commit 63817f7edcb82cc7d876563f693b26891d08369f
parent 99ab3684ff82fe5a2d0d226818a489f500288d4c
Author: Dan Baker <dbaker@mozilla.com>
Date: Fri, 24 Oct 2025 12:29:51 -0600
Bug 1995393 - Vendor libwebrtc from 38bb331bda
Upstream commit: https://webrtc.googlesource.com/src/+/38bb331bdabb6f2528dab26613de7d7fccd0e859
Propagate negotiated header extension to mediachannel on answer
Propagates the subset of header extensions to use for both audio and
video media send/receive channels when an answer has been accepted.
Bug: webrtc:383078466
Change-Id: I791ba828114a480bac7bb324a56dd93f59316c39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/405241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45391}
Diffstat:
5 files changed, 164 insertions(+), 27 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 2025-10-24T18:27:33.031750+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-24T18:29:41.623893+00:00.
# base of lastest vendoring
-39dc944355
+38bb331bda
diff --git a/third_party/libwebrtc/pc/channel.cc b/third_party/libwebrtc/pc/channel.cc
@@ -938,6 +938,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
RtpHeaderExtensions header_extensions =
GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
bool update_header_extensions = true;
+ // TODO: issues.webrtc.org/383078466 - remove if pushdown on answer is enough.
media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
AudioReceiverParameters recv_params = last_recv_params_;
@@ -965,6 +966,19 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
last_recv_params_ = recv_params;
+ if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
+ AudioSenderParameter send_params = last_send_params_;
+ send_params.extensions = header_extensions;
+ send_params.extmap_allow_mixed = content->extmap_allow_mixed();
+ if (!media_send_channel()->SetSenderParameters(send_params)) {
+ error_desc = StringFormat(
+ "Failed to set send parameters for m-section with mid='%s'.",
+ mid().c_str());
+ return false;
+ }
+ last_send_params_ = send_params;
+ }
+
if (!UpdateLocalStreams_w(content->streams(), type, error_desc)) {
RTC_DCHECK(!error_desc.empty());
return false;
@@ -1009,6 +1023,18 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
mid().c_str());
return false;
}
+
+ if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
+ AudioReceiverParameters recv_params = last_recv_params_;
+ recv_params.extensions = send_params.extensions;
+ if (!media_receive_channel()->SetReceiverParameters(recv_params)) {
+ error_desc = StringFormat(
+ "Failed to set recv parameters for m-section with mid='%s'.",
+ mid().c_str());
+ return false;
+ }
+ last_recv_params_ = recv_params;
+ }
// The receive channel can send RTCP packets in the reverse direction. It
// should use the reduced size mode if a peer has requested it through the
// remote content.
@@ -1079,14 +1105,13 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string& error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
- RTC_DLOG(LS_INFO) << "Setting local video description for " << ToString();
RTC_LOG_THREAD_BLOCK_COUNT();
RtpHeaderExtensions header_extensions =
GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
bool update_header_extensions = true;
- // TODO: issues.webrtc.org/396640 - remove if pushdown on answer is enough.
+ // TODO: issues.webrtc.org/383078466 - remove if pushdown on answer is enough.
media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
VideoReceiverParameters recv_params = last_recv_params_;
@@ -1096,6 +1121,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
RtpTransceiverDirectionHasRecv(content->direction()), &recv_params);
VideoSenderParameters send_params = last_send_params_;
+ send_params.extensions = header_extensions;
send_params.extmap_allow_mixed = content->extmap_allow_mixed();
// Ensure that there is a matching packetization for each send codec. If the
diff --git a/third_party/libwebrtc/pc/channel_unittest.cc b/third_party/libwebrtc/pc/channel_unittest.cc
@@ -76,6 +76,7 @@ using ::webrtc::FakeVoiceMediaSendChannel;
using ::webrtc::FieldTrials;
using ::webrtc::RidDescription;
using ::webrtc::RidDirection;
+using ::webrtc::RtpExtension;
using ::webrtc::RtpTransceiverDirection;
using ::webrtc::SdpType;
using ::webrtc::StreamParams;
@@ -645,6 +646,65 @@ class ChannelTest : public ::testing::Test {
media_send_channel1_impl()->send_codecs()[0]);
}
+ void TestRemovesExtensionNotPresentInRemoteAnswer() {
+ typename T::Content local;
+ typename T::Content remote;
+ CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local);
+ CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &remote);
+ local.set_rtp_header_extensions({
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri, 0),
+ RtpExtension(RtpExtension::kVideoRotationUri, 1),
+ });
+ remote.set_rtp_header_extensions({
+ RtpExtension(RtpExtension::kVideoRotationUri, 1),
+ });
+
+ CreateChannels(0, 0);
+ std::string err;
+ ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kOffer, err))
+ << err;
+ ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kAnswer, err))
+ << err;
+
+ EXPECT_THAT(media_receive_channel1_impl()->recv_extensions(),
+ ElementsAre(AllOf(Field("id", &RtpExtension::id, 1),
+ Field("uri", &RtpExtension::uri,
+ RtpExtension::kVideoRotationUri))));
+ EXPECT_THAT(media_send_channel1_impl()->send_extensions(),
+ ElementsAre(AllOf(Field("id", &RtpExtension::id, 1),
+ Field("uri", &RtpExtension::uri,
+ RtpExtension::kVideoRotationUri))));
+ }
+ void TestRemovesExtensionNotPresentInLocalAnswer() {
+ typename T::Content local;
+ typename T::Content remote;
+ CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &local);
+ CreateContent(/*flags=*/0, kPcmuCodec, kH264Codec, &remote);
+ local.set_rtp_header_extensions({
+ RtpExtension(RtpExtension::kVideoRotationUri, 1),
+ });
+ remote.set_rtp_header_extensions({
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri, 0),
+ RtpExtension(RtpExtension::kVideoRotationUri, 1),
+ });
+
+ CreateChannels(0, 0);
+ std::string err;
+ ASSERT_TRUE(channel1_->SetRemoteContent(&remote, SdpType::kOffer, err))
+ << err;
+ ASSERT_TRUE(channel1_->SetLocalContent(&local, SdpType::kAnswer, err))
+ << err;
+
+ EXPECT_THAT(media_receive_channel1_impl()->recv_extensions(),
+ ElementsAre(AllOf(Field("id", &RtpExtension::id, 1),
+ Field("uri", &RtpExtension::uri,
+ RtpExtension::kVideoRotationUri))));
+ EXPECT_THAT(media_send_channel1_impl()->send_extensions(),
+ ElementsAre(AllOf(Field("id", &RtpExtension::id, 1),
+ Field("uri", &RtpExtension::uri,
+ RtpExtension::kVideoRotationUri))));
+ }
+
// Test that SetLocalContent and SetRemoteContent properly configure
// extmap-allow-mixed.
void TestSetContentsExtmapAllowMixedCaller(bool offer, bool answer) {
@@ -1877,6 +1937,14 @@ TEST_F(VoiceChannelSingleThreadTest, SocketOptionsMergedOnSetTransport) {
Base::SocketOptionsMergedOnSetTransport();
}
+TEST_F(VoiceChannelSingleThreadTest, RemovesExtensionNotPresentInRemoteAnswer) {
+ Base::TestRemovesExtensionNotPresentInRemoteAnswer();
+}
+
+TEST_F(VoiceChannelSingleThreadTest, RemovesExtensionNotPresentInLocalAnswer) {
+ Base::TestRemovesExtensionNotPresentInLocalAnswer();
+}
+
// VoiceChannelDoubleThreadTest
TEST_F(VoiceChannelDoubleThreadTest, TestInit) {
Base::TestInit();
@@ -2157,6 +2225,14 @@ TEST_F(VideoChannelSingleThreadTest, UpdateLocalStreamsWithSimulcast) {
Base::TestUpdateLocalStreamsWithSimulcast();
}
+TEST_F(VideoChannelSingleThreadTest, RemovesExtensionNotPresentInRemoteAnswer) {
+ Base::TestRemovesExtensionNotPresentInRemoteAnswer();
+}
+
+TEST_F(VideoChannelSingleThreadTest, RemovesExtensionNotPresentInLocalAnswer) {
+ Base::TestRemovesExtensionNotPresentInLocalAnswer();
+}
+
TEST_F(VideoChannelSingleThreadTest, TestSetLocalOfferWithPacketization) {
const webrtc::Codec kVp8Codec = webrtc::CreateVideoCodec(97, "VP8");
webrtc::Codec vp9_codec = webrtc::CreateVideoCodec(98, "VP9");
diff --git a/third_party/libwebrtc/pc/congestion_control_integrationtest.cc b/third_party/libwebrtc/pc/congestion_control_integrationtest.cc
@@ -178,17 +178,26 @@ TEST_F(PeerConnectionCongestionControlTest, NegotiatingCcfbRemovesTsn) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignalingForSdpOnly();
callee()->AddVideoTrack();
+ callee()->AddAudioTrack();
// Add transceivers to caller in order to accomodate reception
caller()->pc()->AddTransceiver(MediaType::VIDEO);
- auto parameters = caller()->pc()->GetSenders()[0]->GetParameters();
+ caller()->pc()->AddTransceiver(MediaType::AUDIO);
+
caller()->CreateAndSetAndSignalOffer();
ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()),
IsRtcOk());
- std::vector<RtpHeaderExtensionCapability> negotiated_header_extensions =
- caller()->pc()->GetTransceivers()[0]->GetNegotiatedHeaderExtensions();
+ ASSERT_THAT(caller()->pc()->GetTransceivers().size(), Eq(2));
EXPECT_THAT(
- negotiated_header_extensions,
+ caller()->pc()->GetTransceivers()[0]->GetNegotiatedHeaderExtensions(),
+ Not(Contains(
+ AllOf(Field("uri", &RtpHeaderExtensionCapability::uri,
+ RtpExtension::kTransportSequenceNumberUri),
+ Not(Field("direction", &RtpHeaderExtensionCapability::direction,
+ RtpTransceiverDirection::kStopped))))))
+ << " in caller negotiated header extensions";
+ EXPECT_THAT(
+ caller()->pc()->GetTransceivers()[1]->GetNegotiatedHeaderExtensions(),
Not(Contains(
AllOf(Field("uri", &RtpHeaderExtensionCapability::uri,
RtpExtension::kTransportSequenceNumberUri),
@@ -196,28 +205,51 @@ TEST_F(PeerConnectionCongestionControlTest, NegotiatingCcfbRemovesTsn) {
RtpTransceiverDirection::kStopped))))))
<< " in caller negotiated header extensions";
- parameters = caller()->pc()->GetSenders()[0]->GetParameters();
- EXPECT_THAT(parameters.header_extensions,
- Not(Contains(Field("uri", &RtpExtension::uri,
- RtpExtension::kTransportSequenceNumberUri))))
+ ASSERT_THAT(caller()->pc()->GetSenders().size(), Eq(2));
+ EXPECT_THAT(
+ caller()->pc()->GetSenders()[0]->GetParameters().header_extensions,
+ Not(Contains(Field("uri", &RtpExtension::uri,
+ RtpExtension::kTransportSequenceNumberUri))))
+ << " in caller sender parameters";
+ EXPECT_THAT(
+ caller()->pc()->GetSenders()[1]->GetParameters().header_extensions,
+ Not(Contains(Field("uri", &RtpExtension::uri,
+ RtpExtension::kTransportSequenceNumberUri))))
<< " in caller sender parameters";
- parameters = caller()->pc()->GetReceivers()[0]->GetParameters();
- EXPECT_THAT(parameters.header_extensions,
- Not(Contains(Field("uri", &RtpExtension::uri,
- RtpExtension::kTransportSequenceNumberUri))))
+ EXPECT_THAT(
+ caller()->pc()->GetReceivers()[0]->GetParameters().header_extensions,
+ Not(Contains(Field("uri", &RtpExtension::uri,
+ RtpExtension::kTransportSequenceNumberUri))))
+ << " in caller receiver parameters";
+ EXPECT_THAT(caller()->pc()->GetReceivers()[1]->media_type(),
+ Eq(MediaType::AUDIO));
+ EXPECT_THAT(
+ caller()->pc()->GetReceivers()[1]->GetParameters().header_extensions,
+ Not(Contains(Field("uri", &RtpExtension::uri,
+ RtpExtension::kTransportSequenceNumberUri))))
<< " in caller receiver parameters";
- /* Callee senders are not fixed yet.
- TODO: issues.webrtc.org/383078466 - enable
- parameters = callee()->pc()->GetSenders()[0]->GetParameters();
- EXPECT_THAT(parameters.header_extensions,
- Not(Contains(Field("uri", &RtpExtension::uri,
- RtpExtension::kTransportSequenceNumberUri))))
+
+ EXPECT_THAT(
+ callee()->pc()->GetSenders()[0]->GetParameters().header_extensions,
+ Not(Contains(Field("uri", &RtpExtension::uri,
+ RtpExtension::kTransportSequenceNumberUri))))
<< " in callee sender parameters";
- */
- parameters = callee()->pc()->GetReceivers()[0]->GetParameters();
- EXPECT_THAT(parameters.header_extensions,
- Not(Contains(Field("uri", &RtpExtension::uri,
- RtpExtension::kTransportSequenceNumberUri))))
+ EXPECT_THAT(
+ callee()->pc()->GetSenders()[1]->GetParameters().header_extensions,
+ Not(Contains(Field("uri", &RtpExtension::uri,
+ RtpExtension::kTransportSequenceNumberUri))))
+ << " in callee sender parameters";
+
+ ASSERT_THAT(callee()->pc()->GetReceivers().size(), Eq(2));
+ EXPECT_THAT(
+ callee()->pc()->GetReceivers()[0]->GetParameters().header_extensions,
+ Not(Contains(Field("uri", &RtpExtension::uri,
+ RtpExtension::kTransportSequenceNumberUri))))
+ << " in callee receiver parameters";
+ EXPECT_THAT(
+ callee()->pc()->GetReceivers()[1]->GetParameters().header_extensions,
+ Not(Contains(Field("uri", &RtpExtension::uri,
+ RtpExtension::kTransportSequenceNumberUri))))
<< " in callee receiver parameters";
}
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/l4s_test.cc b/third_party/libwebrtc/test/peer_scenario/tests/l4s_test.cc
@@ -14,6 +14,7 @@
#include <string>
#include "absl/strings/str_cat.h"
+#include "api/audio_options.h"
#include "api/jsep.h"
#include "api/make_ref_counted.h"
#include "api/rtc_error.h"
@@ -180,7 +181,9 @@ TEST(L4STest, NegotiateAndUseCcfbIfEnabled) {
PeerScenarioClient::VideoSendTrackConfig video_conf;
video_conf.generator.squares_video->framerate = 15;
+ caller->CreateAudio("AUDIO_1", AudioOptions());
caller->CreateVideo("VIDEO_1", video_conf);
+ callee->CreateAudio("AUDIO_2", AudioOptions());
callee->CreateVideo("VIDEO_2", video_conf);
signaling.StartIceSignaling();
