commit 373162d5be83aff736c3c6288a76d964825934f0 parent c7d863ec9c992270acb1499febb2da6ed7dc6240 Author: Michael Froman <mfroman@mozilla.com> Date: Wed, 15 Oct 2025 11:45:12 -0500 Bug 1993083 - Vendor libwebrtc from 59927374f1 Upstream commit: https://webrtc.googlesource.com/src/+/59927374f16ef42cb57760da8a9d378a9037a60c IWYU everything again using git ls-files | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py followed by git cl format Bug: webrtc:42226242 Change-Id: I885a45b8fe48830c228b7d9c6fb02c32210347fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/402900 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#45264} Diffstat:
29 files changed, 76 insertions(+), 58 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor @@ -1,4 +1,4 @@ # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc -libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-15T16:43:59.892303+00:00. +libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-15T16:45:02.948456+00:00. # base of lastest vendoring -59f58760a5 +59927374f1 diff --git a/third_party/libwebrtc/api/test/metrics/metrics_set_proto_file_exporter_test.cc b/third_party/libwebrtc/api/test/metrics/metrics_set_proto_file_exporter_test.cc @@ -19,7 +19,6 @@ #include "api/test/metrics/metric.h" #include "api/test/metrics/proto/metric.pb.h" #include "api/units/timestamp.h" -#include "rtc_base/protobuf_utils.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/third_party/libwebrtc/moz-patch-stack/s0055.patch b/third_party/libwebrtc/moz-patch-stack/s0055.patch @@ -26,10 +26,10 @@ index 1896ed1d07..8704f0f34a 100644 deps += [ "logging:rtc_event_log_proto" ] } diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn -index 06d085e73a..2b3623e53f 100644 +index cd90d6d033..11c94ec229 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn -@@ -570,6 +570,7 @@ if (is_ios || is_mac) { +@@ -571,6 +571,7 @@ if (is_ios || is_mac) { } } @@ -37,7 +37,7 @@ index 06d085e73a..2b3623e53f 100644 rtc_library("videosource_objc") { sources = [ "objc/api/peerconnection/RTCVideoSource+Private.h", -@@ -599,6 +600,7 @@ if (is_ios || is_mac) { +@@ -600,6 +601,7 @@ if (is_ios || is_mac) { ":used_from_extension", ] } @@ -45,7 +45,7 @@ index 06d085e73a..2b3623e53f 100644 rtc_library("videoframebuffer_objc") { visibility = [ "*" ] -@@ -631,6 +633,7 @@ if (is_ios || is_mac) { +@@ -632,6 +634,7 @@ if (is_ios || is_mac) { ] } @@ -53,7 +53,7 @@ index 06d085e73a..2b3623e53f 100644 rtc_library("metal_objc") { visibility = [ "*" ] allow_poison = [ -@@ -692,6 +695,7 @@ if (is_ios || is_mac) { +@@ -693,6 +696,7 @@ if (is_ios || is_mac) { ":videoframebuffer_objc", ] } @@ -61,7 +61,7 @@ index 06d085e73a..2b3623e53f 100644 rtc_library("videocapture_objc") { visibility = [ "*" ] -@@ -720,6 +724,7 @@ if (is_ios || is_mac) { +@@ -721,6 +725,7 @@ if (is_ios || is_mac) { ] } @@ -69,7 +69,7 @@ index 06d085e73a..2b3623e53f 100644 rtc_library("videocodec_objc") { visibility = [ "*" ] configs += [ "..:no_global_constructors" ] -@@ -1819,5 +1824,6 @@ if (is_ios || is_mac) { +@@ -1820,5 +1825,6 @@ if (is_ios || is_mac) { "VideoToolbox.framework", ] } diff --git a/third_party/libwebrtc/moz-patch-stack/s0099.patch b/third_party/libwebrtc/moz-patch-stack/s0099.patch @@ -61,10 +61,10 @@ index 0aec2d94c7..83c1b21260 100644 } } diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn -index 2b3623e53f..b3941ea94d 100644 +index 11c94ec229..df32a6863d 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn -@@ -631,6 +631,20 @@ if (is_ios || is_mac) { +@@ -632,6 +632,20 @@ if (is_ios || is_mac) { "CoreGraphics.framework", "CoreVideo.framework", ] diff --git a/third_party/libwebrtc/moz-patch-stack/s0103.patch b/third_party/libwebrtc/moz-patch-stack/s0103.patch @@ -601,7 +601,7 @@ index 88655146af..70782e51b6 100644 import("../../webrtc.gni") diff --git a/pc/BUILD.gn b/pc/BUILD.gn -index ff47dc6ce7..2beb606167 100644 +index 5a064183f3..8120144fe2 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -30,8 +30,8 @@ @@ -702,7 +702,7 @@ index 4fa98d73a2..72c01d50ab 100644 output_extension = "so" } diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn -index b3941ea94d..eedf7ae7a5 100644 +index df32a6863d..c394964852 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn @@ -9,12 +9,12 @@ @@ -722,7 +722,7 @@ index b3941ea94d..eedf7ae7a5 100644 } group("sdk") { -@@ -66,7 +66,7 @@ if (is_ios || is_mac) { +@@ -67,7 +67,7 @@ if (is_ios || is_mac) { ] if (use_clang_coverage) { @@ -731,7 +731,7 @@ index b3941ea94d..eedf7ae7a5 100644 } } -@@ -1280,7 +1280,7 @@ if (is_ios || is_mac) { +@@ -1281,7 +1281,7 @@ if (is_ios || is_mac) { } public_deps = [ diff --git a/third_party/libwebrtc/pc/BUILD.gn b/third_party/libwebrtc/pc/BUILD.gn @@ -558,7 +558,15 @@ rtc_source_set("rtp_sender_proxy") { sources = [ "rtp_sender_proxy.h" ] deps = [ ":proxy", + "../api:dtls_transport_interface", + "../api:frame_transformer_interface", "../api:libjingle_peerconnection_api", + "../api:media_stream_interface", + "../api:rtc_error", + "../api:rtp_parameters", + "../api:scoped_refptr", + "../api/crypto:frame_encryptor_interface", + "../api/video_codecs:video_codecs_api", ] } @@ -605,6 +613,7 @@ rtc_source_set("rtp_transport_internal") { ":session_description", "../call:rtp_receiver", "../p2p:ice_transport_internal", + "../rtc_base:async_packet_socket", "../rtc_base:callback_list", "../rtc_base:copy_on_write_buffer", "../rtc_base:network_route", @@ -1373,7 +1382,12 @@ rtc_library("legacy_stats_collector") { rtc_source_set("stream_collection") { visibility = [ ":*" ] sources = [ "stream_collection.h" ] - deps = [ "../api:libjingle_peerconnection_api" ] + deps = [ + "../api:libjingle_peerconnection_api", + "../api:make_ref_counted", + "../api:media_stream_interface", + "../api:scoped_refptr", + ] } rtc_library("track_media_info_map") { visibility = [ ":*" ] @@ -1602,6 +1616,7 @@ rtc_library("peer_connection_message_handler") { "../api/task_queue:pending_task_safety_flag", "../api/units:time_delta", "../rtc_base:checks", + "../rtc_base:threading", ] } @@ -1717,6 +1732,7 @@ rtc_library("transceiver_list") { "../api:libjingle_peerconnection_api", "../api:rtc_error", "../api:rtp_parameters", + "../api:rtp_transceiver_direction", "../api:scoped_refptr", "../api:sequence_checker", "../rtc_base:checks", @@ -1868,7 +1884,10 @@ rtc_library("video_track") { rtc_source_set("sdp_state_provider") { visibility = [ ":*" ] sources = [ "sdp_state_provider.h" ] - deps = [ "../api:libjingle_peerconnection_api" ] + deps = [ + "../api:libjingle_peerconnection_api", + "../rtc_base:ssl_adapter", + ] } rtc_library("jitter_buffer_delay") { diff --git a/third_party/libwebrtc/pc/channel_interface.h b/third_party/libwebrtc/pc/channel_interface.h @@ -19,7 +19,6 @@ #include "api/jsep.h" #include "api/media_types.h" #include "media/base/media_channel.h" -#include "media/base/media_config.h" #include "media/base/stream_params.h" #include "pc/rtp_transport_internal.h" #include "pc/session_description.h" diff --git a/third_party/libwebrtc/pc/peer_connection_message_handler.h b/third_party/libwebrtc/pc/peer_connection_message_handler.h @@ -14,13 +14,13 @@ #include <functional> #include "api/jsep.h" -#include "api/legacy_stats_types.h" #include "api/media_stream_interface.h" #include "api/peer_connection_interface.h" #include "api/rtc_error.h" #include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" #include "pc/legacy_stats_collector_interface.h" +#include "rtc_base/thread.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/proxy.h b/third_party/libwebrtc/pc/proxy.h @@ -56,20 +56,13 @@ #ifndef PC_PROXY_H_ #define PC_PROXY_H_ -#include <stddef.h> - -#include <memory> -#include <string> +#include <cstddef> +#include <string> // IWYU pragma: keep #include <tuple> -#include <type_traits> #include <utility> -#include "api/make_ref_counted.h" -#include "api/scoped_refptr.h" -#include "api/task_queue/task_queue_base.h" #include "rtc_base/event.h" -#include "rtc_base/string_utils.h" -#include "rtc_base/system/rtc_export.h" +#include "rtc_base/string_utils.h" // IWYU pragma: keep #include "rtc_base/thread.h" #include "rtc_base/trace_event.h" diff --git a/third_party/libwebrtc/pc/remote_audio_source.h b/third_party/libwebrtc/pc/remote_audio_source.h @@ -15,7 +15,6 @@ #include <list> #include <optional> -#include <string> #include "api/call/audio_sink.h" #include "api/media_stream_interface.h" diff --git a/third_party/libwebrtc/pc/rtp_parameters_conversion.h b/third_party/libwebrtc/pc/rtp_parameters_conversion.h @@ -14,10 +14,8 @@ #include <optional> #include <vector> -#include "api/rtc_error.h" #include "api/rtp_parameters.h" #include "media/base/codec.h" -#include "media/base/stream_params.h" #include "pc/session_description.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/rtp_sender.h b/third_party/libwebrtc/pc/rtp_sender.h @@ -27,7 +27,6 @@ #include "api/dtls_transport_interface.h" #include "api/dtmf_sender_interface.h" #include "api/environment/environment.h" -#include "api/field_trials_view.h" #include "api/frame_transformer_interface.h" #include "api/media_stream_interface.h" #include "api/media_types.h" @@ -36,11 +35,12 @@ #include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" #include "api/sequence_checker.h" +#include "api/video_codecs/video_encoder_factory.h" #include "media/base/audio_source.h" +#include "media/base/codec.h" #include "media/base/media_channel.h" #include "pc/dtmf_sender.h" #include "pc/legacy_stats_collector_interface.h" -#include "rtc_base/checks.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" diff --git a/third_party/libwebrtc/pc/rtp_sender_proxy.h b/third_party/libwebrtc/pc/rtp_sender_proxy.h @@ -11,11 +11,22 @@ #ifndef PC_RTP_SENDER_PROXY_H_ #define PC_RTP_SENDER_PROXY_H_ +#include <cstdint> #include <memory> #include <string> #include <vector> +#include "api/crypto/frame_encryptor_interface.h" +#include "api/dtls_transport_interface.h" +#include "api/dtmf_sender_interface.h" +#include "api/frame_transformer_interface.h" +#include "api/media_stream_interface.h" +#include "api/media_types.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_codecs/video_encoder_factory.h" #include "pc/proxy.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/rtp_transport_internal.h b/third_party/libwebrtc/pc/rtp_transport_internal.h @@ -19,6 +19,7 @@ #include "absl/functional/any_invocable.h" #include "call/rtp_demuxer.h" #include "pc/session_description.h" +#include "rtc_base/async_packet_socket.h" #include "rtc_base/callback_list.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/network/sent_packet.h" diff --git a/third_party/libwebrtc/pc/sctp_data_channel.h b/third_party/libwebrtc/pc/sctp_data_channel.h @@ -11,13 +11,13 @@ #ifndef PC_SCTP_DATA_CHANNEL_H_ #define PC_SCTP_DATA_CHANNEL_H_ -#include <stdint.h> - +#include <cstddef> +#include <cstdint> #include <memory> #include <optional> -#include <set> #include <string> +#include "absl/functional/any_invocable.h" #include "api/data_channel_interface.h" #include "api/priority.h" #include "api/rtc_error.h" diff --git a/third_party/libwebrtc/pc/sctp_utils.h b/third_party/libwebrtc/pc/sctp_utils.h @@ -11,16 +11,14 @@ #ifndef PC_SCTP_UTILS_H_ #define PC_SCTP_UTILS_H_ +#include <cstdint> +#include <optional> #include <string> #include "api/data_channel_interface.h" #include "api/priority.h" -#include "api/transport/data_channel_transport_interface.h" -#include "media/base/media_channel.h" -#include "media/sctp/sctp_transport_internal.h" #include "net/dcsctp/public/types.h" #include "rtc_base/copy_on_write_buffer.h" -#include "rtc_base/ssl_stream_adapter.h" // For SSLRole namespace webrtc { class CopyOnWriteBuffer; diff --git a/third_party/libwebrtc/pc/sdp_state_provider.h b/third_party/libwebrtc/pc/sdp_state_provider.h @@ -11,10 +11,12 @@ #ifndef PC_SDP_STATE_PROVIDER_H_ #define PC_SDP_STATE_PROVIDER_H_ +#include <optional> #include <string> #include "api/jsep.h" #include "api/peer_connection_interface.h" +#include "rtc_base/ssl_stream_adapter.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/sdp_utils.h b/third_party/libwebrtc/pc/sdp_utils.h @@ -13,7 +13,6 @@ #include <functional> #include <memory> -#include <string> #include "api/jsep.h" #include "p2p/base/transport_info.h" diff --git a/third_party/libwebrtc/pc/stream_collection.h b/third_party/libwebrtc/pc/stream_collection.h @@ -11,11 +11,15 @@ #ifndef PC_STREAM_COLLECTION_H_ #define PC_STREAM_COLLECTION_H_ +#include <cstddef> #include <string> #include <utility> #include <vector> +#include "api/make_ref_counted.h" +#include "api/media_stream_interface.h" #include "api/peer_connection_interface.h" +#include "api/scoped_refptr.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/test/mock_channel_interface.h b/third_party/libwebrtc/pc/test/mock_channel_interface.h @@ -22,6 +22,7 @@ #include "media/base/stream_params.h" #include "pc/channel_interface.h" #include "pc/rtp_transport_internal.h" +#include "pc/session_description.h" #include "test/gmock.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/track_media_info_map.h b/third_party/libwebrtc/pc/track_media_info_map.h @@ -11,13 +11,9 @@ #ifndef PC_TRACK_MEDIA_INFO_MAP_H_ #define PC_TRACK_MEDIA_INFO_MAP_H_ -#include <stdint.h> - +#include <cstdint> #include <map> -#include <memory> #include <optional> -#include <string> -#include <vector> #include "api/array_view.h" #include "api/media_stream_interface.h" @@ -25,7 +21,7 @@ #include "media/base/media_channel.h" #include "pc/rtp_receiver.h" #include "pc/rtp_sender.h" -#include "rtc_base/ref_count.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/third_party/libwebrtc/pc/transceiver_list.h b/third_party/libwebrtc/pc/transceiver_list.h @@ -11,18 +11,16 @@ #ifndef PC_TRANSCEIVER_LIST_H_ #define PC_TRANSCEIVER_LIST_H_ -#include <stddef.h> - #include <algorithm> +#include <cstddef> #include <map> #include <optional> #include <string> #include <vector> -#include "api/media_types.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/scoped_refptr.h" #include "api/sequence_checker.h" #include "pc/rtp_transceiver.h" diff --git a/third_party/libwebrtc/pc/used_ids.h b/third_party/libwebrtc/pc/used_ids.h @@ -16,7 +16,6 @@ #include "api/rtp_parameters.h" #include "media/base/codec.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" namespace webrtc { template <typename IdStruct> diff --git a/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc b/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc @@ -19,6 +19,7 @@ #include "api/video/encoded_image.h" #include "api/video/recordable_encoded_frame.h" #include "api/video/video_codec_type.h" +#include "api/video/video_rotation.h" #include "api/video/video_sink_interface.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/third_party/libwebrtc/pc/video_track.h b/third_party/libwebrtc/pc/video_track.h @@ -11,9 +11,9 @@ #ifndef PC_VIDEO_TRACK_H_ #define PC_VIDEO_TRACK_H_ -#include <optional> #include <string> +#include "absl/strings/string_view.h" #include "api/media_stream_interface.h" #include "api/media_stream_track.h" #include "api/scoped_refptr.h" diff --git a/third_party/libwebrtc/pc/video_track_source.h b/third_party/libwebrtc/pc/video_track_source.h @@ -21,7 +21,6 @@ #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" diff --git a/third_party/libwebrtc/sdk/BUILD.gn b/third_party/libwebrtc/sdk/BUILD.gn @@ -46,6 +46,7 @@ rtc_library("sdk_tests") { sources = [ "media_constraints_unittest.cc" ] deps = [ ":media_constraints", + "../api:libjingle_peerconnection_api", "../test:test_support", ] } diff --git a/third_party/libwebrtc/sdk/media_constraints.h b/third_party/libwebrtc/sdk/media_constraints.h @@ -16,8 +16,8 @@ #ifndef SDK_MEDIA_CONSTRAINTS_H_ #define SDK_MEDIA_CONSTRAINTS_H_ -#include <stddef.h> - +#include <cstddef> +#include <initializer_list> #include <string> #include <utility> #include <vector> diff --git a/third_party/libwebrtc/sdk/media_constraints_unittest.cc b/third_party/libwebrtc/sdk/media_constraints_unittest.cc @@ -10,6 +10,7 @@ #include "sdk/media_constraints.h" +#include "api/peer_connection_interface.h" #include "test/gtest.h" namespace webrtc {