d4089cae47
BACKGROUND When SLD is invoked with SetSessionDescriptionObserver, the observer is called by posting a message back to the execution thread, delaying the call. This delay is "artificial" - it's not necessary; the operation is already complete. It's a post from the signaling thread to the signaling thread. The rationale for the post was to avoid the observer making recursive calls back into the PeerConnection. The problem with this is that by the time the observer is called, the PeerConnection could already have executed other operations and modified its states. This causes the referenced bug: one can have a race where SLD is resolved "too late" (after a pending SRD is executed) and the signaling state observed when SLD resolves doesn't make sense. When implementing Unified Plan, we fixed similar issues for SRD by adding a version that takes SetRemoteDescriptionObserverInterface as argument instead of SetSessionDescriptionObserver. The new version did not have the delay. The old version had to be kept around not to break downstream projects that had dependencies both on he delay and on allowing the PC to be destroyed midst-operation without informing its observers. THIS CL This does the old SRD fix for SLD as well: A new observer interface is added, SetLocalDescriptionObserverInterface, and PeerConnection::SetLocalDescription() is overloaded. If you call it with the old observer, you get the delay, but if you call it with the new observer, you don't get a delay. - SetLocalDescriptionObserverInterface is added. - SetLocalDescription is overloaded. - The adapter for SetSessionDescriptionObserver that causes the delay previously only used for SRD is updated to handle both SLD and SRD. - FakeSetLocalDescriptionObserver is added and MockSetRemoteDescriptionObserver is renamed "Fake...". Bug: chromium:1071733 Change-Id: I920368e648bede481058ac22f5b8794752a220b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31798}
311 lines
6.2 KiB
Python
311 lines
6.2 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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"-ios",
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"-infra",
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"-logging",
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"-media",
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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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".*\.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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"+rtc_base/deprecation.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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"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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"jsep_ice_candidate\.h": [
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"+rtc_base/constructor_magic.h",
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],
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"jsep_session_description\.h": [
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"+rtc_base/constructor_magic.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_factory_proxy\.h": [
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"+rtc_base/bind.h",
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],
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"peer_connection_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_allocator.h",
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"+rtc_base/network.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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],
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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/constructor_magic.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_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/constructor_magic.h",
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"+rtc_base/ref_count.h",
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"+rtc_base/string_encode.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_frame\.h": [
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"+rtc_base/constructor_magic.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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"+rtc_base/constructor_magic.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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"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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# .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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