commit ebec12a46cd39406e07ff3d17e5d5829b00ff0fe parent 48498d0d20a3eb7d7bd85b23eb1fa437b6b4ab1a Author: Dan Baker <dbaker@mozilla.com> Date: Mon, 1 Dec 2025 23:12:32 -0700 Bug 2000941 - Vendor libwebrtc from 4ed70dbed2 Upstream commit: https://webrtc.googlesource.com/src/+/4ed70dbed2e2b44786b705eb5754dbd452c6a66a Rename `ecn_1` to `ect_1`. This change aligns the naming with the standard terminology for Explicit Congestion Notification codepoints from RFC 3168. Change-Id: I52f380b28c6d6711271c0c68ebe79cfe106171bb Bug: webrtc:42225697 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/410541 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Frederik Hermans <frederikh@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45709} Diffstat:
8 files changed, 15 insertions(+), 14 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 /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc -libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-12-02T06:09:57.361633+00:00. +libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-12-02T06:12:18.453329+00:00. # base of lastest vendoring -361404b5bc +4ed70dbed2 diff --git a/third_party/libwebrtc/media/base/media_channel_impl.cc b/third_party/libwebrtc/media/base/media_channel_impl.cc @@ -174,7 +174,7 @@ MediaChannelUtil::TransportForMediaChannels::TranslatePacketOptions( rtc_options.info_signaled_after_sent.included_in_allocation = options.included_in_allocation; rtc_options.info_signaled_after_sent.is_media = options.is_media; - rtc_options.ecn_1 = options.send_as_ect1; + rtc_options.ect_1 = options.send_as_ect1; rtc_options.batchable = options.batchable; rtc_options.last_packet_in_batch = options.last_packet_in_batch; return rtc_options; diff --git a/third_party/libwebrtc/moz-patch-stack/s0027.patch b/third_party/libwebrtc/moz-patch-stack/s0027.patch @@ -698,7 +698,7 @@ index cbec6fcfd7..15f4f39141 100644 class VideoMediaReceiveChannelInterface; class VoiceMediaSendChannelInterface; diff --git a/media/base/media_channel_impl.cc b/media/base/media_channel_impl.cc -index 6d4d4a2363..316408a9b3 100644 +index 2cb3560682..01fcd82dc8 100644 --- a/media/base/media_channel_impl.cc +++ b/media/base/media_channel_impl.cc @@ -35,15 +35,6 @@ diff --git a/third_party/libwebrtc/p2p/test/fake_packet_transport.h b/third_party/libwebrtc/p2p/test/fake_packet_transport.h @@ -138,7 +138,7 @@ class FakePacketTransport : public PacketTransportInternal { if (dest_) { dest_->NotifyPacketReceived(ReceivedIpPacket( packet, SocketAddress(), Timestamp::Micros(TimeMicros()), - options.ecn_1 ? EcnMarking::kEct1 : EcnMarking::kNotEct)); + options.ect_1 ? EcnMarking::kEct1 : EcnMarking::kNotEct)); } } diff --git a/third_party/libwebrtc/pc/rtp_transport_unittest.cc b/third_party/libwebrtc/pc/rtp_transport_unittest.cc @@ -328,7 +328,7 @@ TEST(RtpTransportTest, ReceivedPacketEcnMarkingPropagatedToDemuxedPacket) { transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer); AsyncSocketPacketOptions options; - options.ecn_1 = true; + options.ect_1 = true; const int flags = 0; Buffer rtp_data(kRtpData, kRtpLen); fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags); diff --git a/third_party/libwebrtc/rtc_base/async_packet_socket.h b/third_party/libwebrtc/rtc_base/async_packet_socket.h @@ -14,6 +14,7 @@ #include <cstddef> #include <cstdint> #include <functional> +#include <utility> #include <vector> #include "absl/functional/any_invocable.h" @@ -57,10 +58,10 @@ struct RTC_EXPORT AsyncSocketPacketOptions { DiffServCodePoint dscp = DSCP_NO_CHANGE; - // Packet will be sent with ECN(1), RFC-3168, Section 5. + // Packet will be sent with ECT(1), RFC-3168, Section 5. // Intended to be used with L4S // https://www.rfc-editor.org/rfc/rfc9331.html - bool ecn_1 = false; + bool ect_1 = false; // When used with RTP packets (for example, PacketOptions), the value // should be 16 bits. A value of -1 represents "not set". diff --git a/third_party/libwebrtc/rtc_base/async_udp_socket.cc b/third_party/libwebrtc/rtc_base/async_udp_socket.cc @@ -84,13 +84,13 @@ int AsyncUDPSocket::SendTo(const void* pv, env_.clock().TimeInMilliseconds(), options.info_signaled_after_sent); CopySocketInformationToPacketInfo(cb, *this, &sent_packet.info); - if (has_set_ect1_options_ != options.ecn_1) { + if (has_set_ect1_options_ != options.ect_1) { // It is unclear what is most efficient, setting options on every sent // packet or when changed. Potentially, can separate send sockets be used? // This is the easier implementation. if (socket_->SetOption(Socket::Option::OPT_SEND_ECN, - options.ecn_1 ? 1 : 0) == 0) { - has_set_ect1_options_ = options.ecn_1; + options.ect_1 ? 1 : 0) == 0) { + has_set_ect1_options_ = options.ect_1; } } int ret = socket_->SendTo(pv, cb, addr); diff --git a/third_party/libwebrtc/rtc_base/async_udp_socket_unittest.cc b/third_party/libwebrtc/rtc_base/async_udp_socket_unittest.cc @@ -38,17 +38,17 @@ TEST(AsyncUDPSocketTest, SetSocketOptionIfEctChange) { uint8_t buffer[] = "hello"; AsyncSocketPacketOptions packet_options; - packet_options.ecn_1 = false; + packet_options.ect_1 = false; udp_socket->SendTo(buffer, 5, kAddr, packet_options); udp_socket->GetOption(Socket::OPT_SEND_ECN, &ect); EXPECT_EQ(ect, 0); - packet_options.ecn_1 = true; + packet_options.ect_1 = true; udp_socket->SendTo(buffer, 5, kAddr, packet_options); udp_socket->GetOption(Socket::OPT_SEND_ECN, &ect); EXPECT_EQ(ect, 1); - packet_options.ecn_1 = false; + packet_options.ect_1 = false; udp_socket->SendTo(buffer, 5, kAddr, packet_options); udp_socket->GetOption(Socket::OPT_SEND_ECN, &ect); EXPECT_EQ(ect, 0);