+test",
"+tools",
"+tools_webrtc",
"+video",
"+third_party",
],
}