diff --git a/third_party/libwebrtc/README.moz-ff-commit b/third_party/libwebrtc/README.moz-ff-commit index d4ba83074dc5..5941c26d23c7 100644 --- a/third_party/libwebrtc/README.moz-ff-commit +++ b/third_party/libwebrtc/README.moz-ff-commit @@ -11730,3 +11730,6 @@ b50cfc9fbb1 # MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh # base of lastest vendoring 3be0c39f910 +# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh +# base of lastest vendoring +c24a2189d72 diff --git a/third_party/libwebrtc/README.mozilla b/third_party/libwebrtc/README.mozilla index 212fed552ba5..4efc1221a88e 100644 --- a/third_party/libwebrtc/README.mozilla +++ b/third_party/libwebrtc/README.mozilla @@ -7830,3 +7830,5 @@ libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwe libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-11T21:06:27.209238. # python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-11T21:07:06.078426. +# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc +libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-11T21:07:44.834001. diff --git a/third_party/libwebrtc/pc/BUILD.gn b/third_party/libwebrtc/pc/BUILD.gn index 1a296c1658b5..c088e2cdaf17 100644 --- a/third_party/libwebrtc/pc/BUILD.gn +++ b/third_party/libwebrtc/pc/BUILD.gn @@ -142,13 +142,16 @@ rtc_library("rtc_pc_base") { "../api:rtp_transceiver_direction", "../api:scoped_refptr", "../api:sequence_checker", + "../api:video_track_source_constraints", "../api/crypto:options", "../api/rtc_event_log", "../api/task_queue", "../api/transport:datagram_transport_interface", "../api/transport:enums", "../api/transport:sctp_transport_factory_interface", + "../api/units:timestamp", "../api/video:builtin_video_bitrate_allocator_factory", + "../api/video:recordable_encoded_frame", "../api/video:video_bitrate_allocator_factory", "../api/video:video_frame", "../api/video:video_rtp_headers", @@ -156,6 +159,7 @@ rtc_library("rtc_pc_base") { "../call:call_interfaces", "../call:rtp_interfaces", "../call:rtp_receiver", + "../call:video_stream_api", "../common_video", "../common_video:common_video", "../logging:ice_log", @@ -174,6 +178,8 @@ rtc_library("rtc_pc_base") { "../rtc_base:socket_address", "../rtc_base:stringutils", "../rtc_base:threading", + "../rtc_base/containers:flat_map", + "../rtc_base/containers:flat_set", "../rtc_base/network:sent_packet", "../rtc_base/synchronization:mutex", "../rtc_base/system:file_wrapper", @@ -219,6 +225,7 @@ rtc_source_set("session_description") { "../p2p:rtc_p2p", "../rtc_base:checks", "../rtc_base:socket_address", + "../rtc_base:stringutils", "../rtc_base/system:rtc_export", ] absl_deps = [ @@ -471,6 +478,8 @@ rtc_library("connection_context") { "../p2p:rtc_p2p", "../rtc_base", "../rtc_base:checks", + "../rtc_base:socket_factory", + "../rtc_base:socket_server", "../rtc_base:threading", "../rtc_base/task_utils:to_queued_task", ] @@ -547,6 +556,7 @@ rtc_source_set("rtc_stats_collector") { "../api:scoped_refptr", "../api:sequence_checker", "../api/task_queue:task_queue", + "../api/units:time_delta", "../api/video:video_rtp_headers", "../call:call_interfaces", "../common_video:common_video", @@ -568,7 +578,10 @@ rtc_source_set("rtc_stats_collector") { "../rtc_base:timeutils", "../rtc_base/third_party/sigslot:sigslot", ] - absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] + absl_deps = [ + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] } rtc_source_set("rtc_stats_traversal") { @@ -818,7 +831,10 @@ rtc_source_set("stats_collector") { "../rtc_base:timeutils", "../system_wrappers:field_trial", ] - absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] + absl_deps = [ + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] } rtc_source_set("stream_collection") { visibility = [ ":*" ] @@ -879,6 +895,7 @@ rtc_source_set("webrtc_sdp") { ] absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container", + "//third_party/abseil-cpp/absl/strings", "//third_party/abseil-cpp/absl/types:optional", ] } @@ -965,6 +982,7 @@ rtc_source_set("peer_connection_factory") { "../api:rtp_parameters", "../api:scoped_refptr", "../api:sequence_checker", + "../api/metronome", "../api/neteq:neteq_api", "../api/rtc_event_log:rtc_event_log", "../api/task_queue:task_queue", @@ -1057,6 +1075,7 @@ rtc_library("rtp_transceiver") { "../rtc_base:logging", "../rtc_base:macromagic", "../rtc_base:refcount", + "../rtc_base:rtc_base_approved", "../rtc_base:threading", "../rtc_base/task_utils:pending_task_safety_flag", "../rtc_base/task_utils:to_queued_task", @@ -1286,6 +1305,7 @@ rtc_library("video_track") { "../rtc_base:threading", "../rtc_base/system:no_unique_address", ] + absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] } rtc_source_set("sdp_state_provider") { @@ -1306,6 +1326,7 @@ rtc_library("jitter_buffer_delay") { deps = [ "../api:sequence_checker", "../rtc_base:checks", + "../rtc_base:macromagic", "../rtc_base:safe_conversions", "../rtc_base:safe_minmax", "../rtc_base/system:no_unique_address", @@ -1517,6 +1538,7 @@ if (rtc_include_tests && !build_with_chromium) { deps = [ ":audio_rtp_receiver", ":libjingle_peerconnection", + ":media_protocol_names", ":pc_test_utils", ":peerconnection", ":rtc_pc", @@ -1530,7 +1552,12 @@ if (rtc_include_tests && !build_with_chromium) { "../api:rtc_error", "../api:rtp_headers", "../api:rtp_parameters", + "../api:scoped_refptr", + "../api/task_queue:task_queue", + "../api/transport:datagram_transport_interface", + "../api/transport:enums", "../api/video:builtin_video_bitrate_allocator_factory", + "../api/video:recordable_encoded_frame", "../api/video/test:mock_recordable_encoded_frame", "../call:rtp_interfaces", "../call:rtp_receiver", @@ -1547,7 +1574,9 @@ if (rtc_include_tests && !build_with_chromium) { "../rtc_base:gunit_helpers", "../rtc_base:rtc_base_approved", "../rtc_base:rtc_base_tests_utils", + "../rtc_base:socket_address", "../rtc_base:threading", + "../rtc_base/containers:flat_set", "../rtc_base/task_utils:pending_task_safety_flag", "../rtc_base/task_utils:to_queued_task", "../rtc_base/third_party/sigslot", @@ -1559,6 +1588,7 @@ if (rtc_include_tests && !build_with_chromium) { "//third_party/abseil-cpp/absl/algorithm:container", "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", ] if (rtc_build_libsrtp) { @@ -1582,6 +1612,7 @@ if (rtc_include_tests && !build_with_chromium) { "../api:create_peerconnection_factory", "../api:libjingle_peerconnection_api", "../api:media_stream_interface", + "../api:rtc_error", "../api:rtc_stats_api", "../api:scoped_refptr", "../api/audio:audio_mixer_api", @@ -1601,6 +1632,7 @@ if (rtc_include_tests && !build_with_chromium) { "../rtc_base:gunit_helpers", "../rtc_base:rtc_base_tests_utils", "../rtc_base:socket_address", + "../rtc_base:socket_factory", "../rtc_base:threading", "../system_wrappers", "../test:perf_test", @@ -1690,6 +1722,7 @@ if (rtc_include_tests && !build_with_chromium) { ":integration_test_helpers", ":jitter_buffer_delay", ":local_audio_source", + ":media_protocol_names", ":media_stream", ":peer_connection", ":peer_connection_factory", @@ -1708,6 +1741,7 @@ if (rtc_include_tests && !build_with_chromium) { ":sdp_serializer", ":sdp_utils", ":session_description", + ":simulcast_description", ":stats_collector", ":stream_collection", ":track_media_info_map", @@ -1729,9 +1763,11 @@ if (rtc_include_tests && !build_with_chromium) { "../api:mock_rtp", "../api:mock_video_track", "../api:packet_socket_factory", + "../api:priority", "../api:rtc_error", "../api:rtp_transceiver_direction", "../api:scoped_refptr", + "../api/adaptation:resource_adaptation_api", "../api/audio:audio_mixer_api", "../api/crypto:frame_decryptor_interface", "../api/crypto:frame_encryptor_interface", @@ -1740,13 +1776,22 @@ if (rtc_include_tests && !build_with_chromium) { "../api/rtc_event_log:rtc_event_log_factory", "../api/task_queue", "../api/task_queue:default_task_queue_factory", + "../api/transport:datagram_transport_interface", "../api/transport:field_trial_based_config", + "../api/transport:sctp_transport_factory_interface", "../api/transport:webrtc_key_value_config", "../api/transport/rtp:rtp_source", "../api/units:time_delta", + "../api/units:timestamp", "../api/video:builtin_video_bitrate_allocator_factory", + "../api/video:encoded_image", + "../api/video:recordable_encoded_frame", + "../api/video:video_bitrate_allocator_factory", + "../api/video:video_codec_constants", + "../api/video:video_frame", "../api/video:video_rtp_headers", "../call/adaptation:resource_adaptation_test_utilities", + "../common_video", "../logging:fake_rtc_event_log", "../media:rtc_data_sctp_transport_internal", "../media:rtc_media_config", @@ -1761,9 +1806,11 @@ if (rtc_include_tests && !build_with_chromium) { "../rtc_base:checks", "../rtc_base:gunit_helpers", "../rtc_base:ip_address", + "../rtc_base:network_constants", "../rtc_base:rtc_base_tests_utils", "../rtc_base:rtc_json", "../rtc_base:socket_address", + "../rtc_base:socket_factory", "../rtc_base:threading", "../rtc_base/synchronization:mutex", "../rtc_base/third_party/base64", diff --git a/third_party/libwebrtc/pc/audio_rtp_receiver.cc b/third_party/libwebrtc/pc/audio_rtp_receiver.cc index 43294c7e93f9..3a306720c785 100644 --- a/third_party/libwebrtc/pc/audio_rtp_receiver.cc +++ b/third_party/libwebrtc/pc/audio_rtp_receiver.cc @@ -12,6 +12,7 @@ #include +#include #include #include @@ -20,7 +21,7 @@ #include "pc/media_stream_track_proxy.h" #include "rtc_base/checks.h" #include "rtc_base/location.h" -#include "rtc_base/logging.h" +#include "rtc_base/ref_counted_object.h" #include "rtc_base/task_utils/to_queued_task.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc b/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc index 294e5805252c..ac843fe9c2ed 100644 --- a/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc +++ b/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc @@ -10,9 +10,11 @@ #include "pc/audio_rtp_receiver.h" -#include "media/base/media_channel.h" +#include + #include "pc/test/mock_voice_media_channel.h" #include "rtc_base/gunit.h" +#include "rtc_base/ref_counted_object.h" #include "rtc_base/thread.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/channel.cc b/third_party/libwebrtc/pc/channel.cc index b322ae2e7310..629f4ff36dcb 100644 --- a/third_party/libwebrtc/pc/channel.cc +++ b/third_party/libwebrtc/pc/channel.cc @@ -12,26 +12,26 @@ #include #include -#include -#include +#include +#include #include -#include "absl/algorithm/container.h" #include "absl/strings/string_view.h" #include "api/rtp_parameters.h" #include "api/sequence_checker.h" -#include "api/task_queue/queued_task.h" +#include "api/units/timestamp.h" #include "media/base/codec.h" #include "media/base/rid_description.h" #include "media/base/rtp_utils.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "p2p/base/dtls_transport_internal.h" #include "pc/rtp_media_utils.h" #include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/location.h" #include "rtc_base/logging.h" #include "rtc_base/network_route.h" #include "rtc_base/strings/string_format.h" -#include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_utils/pending_task_safety_flag.h" #include "rtc_base/task_utils/to_queued_task.h" #include "rtc_base/trace_event.h" diff --git a/third_party/libwebrtc/pc/channel.h b/third_party/libwebrtc/pc/channel.h index 930ca9bdfd12..018cb43ba30a 100644 --- a/third_party/libwebrtc/pc/channel.h +++ b/third_party/libwebrtc/pc/channel.h @@ -14,6 +14,7 @@ #include #include +#include #include #include #include @@ -21,12 +22,14 @@ #include #include +#include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/call/audio_sink.h" #include "api/crypto/crypto_options.h" #include "api/function_view.h" #include "api/jsep.h" #include "api/media_types.h" +#include "api/rtp_parameters.h" #include "api/rtp_receiver_interface.h" #include "api/rtp_transceiver_direction.h" #include "api/scoped_refptr.h" @@ -52,6 +55,7 @@ #include "rtc_base/async_packet_socket.h" #include "rtc_base/async_udp_socket.h" #include "rtc_base/checks.h" +#include "rtc_base/containers/flat_set.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/location.h" #include "rtc_base/network.h" diff --git a/third_party/libwebrtc/pc/channel_manager.cc b/third_party/libwebrtc/pc/channel_manager.cc index 2146ed50877d..1482d7f862f4 100644 --- a/third_party/libwebrtc/pc/channel_manager.cc +++ b/third_party/libwebrtc/pc/channel_manager.cc @@ -10,17 +10,16 @@ #include "pc/channel_manager.h" -#include #include #include "absl/algorithm/container.h" #include "absl/memory/memory.h" #include "absl/strings/match.h" +#include "api/media_types.h" #include "api/sequence_checker.h" #include "media/base/media_constants.h" #include "rtc_base/checks.h" #include "rtc_base/location.h" -#include "rtc_base/logging.h" #include "rtc_base/trace_event.h" namespace cricket { diff --git a/third_party/libwebrtc/pc/channel_manager.h b/third_party/libwebrtc/pc/channel_manager.h index a1c4efd55b2c..72bd1328b343 100644 --- a/third_party/libwebrtc/pc/channel_manager.h +++ b/third_party/libwebrtc/pc/channel_manager.h @@ -27,10 +27,12 @@ #include "media/base/media_config.h" #include "media/base/media_engine.h" #include "pc/channel.h" +#include "pc/channel_interface.h" #include "pc/rtp_transport_internal.h" #include "pc/session_description.h" #include "rtc_base/system/file_wrapper.h" #include "rtc_base/thread.h" +#include "rtc_base/thread_annotations.h" #include "rtc_base/unique_id_generator.h" namespace cricket { diff --git a/third_party/libwebrtc/pc/channel_manager_unittest.cc b/third_party/libwebrtc/pc/channel_manager_unittest.cc index 765e8e144da5..9503243a09b2 100644 --- a/third_party/libwebrtc/pc/channel_manager_unittest.cc +++ b/third_party/libwebrtc/pc/channel_manager_unittest.cc @@ -10,19 +10,18 @@ #include "pc/channel_manager.h" -#include - -#include "api/rtc_error.h" +#include "api/sequence_checker.h" #include "api/video/builtin_video_bitrate_allocator_factory.h" #include "media/base/fake_media_engine.h" #include "media/base/test_utils.h" #include "media/engine/fake_webrtc_call.h" -#include "p2p/base/dtls_transport_internal.h" #include "p2p/base/fake_dtls_transport.h" #include "p2p/base/p2p_constants.h" -#include "p2p/base/packet_transport_internal.h" #include "pc/dtls_srtp_transport.h" +#include "pc/rtp_transport_internal.h" +#include "rtc_base/arraysize.h" #include "rtc_base/checks.h" +#include "rtc_base/location.h" #include "rtc_base/thread.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/channel_unittest.cc b/third_party/libwebrtc/pc/channel_unittest.cc index 304c3b28b4e8..3bf955640347 100644 --- a/third_party/libwebrtc/pc/channel_unittest.cc +++ b/third_party/libwebrtc/pc/channel_unittest.cc @@ -10,9 +10,11 @@ #include "pc/channel.h" +#include + #include -#include -#include +#include +#include #include "api/array_view.h" #include "api/audio_options.h" @@ -21,11 +23,15 @@ #include "media/base/fake_media_engine.h" #include "media/base/fake_rtp.h" #include "media/base/media_channel.h" +#include "media/base/media_constants.h" +#include "media/base/rid_description.h" #include "p2p/base/candidate_pair_interface.h" +#include "p2p/base/dtls_transport_internal.h" #include "p2p/base/fake_dtls_transport.h" #include "p2p/base/fake_packet_transport.h" #include "p2p/base/ice_transport_internal.h" #include "p2p/base/p2p_constants.h" +#include "p2p/base/packet_transport_internal.h" #include "pc/dtls_srtp_transport.h" #include "pc/jsep_transport.h" #include "pc/rtp_transport.h" @@ -33,6 +39,7 @@ #include "rtc_base/buffer.h" #include "rtc_base/byte_order.h" #include "rtc_base/checks.h" +#include "rtc_base/location.h" #include "rtc_base/rtc_certificate.h" #include "rtc_base/ssl_identity.h" #include "rtc_base/task_utils/pending_task_safety_flag.h" diff --git a/third_party/libwebrtc/pc/connection_context.cc b/third_party/libwebrtc/pc/connection_context.cc index 6e531339f4cc..d093ee3cf5d6 100644 --- a/third_party/libwebrtc/pc/connection_context.cc +++ b/third_party/libwebrtc/pc/connection_context.cc @@ -10,14 +10,15 @@ #include "pc/connection_context.h" -#include #include #include #include "api/transport/field_trial_based_config.h" +#include "media/base/media_engine.h" #include "media/sctp/sctp_transport_factory.h" #include "rtc_base/helpers.h" #include "rtc_base/internal/default_socket_server.h" +#include "rtc_base/socket_server.h" #include "rtc_base/task_utils/to_queued_task.h" #include "rtc_base/time_utils.h" diff --git a/third_party/libwebrtc/pc/connection_context.h b/third_party/libwebrtc/pc/connection_context.h index 5e814079f9a1..2aaa840df15a 100644 --- a/third_party/libwebrtc/pc/connection_context.h +++ b/third_party/libwebrtc/pc/connection_context.h @@ -29,6 +29,7 @@ #include "rtc_base/network.h" #include "rtc_base/network_monitor_factory.h" #include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/socket_factory.h" #include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" diff --git a/third_party/libwebrtc/pc/data_channel_controller.cc b/third_party/libwebrtc/pc/data_channel_controller.cc index adbf303105e1..832eb03f7981 100644 --- a/third_party/libwebrtc/pc/data_channel_controller.cc +++ b/third_party/libwebrtc/pc/data_channel_controller.cc @@ -10,11 +10,8 @@ #include "pc/data_channel_controller.h" -#include #include -#include "absl/algorithm/container.h" -#include "absl/types/optional.h" #include "api/peer_connection_interface.h" #include "api/rtc_error.h" #include "pc/peer_connection_internal.h" diff --git a/third_party/libwebrtc/pc/data_channel_controller.h b/third_party/libwebrtc/pc/data_channel_controller.h index 00d38f0c8446..fa10b745c688 100644 --- a/third_party/libwebrtc/pc/data_channel_controller.h +++ b/third_party/libwebrtc/pc/data_channel_controller.h @@ -19,6 +19,7 @@ #include #include "api/data_channel_interface.h" +#include "api/rtc_error.h" #include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/transport/data_channel_transport_interface.h" diff --git a/third_party/libwebrtc/pc/data_channel_integrationtest.cc b/third_party/libwebrtc/pc/data_channel_integrationtest.cc index c0dbfdd4bf4b..c76a4d97b041 100644 --- a/third_party/libwebrtc/pc/data_channel_integrationtest.cc +++ b/third_party/libwebrtc/pc/data_channel_integrationtest.cc @@ -10,24 +10,37 @@ #include -#include -#include +#include +#include #include +#include #include +#include "absl/algorithm/container.h" #include "absl/types/optional.h" #include "api/data_channel_interface.h" -#include "api/dtmf_sender_interface.h" +#include "api/dtls_transport_interface.h" #include "api/peer_connection_interface.h" #include "api/scoped_refptr.h" +#include "api/sctp_transport_interface.h" +#include "api/stats/rtc_stats_report.h" +#include "api/stats/rtcstats_objects.h" #include "api/units/time_delta.h" +#include "p2p/base/transport_description.h" +#include "p2p/base/transport_info.h" +#include "pc/media_session.h" +#include "pc/session_description.h" #include "pc/test/integration_test_helpers.h" #include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/fake_clock.h" #include "rtc_base/gunit.h" -#include "rtc_base/ref_counted_object.h" +#include "rtc_base/helpers.h" +#include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/virtual_socket_server.h" #include "system_wrappers/include/field_trial.h" +#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/data_channel_unittest.cc b/third_party/libwebrtc/pc/data_channel_unittest.cc index 44c080bbda42..5797d1da4407 100644 --- a/third_party/libwebrtc/pc/data_channel_unittest.cc +++ b/third_party/libwebrtc/pc/data_channel_unittest.cc @@ -8,17 +8,27 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include #include #include +#include #include +#include "api/data_channel_interface.h" +#include "api/rtc_error.h" +#include "api/scoped_refptr.h" +#include "api/transport/data_channel_transport_interface.h" +#include "media/base/media_channel.h" #include "media/sctp/sctp_transport_internal.h" #include "pc/sctp_data_channel.h" #include "pc/sctp_utils.h" #include "pc/test/fake_data_channel_provider.h" +#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/gunit.h" -#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/ssl_stream_adapter.h" +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "rtc_base/thread.h" #include "test/gtest.h" using webrtc::DataChannelInterface; diff --git a/third_party/libwebrtc/pc/dtls_srtp_transport.h b/third_party/libwebrtc/pc/dtls_srtp_transport.h index da068c9b8ae7..c2c51c22f2c0 100644 --- a/third_party/libwebrtc/pc/dtls_srtp_transport.h +++ b/third_party/libwebrtc/pc/dtls_srtp_transport.h @@ -11,6 +11,7 @@ #ifndef PC_DTLS_SRTP_TRANSPORT_H_ #define PC_DTLS_SRTP_TRANSPORT_H_ +#include #include #include diff --git a/third_party/libwebrtc/pc/dtls_srtp_transport_unittest.cc b/third_party/libwebrtc/pc/dtls_srtp_transport_unittest.cc index 72df81a923d3..76d9c30c5efd 100644 --- a/third_party/libwebrtc/pc/dtls_srtp_transport_unittest.cc +++ b/third_party/libwebrtc/pc/dtls_srtp_transport_unittest.cc @@ -14,7 +14,6 @@ #include #include -#include #include "call/rtp_demuxer.h" #include "media/base/fake_rtp.h" @@ -26,9 +25,11 @@ #include "pc/test/rtp_transport_test_util.h" #include "rtc_base/async_packet_socket.h" #include "rtc_base/byte_order.h" +#include "rtc_base/containers/flat_set.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/rtc_certificate.h" #include "rtc_base/ssl_identity.h" +#include "rtc_base/third_party/sigslot/sigslot.h" #include "test/gtest.h" using cricket::FakeDtlsTransport; diff --git a/third_party/libwebrtc/pc/dtls_transport.cc b/third_party/libwebrtc/pc/dtls_transport.cc index e8d6ae9b6a1c..c9f3279fbc9c 100644 --- a/third_party/libwebrtc/pc/dtls_transport.cc +++ b/third_party/libwebrtc/pc/dtls_transport.cc @@ -19,7 +19,7 @@ #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/ref_counted_object.h" -#include "rtc_base/ssl_certificate.h" +#include "rtc_base/ssl_stream_adapter.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/dtls_transport_unittest.cc b/third_party/libwebrtc/pc/dtls_transport_unittest.cc index 1400ff94c21b..a9ac73c9b8b4 100644 --- a/third_party/libwebrtc/pc/dtls_transport_unittest.cc +++ b/third_party/libwebrtc/pc/dtls_transport_unittest.cc @@ -13,9 +13,15 @@ #include #include -#include "absl/memory/memory.h" +#include "absl/types/optional.h" +#include "api/rtc_error.h" #include "p2p/base/fake_dtls_transport.h" +#include "p2p/base/p2p_constants.h" +#include "rtc_base/fake_ssl_identity.h" #include "rtc_base/gunit.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/rtc_certificate.h" +#include "rtc_base/ssl_identity.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/dtmf_sender.cc b/third_party/libwebrtc/pc/dtmf_sender.cc index 1148350aa2ff..8b82c31aa911 100644 --- a/third_party/libwebrtc/pc/dtmf_sender.cc +++ b/third_party/libwebrtc/pc/dtmf_sender.cc @@ -13,8 +13,6 @@ #include #include -#include - #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/ref_counted_object.h" diff --git a/third_party/libwebrtc/pc/dtmf_sender.h b/third_party/libwebrtc/pc/dtmf_sender.h index 915d9874b3f9..ae213b3bf48c 100644 --- a/third_party/libwebrtc/pc/dtmf_sender.h +++ b/third_party/libwebrtc/pc/dtmf_sender.h @@ -17,12 +17,14 @@ #include "api/dtmf_sender_interface.h" #include "api/scoped_refptr.h" +#include "api/sequence_checker.h" #include "pc/proxy.h" #include "rtc_base/location.h" #include "rtc_base/ref_count.h" #include "rtc_base/task_utils/pending_task_safety_flag.h" #include "rtc_base/third_party/sigslot/sigslot.h" #include "rtc_base/thread.h" +#include "rtc_base/thread_annotations.h" // DtmfSender is the native implementation of the RTCDTMFSender defined by // the WebRTC W3C Editor's Draft. @@ -102,6 +104,7 @@ class DtmfSender : public DtmfSenderInterface, public sigslot::has_slots<> { // Define proxy for DtmfSenderInterface. BEGIN_PRIMARY_PROXY_MAP(DtmfSender) + PROXY_PRIMARY_THREAD_DESTRUCTOR() PROXY_METHOD1(void, RegisterObserver, DtmfSenderObserverInterface*) PROXY_METHOD0(void, UnregisterObserver) diff --git a/third_party/libwebrtc/pc/external_hmac.h b/third_party/libwebrtc/pc/external_hmac.h index 3319beaed4d1..3c2936c685e2 100644 --- a/third_party/libwebrtc/pc/external_hmac.h +++ b/third_party/libwebrtc/pc/external_hmac.h @@ -33,6 +33,7 @@ #include "third_party/libsrtp/crypto/include/auth.h" #include "third_party/libsrtp/crypto/include/crypto_types.h" #include "third_party/libsrtp/include/srtp.h" +#include "third_party/libsrtp/include/srtp_priv.h" #define EXTERNAL_HMAC_SHA1 SRTP_HMAC_SHA1 + 1 #define HMAC_KEY_LENGTH 20 diff --git a/third_party/libwebrtc/pc/ice_server_parsing.cc b/third_party/libwebrtc/pc/ice_server_parsing.cc index 88f77bf0a9b6..cb4145be1af8 100644 --- a/third_party/libwebrtc/pc/ice_server_parsing.cc +++ b/third_party/libwebrtc/pc/ice_server_parsing.cc @@ -12,9 +12,7 @@ #include -#include #include // For std::isdigit. -#include #include #include "p2p/base/port_interface.h" diff --git a/third_party/libwebrtc/pc/ice_transport_unittest.cc b/third_party/libwebrtc/pc/ice_transport_unittest.cc index ebb46cb5d53d..95af2cd552e3 100644 --- a/third_party/libwebrtc/pc/ice_transport_unittest.cc +++ b/third_party/libwebrtc/pc/ice_transport_unittest.cc @@ -12,13 +12,12 @@ #include #include -#include #include "api/ice_transport_factory.h" +#include "api/scoped_refptr.h" #include "p2p/base/fake_ice_transport.h" #include "p2p/base/fake_port_allocator.h" -#include "rtc_base/gunit.h" -#include "test/gmock.h" +#include "rtc_base/ref_counted_object.h" #include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/jitter_buffer_delay.h b/third_party/libwebrtc/pc/jitter_buffer_delay.h index dc10e3d2ba27..a6bec01ce764 100644 --- a/third_party/libwebrtc/pc/jitter_buffer_delay.h +++ b/third_party/libwebrtc/pc/jitter_buffer_delay.h @@ -16,6 +16,7 @@ #include "absl/types/optional.h" #include "api/sequence_checker.h" #include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/jitter_buffer_delay_unittest.cc b/third_party/libwebrtc/pc/jitter_buffer_delay_unittest.cc index b00075ceb56e..79c39fffb828 100644 --- a/third_party/libwebrtc/pc/jitter_buffer_delay_unittest.cc +++ b/third_party/libwebrtc/pc/jitter_buffer_delay_unittest.cc @@ -10,9 +10,6 @@ #include "pc/jitter_buffer_delay.h" -#include - -#include "absl/types/optional.h" #include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/jsep_ice_candidate.cc b/third_party/libwebrtc/pc/jsep_ice_candidate.cc index 6dacde629c7d..1e97ad42d84a 100644 --- a/third_party/libwebrtc/pc/jsep_ice_candidate.cc +++ b/third_party/libwebrtc/pc/jsep_ice_candidate.cc @@ -10,8 +10,6 @@ #include "api/jsep_ice_candidate.h" -#include - #include "pc/webrtc_sdp.h" // This file contains JsepIceCandidate-related functions that are not diff --git a/third_party/libwebrtc/pc/jsep_session_description.cc b/third_party/libwebrtc/pc/jsep_session_description.cc index 57ccf7ca6e7e..4c57396f08b9 100644 --- a/third_party/libwebrtc/pc/jsep_session_description.cc +++ b/third_party/libwebrtc/pc/jsep_session_description.cc @@ -11,11 +11,20 @@ #include "api/jsep_session_description.h" #include +#include +#include "absl/types/optional.h" +#include "p2p/base/p2p_constants.h" #include "p2p/base/port.h" -#include "pc/media_session.h" +#include "p2p/base/transport_description.h" +#include "p2p/base/transport_info.h" +#include "pc/media_session.h" // IWYU pragma: keep #include "pc/webrtc_sdp.h" -#include "rtc_base/arraysize.h" +#include "rtc_base/checks.h" +#include "rtc_base/ip_address.h" +#include "rtc_base/logging.h" +#include "rtc_base/net_helper.h" +#include "rtc_base/socket_address.h" using cricket::SessionDescription; diff --git a/third_party/libwebrtc/pc/jsep_session_description_unittest.cc b/third_party/libwebrtc/pc/jsep_session_description_unittest.cc index 2202aa81d0b4..ee446cbbdad2 100644 --- a/third_party/libwebrtc/pc/jsep_session_description_unittest.cc +++ b/third_party/libwebrtc/pc/jsep_session_description_unittest.cc @@ -13,8 +13,6 @@ #include #include -#include -#include #include #include @@ -29,6 +27,7 @@ #include "pc/session_description.h" #include "pc/webrtc_sdp.h" #include "rtc_base/helpers.h" +#include "rtc_base/net_helper.h" #include "rtc_base/socket_address.h" #include "rtc_base/string_encode.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/jsep_transport.cc b/third_party/libwebrtc/pc/jsep_transport.cc index 706c342a8d1e..00447b088d00 100644 --- a/third_party/libwebrtc/pc/jsep_transport.cc +++ b/third_party/libwebrtc/pc/jsep_transport.cc @@ -15,6 +15,7 @@ #include #include +#include #include #include "api/array_view.h" diff --git a/third_party/libwebrtc/pc/jsep_transport_collection.h b/third_party/libwebrtc/pc/jsep_transport_collection.h index aa5293475e88..099e24a457b6 100644 --- a/third_party/libwebrtc/pc/jsep_transport_collection.h +++ b/third_party/libwebrtc/pc/jsep_transport_collection.h @@ -18,6 +18,7 @@ #include #include +#include "api/jsep.h" #include "api/peer_connection_interface.h" #include "api/sequence_checker.h" #include "pc/jsep_transport.h" diff --git a/third_party/libwebrtc/pc/jsep_transport_controller.cc b/third_party/libwebrtc/pc/jsep_transport_controller.cc index b7e9f361bc8e..e63742aaee23 100644 --- a/third_party/libwebrtc/pc/jsep_transport_controller.cc +++ b/third_party/libwebrtc/pc/jsep_transport_controller.cc @@ -12,9 +12,9 @@ #include -#include #include #include +#include #include #include diff --git a/third_party/libwebrtc/pc/jsep_transport_controller.h b/third_party/libwebrtc/pc/jsep_transport_controller.h index d207269d0976..ed4d20ba8412 100644 --- a/third_party/libwebrtc/pc/jsep_transport_controller.h +++ b/third_party/libwebrtc/pc/jsep_transport_controller.h @@ -17,6 +17,7 @@ #include #include #include +#include #include #include diff --git a/third_party/libwebrtc/pc/jsep_transport_controller_unittest.cc b/third_party/libwebrtc/pc/jsep_transport_controller_unittest.cc index 52d818217951..622a9b90e3df 100644 --- a/third_party/libwebrtc/pc/jsep_transport_controller_unittest.cc +++ b/third_party/libwebrtc/pc/jsep_transport_controller_unittest.cc @@ -11,14 +11,26 @@ #include "pc/jsep_transport_controller.h" #include -#include +#include +#include #include "api/dtls_transport_interface.h" +#include "api/transport/enums.h" +#include "p2p/base/candidate_pair_interface.h" #include "p2p/base/dtls_transport_factory.h" #include "p2p/base/fake_dtls_transport.h" #include "p2p/base/fake_ice_transport.h" +#include "p2p/base/p2p_constants.h" #include "p2p/base/transport_info.h" +#include "rtc_base/fake_ssl_identity.h" #include "rtc_base/gunit.h" +#include "rtc_base/location.h" +#include "rtc_base/logging.h" +#include "rtc_base/net_helper.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/ssl_fingerprint.h" +#include "rtc_base/ssl_identity.h" #include "rtc_base/thread.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/jsep_transport_unittest.cc b/third_party/libwebrtc/pc/jsep_transport_unittest.cc index d06fa9596a7c..2ec96d2af788 100644 --- a/third_party/libwebrtc/pc/jsep_transport_unittest.cc +++ b/third_party/libwebrtc/pc/jsep_transport_unittest.cc @@ -10,15 +10,33 @@ #include "pc/jsep_transport.h" -#include +#include +#include + +#include +#include #include #include -#include "api/ice_transport_factory.h" +#include "api/candidate.h" #include "media/base/fake_rtp.h" #include "p2p/base/fake_dtls_transport.h" #include "p2p/base/fake_ice_transport.h" -#include "rtc_base/gunit.h" +#include "p2p/base/p2p_constants.h" +#include "p2p/base/packet_transport_internal.h" +#include "rtc_base/async_packet_socket.h" +#include "rtc_base/buffer.h" +#include "rtc_base/byte_order.h" +#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/helpers.h" +#include "rtc_base/logging.h" +#include "rtc_base/net_helper.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/ssl_certificate.h" +#include "rtc_base/ssl_identity.h" +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "test/gtest.h" namespace cricket { namespace { diff --git a/third_party/libwebrtc/pc/media_session.cc b/third_party/libwebrtc/pc/media_session.cc index 96c285e67eed..2a3a6988741c 100644 --- a/third_party/libwebrtc/pc/media_session.cc +++ b/third_party/libwebrtc/pc/media_session.cc @@ -14,8 +14,7 @@ #include #include -#include -#include +#include #include #include @@ -24,7 +23,6 @@ #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/crypto_params.h" -#include "api/video_codecs/h264_profile_level_id.h" #include "media/base/codec.h" #include "media/base/media_constants.h" #include "media/base/sdp_video_format_utils.h" diff --git a/third_party/libwebrtc/pc/media_session_unittest.cc b/third_party/libwebrtc/pc/media_session_unittest.cc index cfe9ac8a3109..6bc842d22252 100644 --- a/third_party/libwebrtc/pc/media_session_unittest.cc +++ b/third_party/libwebrtc/pc/media_session_unittest.cc @@ -10,32 +10,42 @@ #include "pc/media_session.h" +#include + #include +#include +#include #include #include -#include +#include #include #include "absl/algorithm/container.h" -#include "absl/memory/memory.h" #include "absl/strings/match.h" +#include "absl/strings/string_view.h" +#include "api/candidate.h" +#include "api/crypto_params.h" #include "media/base/codec.h" +#include "media/base/media_constants.h" #include "media/base/test_utils.h" #include "media/sctp/sctp_transport_internal.h" #include "p2p/base/p2p_constants.h" #include "p2p/base/transport_description.h" #include "p2p/base/transport_info.h" +#include "pc/media_protocol_names.h" #include "pc/rtp_media_utils.h" -#include "pc/srtp_filter.h" +#include "rtc_base/arraysize.h" #include "rtc_base/checks.h" #include "rtc_base/fake_ssl_identity.h" -#include "rtc_base/gunit.h" -#include "rtc_base/message_digest.h" -#include "rtc_base/ssl_adapter.h" +#include "rtc_base/rtc_certificate.h" +#include "rtc_base/ssl_identity.h" +#include "rtc_base/ssl_stream_adapter.h" +#include "rtc_base/string_encode.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/unique_id_generator.h" #include "test/field_trial.h" #include "test/gmock.h" +#include "test/gtest.h" #define ASSERT_CRYPTO(cd, s, cs) \ ASSERT_EQ(s, cd->cryptos().size()); \ diff --git a/third_party/libwebrtc/pc/media_stream.cc b/third_party/libwebrtc/pc/media_stream.cc index 6fe308827cb3..011d8abf5517 100644 --- a/third_party/libwebrtc/pc/media_stream.cc +++ b/third_party/libwebrtc/pc/media_stream.cc @@ -13,7 +13,6 @@ #include #include -#include #include "rtc_base/checks.h" #include "rtc_base/ref_counted_object.h" diff --git a/third_party/libwebrtc/pc/media_stream_unittest.cc b/third_party/libwebrtc/pc/media_stream_unittest.cc index 6ce8de9a1a07..1d6935690e9f 100644 --- a/third_party/libwebrtc/pc/media_stream_unittest.cc +++ b/third_party/libwebrtc/pc/media_stream_unittest.cc @@ -12,8 +12,6 @@ #include -#include - #include "pc/audio_track.h" #include "pc/test/fake_video_track_source.h" #include "pc/video_track.h" diff --git a/third_party/libwebrtc/pc/peer_connection.h b/third_party/libwebrtc/pc/peer_connection.h index 4855d32be1e6..cd4af9e42097 100644 --- a/third_party/libwebrtc/pc/peer_connection.h +++ b/third_party/libwebrtc/pc/peer_connection.h @@ -14,7 +14,6 @@ #include #include -#include #include #include #include diff --git a/third_party/libwebrtc/pc/peer_connection_adaptation_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_adaptation_integrationtest.cc index dfb12971b4e1..b5a5f5231dff 100644 --- a/third_party/libwebrtc/pc/peer_connection_adaptation_integrationtest.cc +++ b/third_party/libwebrtc/pc/peer_connection_adaptation_integrationtest.cc @@ -8,12 +8,22 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include +#include +#include +#include + +#include "absl/types/optional.h" +#include "api/adaptation/resource.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/media_stream_interface.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" #include "api/rtp_parameters.h" +#include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" +#include "api/video/video_source_interface.h" #include "call/adaptation/test/fake_resource.h" #include "pc/test/fake_periodic_video_source.h" #include "pc/test/fake_periodic_video_track_source.h" @@ -22,6 +32,7 @@ #include "rtc_base/gunit.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/thread.h" +#include "rtc_base/time_utils.h" #include "rtc_base/virtual_socket_server.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/peer_connection_bundle_unittest.cc b/third_party/libwebrtc/pc/peer_connection_bundle_unittest.cc index e5eb6c4e18b7..4a5d3733106a 100644 --- a/third_party/libwebrtc/pc/peer_connection_bundle_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_bundle_unittest.cc @@ -8,21 +8,59 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "api/audio/audio_mixer.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/candidate.h" #include "api/create_peerconnection_factory.h" +#include "api/jsep.h" +#include "api/media_types.h" +#include "api/peer_connection_interface.h" +#include "api/rtp_receiver_interface.h" +#include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/stats/rtc_stats.h" +#include "api/stats/rtc_stats_report.h" +#include "api/stats/rtcstats_objects.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" -#include "p2p/base/fake_port_allocator.h" -#include "p2p/base/test_stun_server.h" +#include "media/base/stream_params.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "p2p/base/p2p_constants.h" +#include "p2p/base/port.h" +#include "p2p/base/port_allocator.h" +#include "p2p/base/transport_info.h" #include "p2p/client/basic_port_allocator.h" -#include "pc/media_session.h" +#include "pc/channel.h" #include "pc/peer_connection.h" #include "pc/peer_connection_proxy.h" #include "pc/peer_connection_wrapper.h" +#include "pc/rtp_transceiver.h" +#include "pc/rtp_transport_internal.h" #include "pc/sdp_utils.h" +#include "pc/session_description.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_base/net_helper.h" +#include "rtc_base/network.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/thread.h" +#include "test/gtest.h" #ifdef WEBRTC_ANDROID #include "pc/test/android_test_initializer.h" #endif diff --git a/third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc b/third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc index 394203cb022a..1741b992891d 100644 --- a/third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc @@ -8,17 +8,47 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "absl/types/optional.h" +#include "api/audio/audio_mixer.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/create_peerconnection_factory.h" +#include "api/crypto/crypto_options.h" +#include "api/crypto_params.h" +#include "api/jsep.h" +#include "api/peer_connection_interface.h" +#include "api/scoped_refptr.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" #include "p2p/base/fake_port_allocator.h" +#include "p2p/base/port_allocator.h" +#include "p2p/base/transport_description.h" +#include "p2p/base/transport_info.h" +#include "pc/media_protocol_names.h" #include "pc/media_session.h" #include "pc/peer_connection_wrapper.h" #include "pc/sdp_utils.h" +#include "pc/session_description.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/checks.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/rtc_certificate.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/ssl_fingerprint.h" +#include "rtc_base/thread.h" +#include "test/gtest.h" #ifdef WEBRTC_ANDROID #include "pc/test/android_test_initializer.h" #endif diff --git a/third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc b/third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc index 5a6377b00d02..0fb1c638ebf8 100644 --- a/third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc @@ -20,24 +20,24 @@ #include "api/media_types.h" #include "api/peer_connection_interface.h" #include "api/scoped_refptr.h" +#include "api/sctp_transport_interface.h" #include "api/task_queue/default_task_queue_factory.h" -#include "media/base/codec.h" +#include "api/task_queue/task_queue_factory.h" +#include "api/transport/sctp_transport_factory_interface.h" #include "media/base/fake_media_engine.h" -#include "media/base/media_constants.h" #include "media/base/media_engine.h" -#include "media/sctp/sctp_transport_internal.h" #include "p2p/base/p2p_constants.h" #include "p2p/base/port_allocator.h" #include "pc/media_session.h" #include "pc/peer_connection.h" -#include "pc/peer_connection_factory.h" #include "pc/peer_connection_proxy.h" #include "pc/peer_connection_wrapper.h" +#include "pc/sctp_transport.h" #include "pc/sdp_utils.h" #include "pc/session_description.h" #include "pc/test/mock_peer_connection_observers.h" #include "rtc_base/checks.h" -#include "rtc_base/ref_counted_object.h" +#include "rtc_base/logging.h" #include "rtc_base/rtc_certificate_generator.h" #include "rtc_base/thread.h" #include "test/gmock.h" diff --git a/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc b/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc index 3a6402703fbe..78dcda3202ec 100644 --- a/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc @@ -8,20 +8,46 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include + +#include +#include #include +#include +#include +#include +#include #include "absl/strings/match.h" +#include "absl/types/optional.h" #include "api/audio_codecs/L16/audio_decoder_L16.h" #include "api/audio_codecs/L16/audio_encoder_L16.h" #include "api/audio_codecs/audio_codec_pair_id.h" +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_decoder_factory_template.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/audio_encoder_factory_template.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/opus_audio_decoder_factory.h" #include "api/audio_codecs/opus_audio_encoder_factory.h" +#include "api/audio_options.h" +#include "api/data_channel_interface.h" +#include "api/media_stream_interface.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" +#include "api/scoped_refptr.h" #include "media/sctp/sctp_transport_internal.h" +#include "rtc_base/checks.h" +#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/gunit.h" -#include "rtc_base/logging.h" #include "rtc_base/physical_socket_server.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "rtc_base/thread.h" +#include "test/gmock.h" +#include "test/gtest.h" #ifdef WEBRTC_ANDROID #include "pc/test/android_test_initializer.h" diff --git a/third_party/libwebrtc/pc/peer_connection_factory.cc b/third_party/libwebrtc/pc/peer_connection_factory.cc index db2f468a0149..262ca90cd943 100644 --- a/third_party/libwebrtc/pc/peer_connection_factory.cc +++ b/third_party/libwebrtc/pc/peer_connection_factory.cc @@ -10,7 +10,7 @@ #include "pc/peer_connection_factory.h" -#include +#include #include #include "absl/strings/match.h" diff --git a/third_party/libwebrtc/pc/peer_connection_factory.h b/third_party/libwebrtc/pc/peer_connection_factory.h index c1599f48854b..f09ca66e6e4f 100644 --- a/third_party/libwebrtc/pc/peer_connection_factory.h +++ b/third_party/libwebrtc/pc/peer_connection_factory.h @@ -23,6 +23,7 @@ #include "api/fec_controller.h" #include "api/media_stream_interface.h" #include "api/media_types.h" +#include "api/metronome/metronome.h" #include "api/neteq/neteq_factory.h" #include "api/network_state_predictor.h" #include "api/peer_connection_interface.h" diff --git a/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc b/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc index c12b563e2718..4e97053feaab 100644 --- a/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc @@ -10,16 +10,10 @@ #include "pc/peer_connection_factory.h" -#include - -#include -#include #include #include #include "api/audio/audio_mixer.h" -#include "api/audio_codecs/audio_decoder_factory.h" -#include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/create_peerconnection_factory.h" @@ -28,17 +22,18 @@ #include "api/media_stream_interface.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" -#include "api/video_codecs/video_decoder_factory.h" -#include "api/video_codecs/video_encoder_factory.h" #include "media/base/fake_frame_source.h" #include "modules/audio_device/include/audio_device.h" #include "modules/audio_processing/include/audio_processing.h" #include "p2p/base/fake_port_allocator.h" #include "p2p/base/port.h" +#include "p2p/base/port_allocator.h" #include "p2p/base/port_interface.h" #include "pc/test/fake_audio_capture_module.h" #include "pc/test/fake_video_track_source.h" +#include "rtc_base/rtc_certificate_generator.h" #include "rtc_base/socket_address.h" +#include "rtc_base/time_utils.h" #include "test/gtest.h" #ifdef WEBRTC_ANDROID diff --git a/third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc b/third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc index 8bf6c7ab440d..d7d160b631de 100644 --- a/third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc @@ -9,17 +9,38 @@ */ #include +#include #include +#include +#include +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/call/call_factory_interface.h" +#include "api/jsep.h" +#include "api/media_types.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" #include "api/rtc_event_log/rtc_event_log_factory.h" +#include "api/rtc_event_log/rtc_event_log_factory_interface.h" +#include "api/rtp_parameters.h" +#include "api/rtp_transceiver_direction.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" #include "api/task_queue/default_task_queue_factory.h" +#include "api/task_queue/task_queue_factory.h" #include "media/base/fake_media_engine.h" +#include "media/base/media_engine.h" #include "p2p/base/fake_port_allocator.h" -#include "pc/media_session.h" +#include "p2p/base/port_allocator.h" #include "pc/peer_connection_wrapper.h" -#include "rtc_base/gunit.h" +#include "pc/session_description.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/rtc_certificate_generator.h" #include "rtc_base/strings/string_builder.h" +#include "rtc_base/thread.h" #include "test/gmock.h" +#include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc b/third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc index 27bae854d67c..ae88b652131b 100644 --- a/third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc @@ -15,6 +15,7 @@ #include #include "absl/types/optional.h" +#include "api/async_resolver_factory.h" #include "api/call/call_factory_interface.h" #include "api/jsep.h" #include "api/jsep_session_description.h" @@ -22,7 +23,9 @@ #include "api/rtc_error.h" #include "api/scoped_refptr.h" #include "api/task_queue/default_task_queue_factory.h" +#include "api/task_queue/task_queue_factory.h" #include "media/base/fake_media_engine.h" +#include "media/base/media_engine.h" #include "p2p/base/mock_async_resolver.h" #include "p2p/base/port_allocator.h" #include "p2p/client/basic_port_allocator.h" @@ -39,13 +42,14 @@ #include "rtc_base/fake_mdns_responder.h" #include "rtc_base/fake_network.h" #include "rtc_base/gunit.h" +#include "rtc_base/mdns_responder_interface.h" #include "rtc_base/ref_counted_object.h" -#include "rtc_base/rtc_certificate_generator.h" #include "rtc_base/socket_address.h" #include "rtc_base/thread.h" #include "rtc_base/virtual_socket_server.h" #include "system_wrappers/include/metrics.h" #include "test/gmock.h" +#include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/peer_connection_ice_unittest.cc b/third_party/libwebrtc/pc/peer_connection_ice_unittest.cc index c04ff8e20479..ed64aa24ea55 100644 --- a/third_party/libwebrtc/pc/peer_connection_ice_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_ice_unittest.cc @@ -8,15 +8,52 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "absl/types/optional.h" +#include "api/audio/audio_mixer.h" +#include "api/candidate.h" +#include "api/ice_transport_interface.h" +#include "api/jsep.h" +#include "api/media_types.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" +#include "api/scoped_refptr.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" #include "p2p/base/fake_port_allocator.h" -#include "p2p/base/test_stun_server.h" +#include "p2p/base/ice_transport_internal.h" +#include "p2p/base/p2p_constants.h" +#include "p2p/base/port.h" +#include "p2p/base/port_allocator.h" +#include "p2p/base/transport_description.h" +#include "p2p/base/transport_info.h" #include "p2p/client/basic_port_allocator.h" +#include "pc/channel_interface.h" +#include "pc/dtls_transport.h" #include "pc/media_session.h" #include "pc/peer_connection.h" #include "pc/peer_connection_wrapper.h" +#include "pc/rtp_transceiver.h" #include "pc/sdp_utils.h" +#include "pc/session_description.h" +#include "rtc_base/checks.h" +#include "rtc_base/ip_address.h" +#include "rtc_base/logging.h" +#include "rtc_base/net_helper.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/thread.h" +#include "test/gtest.h" #ifdef WEBRTC_ANDROID #include "pc/test/android_test_initializer.h" #endif diff --git a/third_party/libwebrtc/pc/peer_connection_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_integrationtest.cc index 0c97d0da956b..1ee85c3be6e8 100644 --- a/third_party/libwebrtc/pc/peer_connection_integrationtest.cc +++ b/third_party/libwebrtc/pc/peer_connection_integrationtest.cc @@ -14,6 +14,7 @@ #include #include #include +#include #include #include @@ -86,6 +87,8 @@ #include "rtc_base/time_utils.h" #include "rtc_base/virtual_socket_server.h" #include "system_wrappers/include/metrics.h" +#include "test/gmock.h" +#include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc b/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc index 0d9f4191499e..7dd3b31a0c33 100644 --- a/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc @@ -12,9 +12,7 @@ #include #include -#include -#include #include #include #include @@ -22,32 +20,26 @@ #include "absl/strings/str_replace.h" #include "absl/types/optional.h" #include "api/audio/audio_mixer.h" -#include "api/audio_codecs/audio_decoder_factory.h" -#include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/call/call_factory_interface.h" #include "api/create_peerconnection_factory.h" #include "api/data_channel_interface.h" #include "api/jsep.h" -#include "api/jsep_session_description.h" #include "api/media_stream_interface.h" #include "api/media_types.h" #include "api/rtc_error.h" #include "api/rtc_event_log/rtc_event_log.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/rtc_event_log_output.h" -#include "api/rtc_event_log_output_file.h" #include "api/rtp_receiver_interface.h" #include "api/rtp_sender_interface.h" -#include "api/rtp_transceiver_interface.h" +#include "api/rtp_transceiver_direction.h" #include "api/scoped_refptr.h" #include "api/task_queue/default_task_queue_factory.h" #include "api/transport/field_trial_based_config.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" -#include "api/video_codecs/video_decoder_factory.h" -#include "api/video_codecs/video_encoder_factory.h" #include "media/base/codec.h" #include "media/base/media_config.h" #include "media/base/media_engine.h" @@ -68,8 +60,8 @@ #include "pc/media_stream.h" #include "pc/peer_connection.h" #include "pc/peer_connection_factory.h" -#include "pc/rtc_stats_collector.h" #include "pc/rtp_sender.h" +#include "pc/rtp_sender_proxy.h" #include "pc/session_description.h" #include "pc/stream_collection.h" #include "pc/test/fake_audio_capture_module.h" @@ -79,17 +71,14 @@ #include "pc/test/test_sdp_strings.h" #include "pc/video_track.h" #include "rtc_base/checks.h" -#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/gunit.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/rtc_certificate_generator.h" #include "rtc_base/socket_address.h" #include "rtc_base/thread.h" -#include "rtc_base/time_utils.h" #include "rtc_base/virtual_socket_server.h" #include "test/gmock.h" #include "test/gtest.h" -#include "test/testsupport/file_utils.h" #ifdef WEBRTC_ANDROID #include "pc/test/android_test_initializer.h" diff --git a/third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc b/third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc index 66581ca8525a..590fa9010299 100644 --- a/third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc @@ -8,21 +8,56 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/call/call_factory_interface.h" +#include "api/jsep.h" +#include "api/media_stream_interface.h" +#include "api/media_types.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" +#include "api/rtp_parameters.h" +#include "api/rtp_receiver_interface.h" +#include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_direction.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" #include "api/task_queue/default_task_queue_factory.h" +#include "api/task_queue/task_queue_factory.h" #include "api/transport/field_trial_based_config.h" +#include "api/transport/sctp_transport_factory_interface.h" +#include "api/transport/webrtc_key_value_config.h" +#include "media/base/media_engine.h" +#include "media/base/stream_params.h" #include "media/engine/webrtc_media_engine.h" #include "media/engine/webrtc_media_engine_defaults.h" +#include "modules/audio_device/include/audio_device.h" +#include "p2p/base/p2p_constants.h" +#include "p2p/base/port_allocator.h" +#include "p2p/base/transport_info.h" +#include "pc/channel_interface.h" #include "pc/media_session.h" -#include "pc/peer_connection_factory.h" #include "pc/peer_connection_wrapper.h" #include "pc/sdp_utils.h" +#include "pc/session_description.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/thread.h" +#include "test/gtest.h" #ifdef WEBRTC_ANDROID #include "pc/test/android_test_initializer.h" #endif #include "pc/test/fake_audio_capture_module.h" -#include "rtc_base/gunit.h" #include "rtc_base/virtual_socket_server.h" #include "test/gmock.h" #include "test/pc/sctp/fake_sctp_transport.h" diff --git a/third_party/libwebrtc/pc/peer_connection_media_unittest.cc b/third_party/libwebrtc/pc/peer_connection_media_unittest.cc index 30034b4cd51b..2a3a0aff9970 100644 --- a/third_party/libwebrtc/pc/peer_connection_media_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_media_unittest.cc @@ -12,25 +12,56 @@ // PeerConnection and the underlying media engine, as well as tests that check // the media-related aspects of SDP. +#include +#include +#include +#include #include #include +#include #include +#include +#include +#include #include "absl/algorithm/container.h" #include "absl/types/optional.h" +#include "api/audio_options.h" #include "api/call/call_factory_interface.h" +#include "api/jsep.h" +#include "api/media_types.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" #include "api/rtc_event_log/rtc_event_log_factory.h" +#include "api/rtc_event_log/rtc_event_log_factory_interface.h" +#include "api/rtp_parameters.h" +#include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_direction.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" #include "api/task_queue/default_task_queue_factory.h" +#include "api/task_queue/task_queue_factory.h" +#include "media/base/codec.h" #include "media/base/fake_media_engine.h" +#include "media/base/media_constants.h" +#include "media/base/media_engine.h" +#include "media/base/stream_params.h" #include "p2p/base/fake_port_allocator.h" +#include "p2p/base/p2p_constants.h" +#include "p2p/base/port_allocator.h" +#include "p2p/base/transport_info.h" #include "pc/media_session.h" #include "pc/peer_connection_wrapper.h" #include "pc/rtp_media_utils.h" -#include "pc/sdp_utils.h" +#include "pc/session_description.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/checks.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/thread.h" +#include "test/gtest.h" #ifdef WEBRTC_ANDROID #include "pc/test/android_test_initializer.h" #endif -#include "pc/test/fake_rtc_certificate_generator.h" #include "rtc_base/gunit.h" #include "rtc_base/virtual_socket_server.h" #include "test/gmock.h" diff --git a/third_party/libwebrtc/pc/peer_connection_message_handler.cc b/third_party/libwebrtc/pc/peer_connection_message_handler.cc index 54f75f00a99d..77db3e45e2a0 100644 --- a/third_party/libwebrtc/pc/peer_connection_message_handler.cc +++ b/third_party/libwebrtc/pc/peer_connection_message_handler.cc @@ -10,6 +10,7 @@ #include "pc/peer_connection_message_handler.h" +#include #include #include "api/jsep.h" diff --git a/third_party/libwebrtc/pc/peer_connection_rampup_tests.cc b/third_party/libwebrtc/pc/peer_connection_rampup_tests.cc index 692ca9d68901..dc5a11f58ad2 100644 --- a/third_party/libwebrtc/pc/peer_connection_rampup_tests.cc +++ b/third_party/libwebrtc/pc/peer_connection_rampup_tests.cc @@ -15,8 +15,6 @@ #include "absl/types/optional.h" #include "api/audio/audio_mixer.h" -#include "api/audio_codecs/audio_decoder_factory.h" -#include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/audio_options.h" @@ -24,14 +22,13 @@ #include "api/jsep.h" #include "api/media_stream_interface.h" #include "api/peer_connection_interface.h" +#include "api/rtc_error.h" #include "api/scoped_refptr.h" #include "api/stats/rtc_stats.h" #include "api/stats/rtc_stats_report.h" #include "api/stats/rtcstats_objects.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" -#include "api/video_codecs/video_decoder_factory.h" -#include "api/video_codecs/video_encoder_factory.h" #include "modules/audio_device/include/audio_device.h" #include "modules/audio_processing/include/audio_processing.h" #include "p2p/base/port_allocator.h" @@ -51,6 +48,7 @@ #include "rtc_base/location.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/socket_address.h" +#include "rtc_base/socket_factory.h" #include "rtc_base/ssl_certificate.h" #include "rtc_base/test_certificate_verifier.h" #include "rtc_base/thread.h" diff --git a/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc b/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc index fac738b7bab0..6c08ba47bc42 100644 --- a/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc @@ -8,8 +8,9 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include +#include +#include #include #include #include @@ -17,8 +18,6 @@ #include "absl/types/optional.h" #include "api/audio/audio_mixer.h" -#include "api/audio_codecs/audio_decoder_factory.h" -#include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/create_peerconnection_factory.h" @@ -30,14 +29,13 @@ #include "api/rtp_parameters.h" #include "api/rtp_receiver_interface.h" #include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_direction.h" #include "api/rtp_transceiver_interface.h" #include "api/scoped_refptr.h" #include "api/set_remote_description_observer_interface.h" #include "api/uma_metrics.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" -#include "api/video_codecs/video_decoder_factory.h" -#include "api/video_codecs/video_encoder_factory.h" #include "media/base/stream_params.h" #include "modules/audio_device/include/audio_device.h" #include "modules/audio_processing/include/audio_processing.h" diff --git a/third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc b/third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc index 90dd868de6f9..5923d2c47f77 100644 --- a/third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc @@ -12,20 +12,52 @@ // machine, as well as tests that check basic, media-agnostic aspects of SDP. #include +#include +#include +#include #include +#include +#include #include +#include +#include +#include +#include "absl/types/optional.h" +#include "api/audio/audio_mixer.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/create_peerconnection_factory.h" -#include "api/jsep_session_description.h" +#include "api/dtls_transport_interface.h" +#include "api/jsep.h" +#include "api/media_types.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" +#include "api/rtp_receiver_interface.h" +#include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/set_local_description_observer_interface.h" +#include "api/set_remote_description_observer_interface.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" +#include "media/base/codec.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "p2p/base/port_allocator.h" #include "pc/peer_connection.h" #include "pc/peer_connection_proxy.h" #include "pc/peer_connection_wrapper.h" #include "pc/sdp_utils.h" -#include "pc/webrtc_sdp.h" +#include "pc/session_description.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/checks.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/rtc_certificate.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/string_encode.h" +#include "rtc_base/thread.h" +#include "test/gtest.h" #ifdef WEBRTC_ANDROID #include "pc/test/android_test_initializer.h" #endif @@ -33,7 +65,6 @@ #include "pc/test/fake_rtc_certificate_generator.h" #include "rtc_base/gunit.h" #include "rtc_base/virtual_socket_server.h" -#include "test/gmock.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc b/third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc index 31385754b7db..10c4f3970303 100644 --- a/third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc @@ -8,26 +8,51 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include #include #include // no-presubmit-check TODO(webrtc:8982) +#include +#include +#include #include "absl/algorithm/container.h" +#include "absl/strings/string_view.h" +#include "api/audio/audio_mixer.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/create_peerconnection_factory.h" +#include "api/jsep.h" #include "api/media_types.h" +#include "api/peer_connection_interface.h" #include "api/rtc_error.h" +#include "api/rtp_parameters.h" +#include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_direction.h" #include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" #include "api/uma_metrics.h" +#include "api/video/video_codec_constants.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" -#include "pc/peer_connection.h" +#include "media/base/rid_description.h" +#include "media/base/stream_params.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "pc/channel_interface.h" #include "pc/peer_connection_wrapper.h" +#include "pc/session_description.h" +#include "pc/simulcast_description.h" #include "pc/test/fake_audio_capture_module.h" #include "pc/test/mock_peer_connection_observers.h" -#include "rtc_base/gunit.h" +#include "rtc_base/checks.h" +#include "rtc_base/strings/string_builder.h" +#include "rtc_base/thread.h" +#include "rtc_base/unique_id_generator.h" #include "system_wrappers/include/metrics.h" #include "test/gmock.h" +#include "test/gtest.h" using ::testing::Contains; using ::testing::Each; diff --git a/third_party/libwebrtc/pc/peer_connection_wrapper.cc b/third_party/libwebrtc/pc/peer_connection_wrapper.cc index 641d8bf053b9..6a7a30edc083 100644 --- a/third_party/libwebrtc/pc/peer_connection_wrapper.cc +++ b/third_party/libwebrtc/pc/peer_connection_wrapper.cc @@ -12,8 +12,6 @@ #include -#include -#include #include #include diff --git a/third_party/libwebrtc/pc/proxy.h b/third_party/libwebrtc/pc/proxy.h index 85cb70d34c15..e48f479183d5 100644 --- a/third_party/libwebrtc/pc/proxy.h +++ b/third_party/libwebrtc/pc/proxy.h @@ -59,6 +59,8 @@ #ifndef PC_PROXY_H_ #define PC_PROXY_H_ +#include + #include #include #include @@ -69,6 +71,7 @@ #include "api/task_queue/queued_task.h" #include "api/task_queue/task_queue_base.h" #include "rtc_base/event.h" +#include "rtc_base/location.h" #include "rtc_base/message_handler.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/string_utils.h" diff --git a/third_party/libwebrtc/pc/remote_audio_source.cc b/third_party/libwebrtc/pc/remote_audio_source.cc index 78a35f32a8e3..781e4512beb5 100644 --- a/third_party/libwebrtc/pc/remote_audio_source.cc +++ b/third_party/libwebrtc/pc/remote_audio_source.cc @@ -13,6 +13,7 @@ #include #include +#include #include "absl/algorithm/container.h" #include "api/scoped_refptr.h" diff --git a/third_party/libwebrtc/pc/rtc_stats_collector.cc b/third_party/libwebrtc/pc/rtc_stats_collector.cc index 7e9807e4490e..13c62ec975e2 100644 --- a/third_party/libwebrtc/pc/rtc_stats_collector.cc +++ b/third_party/libwebrtc/pc/rtc_stats_collector.cc @@ -10,26 +10,28 @@ #include "pc/rtc_stats_collector.h" +#include #include -#include #include #include #include #include +#include #include #include +#include "absl/strings/string_view.h" #include "api/array_view.h" #include "api/candidate.h" +#include "api/dtls_transport_interface.h" #include "api/media_stream_interface.h" #include "api/rtp_parameters.h" -#include "api/rtp_receiver_interface.h" -#include "api/rtp_sender_interface.h" #include "api/sequence_checker.h" #include "api/stats/rtc_stats.h" #include "api/stats/rtcstats_objects.h" #include "api/task_queue/queued_task.h" +#include "api/units/time_delta.h" #include "api/video/video_content_type.h" #include "common_video/include/quality_limitation_reason.h" #include "media/base/media_channel.h" @@ -37,7 +39,6 @@ #include "modules/rtp_rtcp/include/report_block_data.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "p2p/base/connection_info.h" -#include "p2p/base/dtls_transport_internal.h" #include "p2p/base/ice_transport_internal.h" #include "p2p/base/p2p_constants.h" #include "p2p/base/port.h" @@ -45,6 +46,8 @@ #include "pc/channel_interface.h" #include "pc/data_channel_utils.h" #include "pc/rtc_stats_traversal.h" +#include "pc/rtp_receiver_proxy.h" +#include "pc/rtp_sender_proxy.h" #include "pc/webrtc_sdp.h" #include "rtc_base/checks.h" #include "rtc_base/ip_address.h" diff --git a/third_party/libwebrtc/pc/rtc_stats_collector.h b/third_party/libwebrtc/pc/rtc_stats_collector.h index c84e6d3fef52..e6d9d184fe7a 100644 --- a/third_party/libwebrtc/pc/rtc_stats_collector.h +++ b/third_party/libwebrtc/pc/rtc_stats_collector.h @@ -12,6 +12,8 @@ #define PC_RTC_STATS_COLLECTOR_H_ #include + +#include #include #include #include diff --git a/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc b/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc index b39621d862f1..d33760abcb29 100644 --- a/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc +++ b/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc @@ -11,28 +11,45 @@ #include "pc/rtc_stats_collector.h" #include +#include +#include #include #include #include #include #include +#include #include #include -#include "absl/memory/memory.h" #include "absl/strings/str_replace.h" +#include "api/candidate.h" #include "api/dtls_transport_interface.h" +#include "api/media_stream_interface.h" #include "api/media_stream_track.h" #include "api/rtp_parameters.h" +#include "api/stats/rtc_stats.h" #include "api/stats/rtc_stats_report.h" #include "api/stats/rtcstats_objects.h" #include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video/recordable_encoded_frame.h" +#include "api/video/video_content_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "common_video/include/quality_limitation_reason.h" +#include "media/base/media_channel.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" #include "modules/rtp_rtcp/include/report_block_data.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "p2p/base/connection_info.h" +#include "p2p/base/ice_transport_internal.h" #include "p2p/base/p2p_constants.h" #include "p2p/base/port.h" #include "pc/media_stream.h" +#include "pc/stream_collection.h" #include "pc/test/fake_data_channel_provider.h" #include "pc/test/fake_peer_connection_for_stats.h" #include "pc/test/mock_data_channel.h" @@ -43,10 +60,19 @@ #include "rtc_base/fake_clock.h" #include "rtc_base/fake_ssl_identity.h" #include "rtc_base/gunit.h" -#include "rtc_base/logging.h" +#include "rtc_base/network_constants.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/rtc_certificate.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/ssl_fingerprint.h" +#include "rtc_base/ssl_identity.h" +#include "rtc_base/ssl_stream_adapter.h" +#include "rtc_base/string_encode.h" #include "rtc_base/strings/json.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/time_utils.h" +#include "test/gmock.h" +#include "test/gtest.h" using ::testing::_; using ::testing::AtLeast; diff --git a/third_party/libwebrtc/pc/rtc_stats_integrationtest.cc b/third_party/libwebrtc/pc/rtc_stats_integrationtest.cc index ad533499ab6b..be9cd6fbc057 100644 --- a/third_party/libwebrtc/pc/rtc_stats_integrationtest.cc +++ b/third_party/libwebrtc/pc/rtc_stats_integrationtest.cc @@ -11,7 +11,6 @@ #include #include -#include #include #include #include @@ -19,8 +18,6 @@ #include "absl/algorithm/container.h" #include "absl/strings/match.h" -#include "api/audio_codecs/audio_decoder_factory.h" -#include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/audio_options.h" diff --git a/third_party/libwebrtc/pc/rtc_stats_traversal_unittest.cc b/third_party/libwebrtc/pc/rtc_stats_traversal_unittest.cc index c7d097117e9b..6e9b78431306 100644 --- a/third_party/libwebrtc/pc/rtc_stats_traversal_unittest.cc +++ b/third_party/libwebrtc/pc/rtc_stats_traversal_unittest.cc @@ -11,7 +11,6 @@ #include "pc/rtc_stats_traversal.h" #include -#include #include #include "api/stats/rtcstats_objects.h" diff --git a/third_party/libwebrtc/pc/rtp_media_utils.h b/third_party/libwebrtc/pc/rtp_media_utils.h index 6f7986f096c8..5c61f5b1d6d7 100644 --- a/third_party/libwebrtc/pc/rtp_media_utils.h +++ b/third_party/libwebrtc/pc/rtp_media_utils.h @@ -11,6 +11,8 @@ #ifndef PC_RTP_MEDIA_UTILS_H_ #define PC_RTP_MEDIA_UTILS_H_ +#include // no-presubmit-check TODO(webrtc:8982) + #include "api/rtp_transceiver_direction.h" #include "api/rtp_transceiver_interface.h" diff --git a/third_party/libwebrtc/pc/rtp_parameters_conversion.cc b/third_party/libwebrtc/pc/rtp_parameters_conversion.cc index 8d3064ed9370..afba4bc94f9e 100644 --- a/third_party/libwebrtc/pc/rtp_parameters_conversion.cc +++ b/third_party/libwebrtc/pc/rtp_parameters_conversion.cc @@ -10,10 +10,10 @@ #include "pc/rtp_parameters_conversion.h" -#include #include #include #include +#include #include #include "api/array_view.h" diff --git a/third_party/libwebrtc/pc/rtp_parameters_conversion.h b/third_party/libwebrtc/pc/rtp_parameters_conversion.h index 62e468572258..959f3fde47f0 100644 --- a/third_party/libwebrtc/pc/rtp_parameters_conversion.h +++ b/third_party/libwebrtc/pc/rtp_parameters_conversion.h @@ -11,7 +11,6 @@ #ifndef PC_RTP_PARAMETERS_CONVERSION_H_ #define PC_RTP_PARAMETERS_CONVERSION_H_ -#include #include #include "absl/types/optional.h" diff --git a/third_party/libwebrtc/pc/rtp_parameters_conversion_unittest.cc b/third_party/libwebrtc/pc/rtp_parameters_conversion_unittest.cc index 99d976abcd2a..50d90e1c30e8 100644 --- a/third_party/libwebrtc/pc/rtp_parameters_conversion_unittest.cc +++ b/third_party/libwebrtc/pc/rtp_parameters_conversion_unittest.cc @@ -10,10 +10,13 @@ #include "pc/rtp_parameters_conversion.h" -#include +#include +#include +#include -#include "rtc_base/gunit.h" +#include "api/media_types.h" #include "test/gmock.h" +#include "test/gtest.h" using ::testing::UnorderedElementsAre; diff --git a/third_party/libwebrtc/pc/rtp_receiver.cc b/third_party/libwebrtc/pc/rtp_receiver.cc index 2444c9b60da6..a2b3353c0e2e 100644 --- a/third_party/libwebrtc/pc/rtp_receiver.cc +++ b/third_party/libwebrtc/pc/rtp_receiver.cc @@ -17,7 +17,7 @@ #include "pc/media_stream.h" #include "pc/media_stream_proxy.h" -#include "rtc_base/location.h" +#include "rtc_base/thread.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/rtp_sender.cc b/third_party/libwebrtc/pc/rtp_sender.cc index 110b5aae0a01..dc53105b6dc9 100644 --- a/third_party/libwebrtc/pc/rtp_sender.cc +++ b/third_party/libwebrtc/pc/rtp_sender.cc @@ -12,6 +12,7 @@ #include #include +#include #include #include diff --git a/third_party/libwebrtc/pc/rtp_sender.h b/third_party/libwebrtc/pc/rtp_sender.h index ca2d1385ce5f..569a6007d389 100644 --- a/third_party/libwebrtc/pc/rtp_sender.h +++ b/third_party/libwebrtc/pc/rtp_sender.h @@ -17,6 +17,7 @@ #include #include + #include #include #include @@ -32,13 +33,16 @@ #include "api/rtp_parameters.h" #include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" +#include "api/sequence_checker.h" #include "media/base/audio_source.h" #include "media/base/media_channel.h" #include "pc/dtmf_sender.h" #include "pc/stats_collector_interface.h" +#include "rtc_base/checks.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/third_party/sigslot/sigslot.h" #include "rtc_base/thread.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc index d947b8b3e975..c0b09e39c3df 100644 --- a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc +++ b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc @@ -11,6 +11,7 @@ #include #include +#include #include #include #include @@ -28,15 +29,20 @@ #include "api/rtc_error.h" #include "api/rtc_event_log/rtc_event_log.h" #include "api/rtp_parameters.h" +#include "api/rtp_receiver_interface.h" #include "api/scoped_refptr.h" #include "api/test/fake_frame_decryptor.h" #include "api/test/fake_frame_encryptor.h" #include "api/video/builtin_video_bitrate_allocator_factory.h" +#include "api/video/video_bitrate_allocator_factory.h" +#include "api/video/video_codec_constants.h" #include "media/base/codec.h" +#include "media/base/delayable.h" #include "media/base/fake_media_engine.h" #include "media/base/media_channel.h" #include "media/base/media_config.h" #include "media/base/media_engine.h" +#include "media/base/rid_description.h" #include "media/base/stream_params.h" #include "media/base/test_utils.h" #include "media/engine/fake_webrtc_call.h" @@ -50,8 +56,6 @@ #include "pc/dtls_srtp_transport.h" #include "pc/local_audio_source.h" #include "pc/media_stream.h" -#include "pc/remote_audio_source.h" -#include "pc/rtp_receiver.h" #include "pc/rtp_sender.h" #include "pc/rtp_transport_internal.h" #include "pc/test/fake_video_track_source.h" @@ -59,6 +63,8 @@ #include "pc/video_track.h" #include "rtc_base/checks.h" #include "rtc_base/gunit.h" +#include "rtc_base/location.h" +#include "rtc_base/ref_counted_object.h" #include "rtc_base/third_party/sigslot/sigslot.h" #include "rtc_base/thread.h" #include "test/gmock.h" diff --git a/third_party/libwebrtc/pc/rtp_transceiver.cc b/third_party/libwebrtc/pc/rtp_transceiver.cc index 013277fa5376..5e3a084d2d8d 100644 --- a/third_party/libwebrtc/pc/rtp_transceiver.cc +++ b/third_party/libwebrtc/pc/rtp_transceiver.cc @@ -10,6 +10,7 @@ #include "pc/rtp_transceiver.h" +#include #include #include #include @@ -24,6 +25,7 @@ #include "pc/rtp_media_utils.h" #include "pc/session_description.h" #include "rtc_base/checks.h" +#include "rtc_base/location.h" #include "rtc_base/logging.h" #include "rtc_base/task_utils/to_queued_task.h" #include "rtc_base/thread.h" diff --git a/third_party/libwebrtc/pc/rtp_transceiver.h b/third_party/libwebrtc/pc/rtp_transceiver.h index b8dbb677dd55..e7e3fb9be1dd 100644 --- a/third_party/libwebrtc/pc/rtp_transceiver.h +++ b/third_party/libwebrtc/pc/rtp_transceiver.h @@ -20,9 +20,12 @@ #include "absl/types/optional.h" #include "api/array_view.h" +#include "api/jsep.h" #include "api/media_types.h" #include "api/rtc_error.h" #include "api/rtp_parameters.h" +#include "api/rtp_receiver_interface.h" +#include "api/rtp_sender_interface.h" #include "api/rtp_transceiver_direction.h" #include "api/rtp_transceiver_interface.h" #include "api/scoped_refptr.h" @@ -34,6 +37,8 @@ #include "pc/rtp_receiver_proxy.h" #include "pc/rtp_sender.h" #include "pc/rtp_sender_proxy.h" +#include "pc/rtp_transport_internal.h" +#include "pc/session_description.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/task_utils/pending_task_safety_flag.h" #include "rtc_base/third_party/sigslot/sigslot.h" diff --git a/third_party/libwebrtc/pc/rtp_transceiver_unittest.cc b/third_party/libwebrtc/pc/rtp_transceiver_unittest.cc index df6dd29f3106..e5c77331780f 100644 --- a/third_party/libwebrtc/pc/rtp_transceiver_unittest.cc +++ b/third_party/libwebrtc/pc/rtp_transceiver_unittest.cc @@ -14,12 +14,15 @@ #include +#include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/rtp_parameters.h" #include "media/base/fake_media_engine.h" +#include "media/base/media_engine.h" #include "pc/test/mock_channel_interface.h" #include "pc/test/mock_rtp_receiver_internal.h" #include "pc/test/mock_rtp_sender_internal.h" +#include "rtc_base/thread.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/rtp_transmission_manager.cc b/third_party/libwebrtc/pc/rtp_transmission_manager.cc index 130dc311a48d..5dbb7650983f 100644 --- a/third_party/libwebrtc/pc/rtp_transmission_manager.cc +++ b/third_party/libwebrtc/pc/rtp_transmission_manager.cc @@ -10,7 +10,7 @@ #include "pc/rtp_transmission_manager.h" -#include +#include #include #include "absl/types/optional.h" @@ -23,6 +23,7 @@ #include "rtc_base/checks.h" #include "rtc_base/helpers.h" #include "rtc_base/logging.h" +#include "rtc_base/ref_counted_object.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/rtp_transmission_manager.h b/third_party/libwebrtc/pc/rtp_transmission_manager.h index 209c8408deda..3496bfa54d8b 100644 --- a/third_party/libwebrtc/pc/rtp_transmission_manager.h +++ b/third_party/libwebrtc/pc/rtp_transmission_manager.h @@ -29,7 +29,9 @@ #include "media/base/media_channel.h" #include "pc/channel_manager.h" #include "pc/rtp_receiver.h" +#include "pc/rtp_receiver_proxy.h" #include "pc/rtp_sender.h" +#include "pc/rtp_sender_proxy.h" #include "pc/rtp_transceiver.h" #include "pc/stats_collector_interface.h" #include "pc/transceiver_list.h" diff --git a/third_party/libwebrtc/pc/rtp_transport.cc b/third_party/libwebrtc/pc/rtp_transport.cc index d4edb9501c6c..334dc4d0b20c 100644 --- a/third_party/libwebrtc/pc/rtp_transport.cc +++ b/third_party/libwebrtc/pc/rtp_transport.cc @@ -11,11 +11,13 @@ #include "pc/rtp_transport.h" #include -#include + +#include #include #include "absl/strings/string_view.h" #include "api/array_view.h" +#include "api/units/timestamp.h" #include "media/base/rtp_utils.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/checks.h" diff --git a/third_party/libwebrtc/pc/rtp_transport.h b/third_party/libwebrtc/pc/rtp_transport.h index 893d91e734ff..39d4ad5b54ac 100644 --- a/third_party/libwebrtc/pc/rtp_transport.h +++ b/third_party/libwebrtc/pc/rtp_transport.h @@ -18,6 +18,7 @@ #include "absl/types/optional.h" #include "call/rtp_demuxer.h" +#include "call/video_receive_stream.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "p2p/base/packet_transport_internal.h" #include "pc/rtp_transport_internal.h" diff --git a/third_party/libwebrtc/pc/rtp_transport_unittest.cc b/third_party/libwebrtc/pc/rtp_transport_unittest.cc index aae6d2c46222..0e6af734f396 100644 --- a/third_party/libwebrtc/pc/rtp_transport_unittest.cc +++ b/third_party/libwebrtc/pc/rtp_transport_unittest.cc @@ -10,16 +10,10 @@ #include "pc/rtp_transport.h" -#include -#include -#include -#include - -#include "api/rtp_headers.h" -#include "api/rtp_parameters.h" #include "p2p/base/fake_packet_transport.h" #include "pc/test/rtp_transport_test_util.h" #include "rtc_base/buffer.h" +#include "rtc_base/containers/flat_set.h" #include "rtc_base/third_party/sigslot/sigslot.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/sctp_data_channel_transport.cc b/third_party/libwebrtc/pc/sctp_data_channel_transport.cc index f01f86ebd832..626d1757b701 100644 --- a/third_party/libwebrtc/pc/sctp_data_channel_transport.cc +++ b/third_party/libwebrtc/pc/sctp_data_channel_transport.cc @@ -10,9 +10,6 @@ #include "pc/sctp_data_channel_transport.h" -#include "absl/types/optional.h" -#include "pc/sctp_utils.h" - namespace webrtc { SctpDataChannelTransport::SctpDataChannelTransport( diff --git a/third_party/libwebrtc/pc/sctp_transport.h b/third_party/libwebrtc/pc/sctp_transport.h index 16b98407b694..4981db4eded7 100644 --- a/third_party/libwebrtc/pc/sctp_transport.h +++ b/third_party/libwebrtc/pc/sctp_transport.h @@ -16,9 +16,11 @@ #include "api/dtls_transport_interface.h" #include "api/scoped_refptr.h" #include "api/sctp_transport_interface.h" +#include "api/sequence_checker.h" #include "media/sctp/sctp_transport_internal.h" #include "p2p/base/dtls_transport_internal.h" #include "pc/dtls_transport.h" +#include "rtc_base/checks.h" #include "rtc_base/third_party/sigslot/sigslot.h" #include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" diff --git a/third_party/libwebrtc/pc/sctp_transport_unittest.cc b/third_party/libwebrtc/pc/sctp_transport_unittest.cc index 679b481f4cc9..e3168d8b9959 100644 --- a/third_party/libwebrtc/pc/sctp_transport_unittest.cc +++ b/third_party/libwebrtc/pc/sctp_transport_unittest.cc @@ -14,10 +14,17 @@ #include #include "absl/memory/memory.h" +#include "absl/types/optional.h" #include "api/dtls_transport_interface.h" +#include "api/transport/data_channel_transport_interface.h" +#include "media/base/media_channel.h" #include "p2p/base/fake_dtls_transport.h" +#include "p2p/base/p2p_constants.h" +#include "p2p/base/packet_transport_internal.h" #include "pc/dtls_transport.h" +#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/gunit.h" +#include "rtc_base/ref_counted_object.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/sctp_utils.cc b/third_party/libwebrtc/pc/sctp_utils.cc index f7458405eac2..c60e339b0876 100644 --- a/third_party/libwebrtc/pc/sctp_utils.cc +++ b/third_party/libwebrtc/pc/sctp_utils.cc @@ -11,12 +11,12 @@ #include "pc/sctp_utils.h" #include -#include + +#include #include "absl/types/optional.h" #include "api/priority.h" #include "rtc_base/byte_buffer.h" -#include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/logging.h" diff --git a/third_party/libwebrtc/pc/sctp_utils_unittest.cc b/third_party/libwebrtc/pc/sctp_utils_unittest.cc index af14fe4f6b2d..146886b8cb5d 100644 --- a/third_party/libwebrtc/pc/sctp_utils_unittest.cc +++ b/third_party/libwebrtc/pc/sctp_utils_unittest.cc @@ -12,6 +12,8 @@ #include +#include "absl/types/optional.h" +#include "api/priority.h" #include "rtc_base/byte_buffer.h" #include "rtc_base/copy_on_write_buffer.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/sdp_offer_answer.h b/third_party/libwebrtc/pc/sdp_offer_answer.h index 622d57af2475..67ead4724241 100644 --- a/third_party/libwebrtc/pc/sdp_offer_answer.h +++ b/third_party/libwebrtc/pc/sdp_offer_answer.h @@ -15,7 +15,6 @@ #include #include -#include #include #include #include diff --git a/third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc b/third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc index df343cce3a4f..b992c3490698 100644 --- a/third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc +++ b/third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc @@ -8,53 +8,29 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include - #include -#include #include #include -#include "absl/types/optional.h" #include "api/audio/audio_mixer.h" -#include "api/audio_codecs/audio_decoder_factory.h" -#include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/create_peerconnection_factory.h" -#include "api/jsep.h" -#include "api/media_stream_interface.h" #include "api/media_types.h" #include "api/peer_connection_interface.h" -#include "api/rtc_error.h" -#include "api/rtp_parameters.h" -#include "api/rtp_receiver_interface.h" -#include "api/rtp_sender_interface.h" #include "api/rtp_transceiver_interface.h" #include "api/scoped_refptr.h" -#include "api/set_remote_description_observer_interface.h" -#include "api/uma_metrics.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" -#include "api/video_codecs/video_decoder_factory.h" -#include "api/video_codecs/video_encoder_factory.h" -#include "media/base/stream_params.h" #include "modules/audio_device/include/audio_device.h" #include "modules/audio_processing/include/audio_processing.h" #include "p2p/base/port_allocator.h" -#include "pc/media_session.h" #include "pc/peer_connection_wrapper.h" -#include "pc/sdp_utils.h" -#include "pc/session_description.h" #include "pc/test/fake_audio_capture_module.h" #include "pc/test/mock_peer_connection_observers.h" -#include "rtc_base/checks.h" -#include "rtc_base/gunit.h" -#include "rtc_base/ref_counted_object.h" #include "rtc_base/rtc_certificate_generator.h" #include "rtc_base/thread.h" #include "system_wrappers/include/metrics.h" -#include "test/gmock.h" #include "test/gtest.h" // This file contains unit tests that relate to the behavior of the diff --git a/third_party/libwebrtc/pc/sdp_serializer.cc b/third_party/libwebrtc/pc/sdp_serializer.cc index cdd2f2e9adc4..6d405d07a9df 100644 --- a/third_party/libwebrtc/pc/sdp_serializer.cc +++ b/third_party/libwebrtc/pc/sdp_serializer.cc @@ -10,9 +10,9 @@ #include "pc/sdp_serializer.h" -#include #include #include +#include #include #include diff --git a/third_party/libwebrtc/pc/sdp_serializer_unittest.cc b/third_party/libwebrtc/pc/sdp_serializer_unittest.cc index 68d4c2acef28..0c31750df452 100644 --- a/third_party/libwebrtc/pc/sdp_serializer_unittest.cc +++ b/third_party/libwebrtc/pc/sdp_serializer_unittest.cc @@ -10,12 +10,14 @@ #include "pc/sdp_serializer.h" +#include + #include #include #include #include -#include "rtc_base/gunit.h" +#include "test/gtest.h" using cricket::RidDescription; using cricket::RidDirection; diff --git a/third_party/libwebrtc/pc/sdp_utils.cc b/third_party/libwebrtc/pc/sdp_utils.cc index b750b04a4684..ca61f0013f51 100644 --- a/third_party/libwebrtc/pc/sdp_utils.cc +++ b/third_party/libwebrtc/pc/sdp_utils.cc @@ -10,8 +10,8 @@ #include "pc/sdp_utils.h" -#include #include +#include #include "api/jsep_session_description.h" #include "rtc_base/checks.h" diff --git a/third_party/libwebrtc/pc/session_description.cc b/third_party/libwebrtc/pc/session_description.cc index 7b878cbf7be1..c1feedbf53d3 100644 --- a/third_party/libwebrtc/pc/session_description.cc +++ b/third_party/libwebrtc/pc/session_description.cc @@ -10,11 +10,10 @@ #include "pc/session_description.h" -#include - #include "absl/algorithm/container.h" #include "absl/memory/memory.h" #include "rtc_base/checks.h" +#include "rtc_base/strings/string_builder.h" namespace cricket { namespace { diff --git a/third_party/libwebrtc/pc/session_description.h b/third_party/libwebrtc/pc/session_description.h index ee7a91c84c62..a68c312f4279 100644 --- a/third_party/libwebrtc/pc/session_description.h +++ b/third_party/libwebrtc/pc/session_description.h @@ -15,9 +15,9 @@ #include #include -#include #include #include +#include #include #include diff --git a/third_party/libwebrtc/pc/session_description_unittest.cc b/third_party/libwebrtc/pc/session_description_unittest.cc index 00ce538398e6..4d0913bad202 100644 --- a/third_party/libwebrtc/pc/session_description_unittest.cc +++ b/third_party/libwebrtc/pc/session_description_unittest.cc @@ -9,8 +9,6 @@ */ #include "pc/session_description.h" -#include - #include "test/gtest.h" namespace cricket { diff --git a/third_party/libwebrtc/pc/srtp_filter.cc b/third_party/libwebrtc/pc/srtp_filter.cc index c48dfdb4cd0d..9d7f39a7a319 100644 --- a/third_party/libwebrtc/pc/srtp_filter.cc +++ b/third_party/libwebrtc/pc/srtp_filter.cc @@ -11,8 +11,8 @@ #include "pc/srtp_filter.h" #include -#include -#include + +#include #include "absl/strings/match.h" #include "rtc_base/logging.h" diff --git a/third_party/libwebrtc/pc/srtp_session.cc b/third_party/libwebrtc/pc/srtp_session.cc index 76ab3a8fe806..a81f2415a5a5 100644 --- a/third_party/libwebrtc/pc/srtp_session.cc +++ b/third_party/libwebrtc/pc/srtp_session.cc @@ -10,12 +10,18 @@ #include "pc/srtp_session.h" +#include + #include +#include #include "absl/base/attributes.h" +#include "absl/base/const_init.h" #include "api/array_view.h" #include "modules/rtp_rtcp/source/rtp_util.h" #include "pc/external_hmac.h" +#include "rtc_base/byte_order.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/ssl_stream_adapter.h" #include "rtc_base/string_encode.h" diff --git a/third_party/libwebrtc/pc/srtp_session.h b/third_party/libwebrtc/pc/srtp_session.h index f1b6a52b47d8..d88eaae3199e 100644 --- a/third_party/libwebrtc/pc/srtp_session.h +++ b/third_party/libwebrtc/pc/srtp_session.h @@ -11,6 +11,9 @@ #ifndef PC_SRTP_SESSION_H_ #define PC_SRTP_SESSION_H_ +#include +#include + #include #include "api/scoped_refptr.h" diff --git a/third_party/libwebrtc/pc/srtp_transport_unittest.cc b/third_party/libwebrtc/pc/srtp_transport_unittest.cc index 59bc8e809093..980ebca08a45 100644 --- a/third_party/libwebrtc/pc/srtp_transport_unittest.cc +++ b/third_party/libwebrtc/pc/srtp_transport_unittest.cc @@ -12,8 +12,6 @@ #include -#include -#include #include #include "call/rtp_demuxer.h" @@ -25,6 +23,7 @@ #include "rtc_base/async_packet_socket.h" #include "rtc_base/byte_order.h" #include "rtc_base/checks.h" +#include "rtc_base/containers/flat_set.h" #include "rtc_base/ssl_stream_adapter.h" #include "rtc_base/third_party/sigslot/sigslot.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/stats_collector.cc b/third_party/libwebrtc/pc/stats_collector.cc index 493c26cfbdcc..6b1cda3ad567 100644 --- a/third_party/libwebrtc/pc/stats_collector.cc +++ b/third_party/libwebrtc/pc/stats_collector.cc @@ -13,18 +13,19 @@ #include #include +#include #include -#include +#include #include #include #include +#include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/audio_codecs/audio_encoder.h" #include "api/candidate.h" #include "api/data_channel_interface.h" #include "api/media_types.h" -#include "api/rtp_receiver_interface.h" #include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" #include "api/sequence_checker.h" @@ -39,6 +40,8 @@ #include "pc/channel_interface.h" #include "pc/data_channel_utils.h" #include "pc/rtp_receiver.h" +#include "pc/rtp_receiver_proxy.h" +#include "pc/rtp_sender_proxy.h" #include "pc/rtp_transceiver.h" #include "pc/transport_stats.h" #include "rtc_base/checks.h" diff --git a/third_party/libwebrtc/pc/stats_collector.h b/third_party/libwebrtc/pc/stats_collector.h index 71b802dd09ef..751a2de09cca 100644 --- a/third_party/libwebrtc/pc/stats_collector.h +++ b/third_party/libwebrtc/pc/stats_collector.h @@ -21,18 +21,24 @@ #include #include #include +#include #include #include +#include "absl/types/optional.h" #include "api/media_stream_interface.h" #include "api/peer_connection_interface.h" +#include "api/scoped_refptr.h" #include "api/stats_types.h" #include "p2p/base/connection_info.h" #include "p2p/base/port.h" #include "pc/peer_connection_internal.h" +#include "pc/rtp_transceiver.h" #include "pc/stats_collector_interface.h" +#include "pc/transport_stats.h" #include "rtc_base/network_constants.h" #include "rtc_base/ssl_certificate.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/stats_collector_unittest.cc b/third_party/libwebrtc/pc/stats_collector_unittest.cc index 7688ffe727e4..144ca34b550c 100644 --- a/third_party/libwebrtc/pc/stats_collector_unittest.cc +++ b/third_party/libwebrtc/pc/stats_collector_unittest.cc @@ -12,7 +12,7 @@ #include -#include +#include #include "absl/algorithm/container.h" #include "absl/types/optional.h" @@ -20,11 +20,16 @@ #include "api/candidate.h" #include "api/data_channel_interface.h" #include "api/media_stream_track.h" +#include "api/media_types.h" +#include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" #include "call/call.h" #include "media/base/media_channel.h" #include "modules/audio_processing/include/audio_processing_statistics.h" +#include "p2p/base/ice_transport_internal.h" #include "pc/media_stream.h" +#include "pc/rtp_receiver.h" +#include "pc/rtp_sender.h" #include "pc/sctp_data_channel.h" #include "pc/test/fake_peer_connection_for_stats.h" #include "pc/test/fake_video_track_source.h" @@ -43,6 +48,7 @@ #include "rtc_base/string_encode.h" #include "rtc_base/third_party/base64/base64.h" #include "rtc_base/thread.h" +#include "test/gmock.h" #include "test/gtest.h" using cricket::ConnectionInfo; diff --git a/third_party/libwebrtc/pc/track_media_info_map.cc b/third_party/libwebrtc/pc/track_media_info_map.cc index e68f2f7a5296..12670dda283c 100644 --- a/third_party/libwebrtc/pc/track_media_info_map.cc +++ b/third_party/libwebrtc/pc/track_media_info_map.cc @@ -12,7 +12,7 @@ #include #include -#include +#include #include #include "api/media_types.h" diff --git a/third_party/libwebrtc/pc/track_media_info_map_unittest.cc b/third_party/libwebrtc/pc/track_media_info_map_unittest.cc index a58331f0df16..8cf3360c6f9e 100644 --- a/third_party/libwebrtc/pc/track_media_info_map_unittest.cc +++ b/third_party/libwebrtc/pc/track_media_info_map_unittest.cc @@ -10,22 +10,28 @@ #include "pc/track_media_info_map.h" +#include + +#include #include -#include #include +#include #include #include -#include "api/rtp_sender_interface.h" +#include "api/media_types.h" +#include "api/rtp_parameters.h" #include "api/test/mock_video_track.h" -#include "api/transport/rtp/rtp_source.h" #include "media/base/media_channel.h" #include "pc/audio_track.h" #include "pc/test/fake_video_track_source.h" #include "pc/test/mock_rtp_receiver_internal.h" #include "pc/test/mock_rtp_sender_internal.h" #include "pc/video_track.h" -#include "rtc_base/ref_count.h" +#include "rtc_base/checks.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/thread.h" +#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/transceiver_list.cc b/third_party/libwebrtc/pc/transceiver_list.cc index 235c9af036da..139c498634c4 100644 --- a/third_party/libwebrtc/pc/transceiver_list.cc +++ b/third_party/libwebrtc/pc/transceiver_list.cc @@ -10,6 +10,8 @@ #include "pc/transceiver_list.h" +#include + #include "rtc_base/checks.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/used_ids_unittest.cc b/third_party/libwebrtc/pc/used_ids_unittest.cc index af66898450c9..6362f2773ad8 100644 --- a/third_party/libwebrtc/pc/used_ids_unittest.cc +++ b/third_party/libwebrtc/pc/used_ids_unittest.cc @@ -10,6 +10,7 @@ #include "pc/used_ids.h" +#include "absl/strings/string_view.h" #include "test/gtest.h" using cricket::UsedIds; diff --git a/third_party/libwebrtc/pc/video_rtp_receiver.cc b/third_party/libwebrtc/pc/video_rtp_receiver.cc index a428603745d4..7659d7c2f906 100644 --- a/third_party/libwebrtc/pc/video_rtp_receiver.cc +++ b/third_party/libwebrtc/pc/video_rtp_receiver.cc @@ -12,15 +12,16 @@ #include +#include #include #include #include "api/video/recordable_encoded_frame.h" -#include "api/video_track_source_proxy_factory.h" #include "pc/video_track.h" #include "rtc_base/checks.h" #include "rtc_base/location.h" #include "rtc_base/logging.h" +#include "rtc_base/ref_counted_object.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/video_rtp_receiver_unittest.cc b/third_party/libwebrtc/pc/video_rtp_receiver_unittest.cc index 56aa3688a866..c13214fcbb6e 100644 --- a/third_party/libwebrtc/pc/video_rtp_receiver_unittest.cc +++ b/third_party/libwebrtc/pc/video_rtp_receiver_unittest.cc @@ -10,11 +10,17 @@ #include "pc/video_rtp_receiver.h" +#include #include +#include "api/task_queue/task_queue_base.h" +#include "api/video/recordable_encoded_frame.h" #include "api/video/test/mock_recordable_encoded_frame.h" #include "media/base/fake_media_engine.h" +#include "rtc_base/location.h" +#include "rtc_base/ref_counted_object.h" #include "test/gmock.h" +#include "test/gtest.h" using ::testing::_; using ::testing::AnyNumber; diff --git a/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc b/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc index 5666b77d5fc0..bb1dc193debd 100644 --- a/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc +++ b/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc @@ -10,6 +10,12 @@ #include "pc/video_rtp_track_source.h" +#include "absl/types/optional.h" +#include "api/scoped_refptr.h" +#include "api/units/timestamp.h" +#include "api/video/color_space.h" +#include "api/video/encoded_image.h" +#include "api/video/video_codec_type.h" #include "rtc_base/ref_counted_object.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/video_track.cc b/third_party/libwebrtc/pc/video_track.cc index 4559181ce7e6..744800c9f347 100644 --- a/third_party/libwebrtc/pc/video_track.cc +++ b/third_party/libwebrtc/pc/video_track.cc @@ -10,7 +10,6 @@ #include "pc/video_track.h" -#include #include #include @@ -18,7 +17,6 @@ #include "api/sequence_checker.h" #include "rtc_base/checks.h" #include "rtc_base/location.h" -#include "rtc_base/logging.h" #include "rtc_base/ref_counted_object.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/video_track.h b/third_party/libwebrtc/pc/video_track.h index 66262d22d1aa..f938b3362c35 100644 --- a/third_party/libwebrtc/pc/video_track.h +++ b/third_party/libwebrtc/pc/video_track.h @@ -13,6 +13,7 @@ #include +#include "absl/types/optional.h" #include "api/media_stream_interface.h" #include "api/media_stream_track.h" #include "api/scoped_refptr.h" @@ -21,8 +22,8 @@ #include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" #include "media/base/video_source_base.h" -#include "rtc_base/system/no_unique_address.h" #include "pc/video_track_source_proxy.h" +#include "rtc_base/system/no_unique_address.h" #include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" diff --git a/third_party/libwebrtc/pc/video_track_source.h b/third_party/libwebrtc/pc/video_track_source.h index 3f568f642bd8..723b10d8f3cd 100644 --- a/third_party/libwebrtc/pc/video_track_source.h +++ b/third_party/libwebrtc/pc/video_track_source.h @@ -20,8 +20,10 @@ #include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" #include "media/base/media_channel.h" +#include "rtc_base/checks.h" #include "rtc_base/system/no_unique_address.h" #include "rtc_base/system/rtc_export.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/video_track_source_proxy.cc b/third_party/libwebrtc/pc/video_track_source_proxy.cc index 309c1f20f89a..26f0ecec988d 100644 --- a/third_party/libwebrtc/pc/video_track_source_proxy.cc +++ b/third_party/libwebrtc/pc/video_track_source_proxy.cc @@ -11,7 +11,9 @@ #include "pc/video_track_source_proxy.h" #include "api/media_stream_interface.h" +#include "api/scoped_refptr.h" #include "api/video_track_source_proxy_factory.h" +#include "rtc_base/thread.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/video_track_source_proxy.h b/third_party/libwebrtc/pc/video_track_source_proxy.h index 1f6d976ba80d..8500a9876672 100644 --- a/third_party/libwebrtc/pc/video_track_source_proxy.h +++ b/third_party/libwebrtc/pc/video_track_source_proxy.h @@ -11,7 +11,13 @@ #ifndef PC_VIDEO_TRACK_SOURCE_PROXY_H_ #define PC_VIDEO_TRACK_SOURCE_PROXY_H_ +#include "absl/types/optional.h" #include "api/media_stream_interface.h" +#include "api/video/recordable_encoded_frame.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "api/video_track_source_constraints.h" #include "pc/proxy.h" namespace webrtc { @@ -21,6 +27,7 @@ namespace webrtc { // TODO(deadbeef): Move this to .cc file. What threads methods are called on is // an implementation detail. BEGIN_PROXY_MAP(VideoTrackSource) + PROXY_PRIMARY_THREAD_DESTRUCTOR() PROXY_CONSTMETHOD0(SourceState, state) BYPASS_PROXY_CONSTMETHOD0(bool, remote) diff --git a/third_party/libwebrtc/pc/video_track_unittest.cc b/third_party/libwebrtc/pc/video_track_unittest.cc index e6dcce793945..2a10c93f7ba6 100644 --- a/third_party/libwebrtc/pc/video_track_unittest.cc +++ b/third_party/libwebrtc/pc/video_track_unittest.cc @@ -13,11 +13,11 @@ #include #include "media/base/fake_frame_source.h" -#include "media/base/video_common.h" #include "pc/test/fake_video_track_renderer.h" #include "pc/test/fake_video_track_source.h" #include "pc/video_track_source.h" #include "rtc_base/ref_counted_object.h" +#include "rtc_base/time_utils.h" #include "test/gtest.h" using webrtc::FakeVideoTrackRenderer; diff --git a/third_party/libwebrtc/pc/webrtc_sdp.cc b/third_party/libwebrtc/pc/webrtc_sdp.cc index 3f06f307a414..ea7a148fe175 100644 --- a/third_party/libwebrtc/pc/webrtc_sdp.cc +++ b/third_party/libwebrtc/pc/webrtc_sdp.cc @@ -12,9 +12,9 @@ #include #include -#include #include +#include #include #include #include @@ -31,6 +31,7 @@ #include "api/jsep_session_description.h" #include "api/media_types.h" // for RtpExtension +#include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/rtc_error.h" #include "api/rtp_parameters.h" diff --git a/third_party/libwebrtc/pc/webrtc_sdp_unittest.cc b/third_party/libwebrtc/pc/webrtc_sdp_unittest.cc index 21d682315ee7..ea1b07d06abe 100644 --- a/third_party/libwebrtc/pc/webrtc_sdp_unittest.cc +++ b/third_party/libwebrtc/pc/webrtc_sdp_unittest.cc @@ -11,10 +11,10 @@ #include #include -#include #include #include #include +#include #include #include #include @@ -22,21 +22,26 @@ #include "absl/algorithm/container.h" #include "absl/memory/memory.h" #include "absl/strings/str_replace.h" +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" #include "api/array_view.h" #include "api/crypto_params.h" #include "api/jsep_session_description.h" #include "api/media_types.h" #include "api/rtp_parameters.h" -#include "api/rtp_transceiver_interface.h" +#include "api/rtp_transceiver_direction.h" #include "media/base/codec.h" #include "media/base/media_constants.h" +#include "media/base/rid_description.h" #include "media/base/stream_params.h" #include "p2p/base/p2p_constants.h" #include "p2p/base/port.h" #include "p2p/base/transport_description.h" #include "p2p/base/transport_info.h" +#include "pc/media_protocol_names.h" #include "pc/media_session.h" #include "pc/session_description.h" +#include "pc/simulcast_description.h" #include "rtc_base/checks.h" #include "rtc_base/message_digest.h" #include "rtc_base/socket_address.h" diff --git a/third_party/libwebrtc/pc/webrtc_session_description_factory.cc b/third_party/libwebrtc/pc/webrtc_session_description_factory.cc index 36130758a332..82ba84954450 100644 --- a/third_party/libwebrtc/pc/webrtc_session_description_factory.cc +++ b/third_party/libwebrtc/pc/webrtc_session_description_factory.cc @@ -11,8 +11,8 @@ #include "pc/webrtc_session_description_factory.h" #include -#include -#include + +#include #include #include #include @@ -23,6 +23,7 @@ #include "api/jsep.h" #include "api/jsep_session_description.h" #include "api/rtc_error.h" +#include "api/sequence_checker.h" #include "pc/sdp_state_provider.h" #include "pc/session_description.h" #include "rtc_base/checks.h" @@ -32,6 +33,7 @@ #include "rtc_base/ssl_identity.h" #include "rtc_base/ssl_stream_adapter.h" #include "rtc_base/string_encode.h" +#include "rtc_base/unique_id_generator.h" using cricket::MediaSessionOptions; using rtc::UniqueRandomIdGenerator; diff --git a/third_party/libwebrtc/tools_webrtc/iwyu/apply-iwyu b/third_party/libwebrtc/tools_webrtc/iwyu/apply-iwyu index e7d50219503c..7f2809b76930 100755 --- a/third_party/libwebrtc/tools_webrtc/iwyu/apply-iwyu +++ b/third_party/libwebrtc/tools_webrtc/iwyu/apply-iwyu @@ -17,7 +17,7 @@ # Set this to 1 to get more debug information. # Set this to 2 to also get a dump of the iwyu tool output. -DEBUG=1 +DEBUG=2 set -e if [ $DEBUG -gt 0 ]; then @@ -26,6 +26,7 @@ fi IWYU_TOOL="${IWYU_TOOL:-/usr/bin/iwyu_tool}" FIX_INCLUDE="${FIX_INCLUDE:-/usr/bin/fix_include}" +IWYU_TOOL_DIR="${IWYU_TOOL_DIR:-tools_webrtc/iwyu}" COMPILE_COMMANDS='' error() { @@ -68,6 +69,7 @@ fi # IWYU has a confusing set of exit codes. Discard it. "$IWYU_TOOL" -p "$COMPILE_COMMANDS" "$FILE_CC" -- -Xiwyu --no_fwd_decls \ + -Xiwyu --mapping_file=../../$IWYU_TOOL_DIR/mappings.imp \ >& /tmp/includefixes$$ || echo "IWYU done, code $?" if grep 'fatal error' /tmp/includefixes$$; then diff --git a/third_party/libwebrtc/tools_webrtc/iwyu/iwyu-filter-list b/third_party/libwebrtc/tools_webrtc/iwyu/iwyu-filter-list index 2332cd3d6196..b5b0fb0721c5 100644 --- a/third_party/libwebrtc/tools_webrtc/iwyu/iwyu-filter-list +++ b/third_party/libwebrtc/tools_webrtc/iwyu/iwyu-filter-list @@ -6,3 +6,4 @@ #include #include <__memory/unique_ptr.h> #include <__tree> +#include diff --git a/third_party/libwebrtc/tools_webrtc/iwyu/mappings.imp b/third_party/libwebrtc/tools_webrtc/iwyu/mappings.imp new file mode 100644 index 000000000000..aa19c2b4c244 --- /dev/null +++ b/third_party/libwebrtc/tools_webrtc/iwyu/mappings.imp @@ -0,0 +1,32 @@ +# +# Mappings file for IWYU in webrtc +# +# Documentation of syntax: +# https://github.com/include-what-you-use/include-what-you-use/blob/master/docs/IWYUMappings.md +# +# Remember that it needs include strings INCLUDING <> or "" inside the quotes. +# +[ +# Redirect to have gmock and gtest includes under our control +{ include: ['"gmock/gmock.h"', "private", '"test/gmock.h"', "public"] }, +{ include: ['"gtest/gtest.h"', "private", '"test/gtest.h"', "public"] }, + +# rtc_base/containers internal defs +{ include: ['"rtc_base/containers/flat_tree.h"', "private", '"rtc_base/containers/flat_set.h"', "public"] }, + +# Revectoring of JSON +{ include: ['"json/reader.h"', "private", '"rtc_base/strings/json.h"', "public"] }, +{ include: ['"json/value.h"', "private", '"rtc_base/strings/json.h"', "public"] }, + +# LIBSRTP overrides +{ include: ['"rdbx.h"', "private", '"third_party/libsrtp/include/srtp_priv.h"', "public"] }, +{ include: ['"auth.h"', "private", '"third_party/libsrtp/include/srtp_priv.h"', "public"] }, + +# Needed to agree with presubmit tests for includes (and not include ) +{ symbol: ["std::string", "public", "", "public"] }, +{ symbol: ["std::move", "public", "", "public"] }, +{ symbol: ["std::make_unique", "public", "", "public"] }, +{ symbol: ["std::unique_ptr", "public", "", "public"] }, +# Needed to avoid +{ symbol: ["std::ostringstream", "public", "", "public"] }, +] \ No newline at end of file