commit d7044f62e19dd3df1a2c22fc1bb849f40b6f060e
parent 085820b380d18dd4a9b46fc6787389a3f8bbe4d5
Author: Dan Baker <dbaker@mozilla.com>
Date: Mon, 1 Dec 2025 19:32:10 -0700
Bug 2000941 - Vendor libwebrtc from bb6b3aab15
Essentially a no-op since we're going to see this change
reverted when we vendor in 29d6eabaf0.
Upstream commit: https://webrtc.googlesource.com/src/+/bb6b3aab159ebcd825526cf831f54170ccb2584d
Add instructions for removing a Trampoline-wrapped Signal
Also apply the instructions to one Signal.
Bug: webrtc:42222066
Change-Id: I63409d968d27a843b0ac7c61a122a1b685cb7a6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/408883
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45624}
Diffstat:
5 files changed, 495 insertions(+), 707 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 2025-12-02T02:29:10.180304+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-12-02T02:31:53.214353+00:00.
# base of lastest vendoring
-3cf5dc1bec
+bb6b3aab15
diff --git a/third_party/libwebrtc/moz-patch-stack/29d6eabaf0.no-op-cherry-pick-msg b/third_party/libwebrtc/moz-patch-stack/29d6eabaf0.no-op-cherry-pick-msg
@@ -0,0 +1 @@
+We already cherry-picked this when we vendored bb6b3aab15.
diff --git a/third_party/libwebrtc/moz-patch-stack/p0001.patch b/third_party/libwebrtc/moz-patch-stack/p0001.patch
@@ -1,425 +1,118 @@
From: Daniel Baker <dbaker@mozilla.com>
-Date: Mon, 1 Dec 2025 19:28:40 -0700
-Subject: (tmp-cherry-pick) Revert "Mark deprecated features with
- [[deprecated]]" (7e1a12d6ff)
+Date: Mon, 1 Dec 2025 19:31:22 -0700
+Subject: (tmp-cherry-pick) [M142] Revert "Add instructions for removing a
+ Trampoline-wrapped Signal" (29d6eabaf0)
-This reverts commit 3cf5dc1bece892a9c6e359bfa9931a88f896df12.
+This reverts commit bb6b3aab159ebcd825526cf831f54170ccb2584d.
-Reason for revert: Broke downstream project (Nearby)
+Reason for revert: Need to fix downstream Chromium issues.
Original change's description:
-> Mark deprecated features with [[deprecated]]
+> Add instructions for removing a Trampoline-wrapped Signal
>
-> This was the result of asking Gemini
-> "Consider the interfaces in @api. There are comments that indicate that certain methods are deprecated, but the methods don't have [[deprecated]] on them. Make a CL that adds [[deprecated]] to those methods"
+> Also apply the instructions to one Signal.
>
-> followed by a few rounds of reverting and working around errors.
->
-> Bug: None
-> Change-Id: I5d7b1f89fff01277ce96e940aa7340cdfcb728cc
-> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407441
+> Bug: webrtc:42222066
+> Change-Id: I63409d968d27a843b0ac7c61a122a1b685cb7a6f
+> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/408883
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
-> Cr-Commit-Position: refs/heads/main@{#45623}
+> Cr-Commit-Position: refs/heads/main@{#45624}
+
+(cherry picked from commit 716c8eb942a74ad641bbace8c1db521b4857c636)
-Bug: None
-No-Presubmit: true
-No-Tree-Checks: true
-No-Try: true
-Change-Id: Ib5c58e36e94b915a01514455c8a2ad561b61a5c9
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/409000
-Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
+Bug: webrtc:42222066, chromium:449225599, chromium:450516559
+No-IWYU: Revert
+Change-Id: I7fdf1694664b11ffcfd91916f1e98ef88254059d
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/413923
+Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
-Cr-Commit-Position: refs/heads/main@{#45629}
+Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
+Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
+Cr-Original-Commit-Position: refs/heads/main@{#45840}
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/415743
+Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
+Reviewed-by: Harald Alvestrand <hta@webrtc.org>
+Cr-Commit-Position: refs/branch-heads/7444@{#2}
+Cr-Branched-From: f70dc714a073397356f6ed866481da73f90f0b96-refs/heads/main@{#45766}
---
- api/audio/audio_processing_statistics.cc | 5 -----
- api/audio/audio_processing_statistics.h | 5 +----
- api/call/bitrate_allocation.h | 1 -
- api/candidate.cc | 5 -----
- api/candidate.h | 4 +++-
- api/data_channel_interface.h | 5 +----
- api/dtls_transport_interface.h | 15 +++++++--------
- api/dtmf_sender_interface.h | 3 +--
- api/jsep.h | 3 ++-
- api/peer_connection_interface.h | 4 ++--
- api/rtp_receiver_interface.h | 5 +++--
- api/rtp_sender_interface.h | 5 +++--
- api/uma_metrics.h | 2 +-
- .../audio_processing/audio_processing_unittest.cc | 3 ---
- .../include/audio_frame_proxies.cc | 4 ----
- pc/data_channel_unittest.cc | 3 ---
- pc/dtmf_sender.cc | 6 ------
- pc/dtmf_sender_unittest.cc | 2 ++
- pc/peer_connection_integrationtest.cc | 3 ---
- pc/sctp_data_channel.cc | 3 ---
- sdk/objc/unittests/RTCIceCandidateTest.mm | 3 ++-
- 21 files changed, 28 insertions(+), 61 deletions(-)
+ p2p/base/ice_transport_internal.cc | 5 +++--
+ p2p/base/ice_transport_internal.h | 9 ++++++---
+ rtc_base/sigslot_trampoline.h | 12 ------------
+ 3 files changed, 9 insertions(+), 17 deletions(-)
-diff --git a/api/audio/audio_processing_statistics.cc b/api/audio/audio_processing_statistics.cc
-index b6e92eaa12..90da7e8874 100644
---- a/api/audio/audio_processing_statistics.cc
-+++ b/api/audio/audio_processing_statistics.cc
-@@ -14,13 +14,8 @@ namespace webrtc {
-
- AudioProcessingStats::AudioProcessingStats() = default;
-
--// TODO: https://issues.webrtc.org/42221314 - remove pragma when deprecated
--// field `voice_detected` is removed.
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- AudioProcessingStats::AudioProcessingStats(const AudioProcessingStats& other) =
- default;
--#pragma clang diagnostic pop
-
- AudioProcessingStats::~AudioProcessingStats() = default;
-
-diff --git a/api/audio/audio_processing_statistics.h b/api/audio/audio_processing_statistics.h
-index 897af0c2c3..d6f8b6e61c 100644
---- a/api/audio/audio_processing_statistics.h
-+++ b/api/audio/audio_processing_statistics.h
-@@ -20,8 +20,6 @@
- namespace webrtc {
- // This version of the stats uses Optionals, it will replace the regular
- // AudioProcessingStatistics struct.
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- struct RTC_EXPORT AudioProcessingStats {
- AudioProcessingStats();
- AudioProcessingStats(const AudioProcessingStats& other);
-@@ -33,7 +31,7 @@ struct RTC_EXPORT AudioProcessingStats {
- // It is conservative in flagging audio as speech, with low likelihood of
- // incorrectly flagging a frame as voice.
- // Only reported if voice detection is enabled in AudioProcessing::Config.
-- [[deprecated("bugs.webrtc.org/11226")]] std::optional<bool> voice_detected;
-+ std::optional<bool> voice_detected;
-
- // AEC Statistics.
- // ERL = 10log_10(P_far / P_echo)
-@@ -64,7 +62,6 @@ struct RTC_EXPORT AudioProcessingStats {
- // call to `GetStatistics()`.
- std::optional<int32_t> delay_ms;
- };
--#pragma clang diagnostic pop
-
- } // namespace webrtc
-
-diff --git a/api/call/bitrate_allocation.h b/api/call/bitrate_allocation.h
-index 20f8b69b13..e0557504d6 100644
---- a/api/call/bitrate_allocation.h
-+++ b/api/call/bitrate_allocation.h
-@@ -28,7 +28,6 @@ struct BitrateAllocationUpdate {
- // Predicted round trip time.
- TimeDelta round_trip_time = TimeDelta::PlusInfinity();
- // `bwe_period` is deprecated.
-- // TODO: https://issues.webrtc.org/442860748 - complete removal.
- TimeDelta bwe_period = TimeDelta::PlusInfinity();
- // Congestion window pushback bitrate reduction fraction. Used in
- // VideoStreamEncoder to reduce the bitrate by the given fraction
-diff --git a/api/candidate.cc b/api/candidate.cc
-index a92f00e6ab..bb12f06a3b 100644
---- a/api/candidate.cc
-+++ b/api/candidate.cc
-@@ -195,11 +195,6 @@ bool Candidate::operator!=(const Candidate& o) const {
- return !(*this == o);
+diff --git a/p2p/base/ice_transport_internal.cc b/p2p/base/ice_transport_internal.cc
+index 7843a8b37d..a4b96db00b 100644
+--- a/p2p/base/ice_transport_internal.cc
++++ b/p2p/base/ice_transport_internal.cc
+@@ -236,7 +236,8 @@ RTCError IceConfig::IsValid() const {
}
--Candidate Candidate::ToSanitizedCopy(bool use_hostname_address,
-- bool filter_related_address) const {
-- return ToSanitizedCopy(use_hostname_address, filter_related_address, false);
--}
--
- Candidate Candidate::ToSanitizedCopy(bool use_hostname_address,
- bool filter_related_address,
- bool filter_ufrag) const {
-diff --git a/api/candidate.h b/api/candidate.h
-index 7e2b13838f..c6f16a50f7 100644
---- a/api/candidate.h
-+++ b/api/candidate.h
-@@ -236,7 +236,9 @@ class RTC_EXPORT Candidate {
- // any remote ice parameters have been set.
- [[deprecated("Use variant with filter_ufrag")]] Candidate ToSanitizedCopy(
- bool use_hostname_address,
-- bool filter_related_address) const;
-+ bool filter_related_address) const {
-+ return ToSanitizedCopy(use_hostname_address, filter_related_address, false);
-+ }
- Candidate ToSanitizedCopy(bool use_hostname_address,
- bool filter_related_address,
- bool filter_ufrag) const;
-diff --git a/api/data_channel_interface.h b/api/data_channel_interface.h
-index e9c0c78750..4a8299a0b7 100644
---- a/api/data_channel_interface.h
-+++ b/api/data_channel_interface.h
-@@ -32,12 +32,10 @@ namespace webrtc {
-
- // C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelinit
- // TODO(deadbeef): Use std::optional for the "-1 if unset" things.
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- struct DataChannelInit {
- // Deprecated. Reliability is assumed, and channel will be unreliable if
- // maxRetransmitTime or MaxRetransmits is set.
-- [[deprecated]] bool reliable = false;
-+ bool reliable = false;
-
- // True if ordered delivery is required.
- bool ordered = true;
-@@ -71,7 +69,6 @@ struct DataChannelInit {
- // https://w3c.github.io/webrtc-priority/#new-rtcdatachannelinit-member
- std::optional<PriorityValue> priority;
- };
--#pragma clang diagnostic pop
-
- // At the JavaScript level, data can be passed in as a string or a blob, so
- // this structure's `binary` flag tells whether the data should be interpreted
-diff --git a/api/dtls_transport_interface.h b/api/dtls_transport_interface.h
-index 3b11015bc0..79099f43a2 100644
---- a/api/dtls_transport_interface.h
-+++ b/api/dtls_transport_interface.h
-@@ -55,14 +55,13 @@ class RTC_EXPORT DtlsTransportInformation {
- std::optional<int> srtp_cipher_suite,
- std::unique_ptr<SSLCertChain> remote_ssl_certificates,
- std::optional<int> ssl_group_id);
-- [[deprecated("Use version with role parameter")]] ABSL_DEPRECATED(
-- "Use version with role parameter")
-- DtlsTransportInformation(
-- DtlsTransportState state,
-- std::optional<int> tls_version,
-- std::optional<int> ssl_cipher_suite,
-- std::optional<int> srtp_cipher_suite,
-- std::unique_ptr<SSLCertChain> remote_ssl_certificates);
-+ ABSL_DEPRECATED("Use version with role parameter")
-+ DtlsTransportInformation(
-+ DtlsTransportState state,
-+ std::optional<int> tls_version,
-+ std::optional<int> ssl_cipher_suite,
-+ std::optional<int> srtp_cipher_suite,
-+ std::unique_ptr<SSLCertChain> remote_ssl_certificates);
-
- // Copy and assign
- DtlsTransportInformation(const DtlsTransportInformation& c);
-diff --git a/api/dtmf_sender_interface.h b/api/dtmf_sender_interface.h
-index ba1948f430..3b184832c0 100644
---- a/api/dtmf_sender_interface.h
-+++ b/api/dtmf_sender_interface.h
-@@ -32,8 +32,7 @@ class DtmfSenderObserverInterface {
- // DEPRECATED: Older API without tone buffer.
- // TODO(bugs.webrtc.org/9725): Remove old API and default implementation
- // when old callers are gone.
-- [[deprecated("Use OnToneChange with tone_buffer")]] virtual void OnToneChange(
-- const std::string& /* tone */) {}
-+ virtual void OnToneChange(const std::string& /* tone */) {}
-
- protected:
- virtual ~DtmfSenderObserverInterface() = default;
-diff --git a/api/jsep.h b/api/jsep.h
-index df0ebdcc43..7495535e31 100644
---- a/api/jsep.h
-+++ b/api/jsep.h
-@@ -89,7 +89,8 @@ class RTC_EXPORT IceCandidate final {
- std::string ToString() const;
-
- // TODO: webrtc:406795492 - Deprecate and remove this method.
-- [[deprecated("Use ToString()")]] bool ToString(std::string* out) const {
-+ // [[deprecated("Use ToString()")]]
-+ bool ToString(std::string* out) const {
- if (!out)
- return false;
- *out = ToString();
-diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
-index aea35c2514..15942a40e1 100644
---- a/api/peer_connection_interface.h
-+++ b/api/peer_connection_interface.h
-@@ -1132,8 +1132,8 @@ class RTC_EXPORT PeerConnectionInterface : public RefCountInterface {
- // to the MID of the m= section that generated the candidate.
- // TODO(bugs.webrtc.org/8395): Use IceCandidate instead of
- // Candidate, which would avoid the transport_name oddity.
-- [[deprecated("Use IceCandidate version")]] virtual bool RemoveIceCandidates(
-- const std::vector<Candidate>& candidates);
-+ [[deprecated("Use IceCandidate version")]]
-+ virtual bool RemoveIceCandidates(const std::vector<Candidate>& candidates);
-
- // SetBitrate limits the bandwidth allocated for all RTP streams sent by
- // this PeerConnection. Other limitations might affect these limits and
-diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
-index 8e301ee575..bf87f463c3 100644
---- a/api/rtp_receiver_interface.h
-+++ b/api/rtp_receiver_interface.h
-@@ -116,8 +116,9 @@ class RTC_EXPORT RtpReceiverInterface : public RefCountInterface,
- // Sets a frame transformer between the depacketizer and the decoder to enable
- // client code to transform received frames according to their own processing
- // logic.
-- [[deprecated("Use SetFrameTransformer")]] virtual void
-- SetDepacketizerToDecoderFrameTransformer(
-+ // TODO: bugs.webrtc.org/15929 - add [[deprecated("Use SetFrameTransformer")]]
-+ // when usage in Chrome is removed
-+ virtual void SetDepacketizerToDecoderFrameTransformer(
- scoped_refptr<FrameTransformerInterface> frame_transformer) {
- SetFrameTransformer(std::move(frame_transformer));
- }
-diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
-index db04b9580c..2a7907dc4e 100644
---- a/api/rtp_sender_interface.h
-+++ b/api/rtp_sender_interface.h
-@@ -118,8 +118,9 @@ class RTC_EXPORT RtpSenderInterface : public RefCountInterface,
- // user. This can be used to update the state of the object.
- virtual scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const = 0;
-
-- [[deprecated("Use SetFrameTransformer")]] virtual void
-- SetEncoderToPacketizerFrameTransformer(
-+ // TODO: bugs.webrtc.org/15929 - add [[deprecated("Use SetFrameTransformer")]]
-+ // when usage in Chrome is removed
-+ virtual void SetEncoderToPacketizerFrameTransformer(
- scoped_refptr<FrameTransformerInterface> frame_transformer) {
- SetFrameTransformer(std::move(frame_transformer));
- }
-diff --git a/api/uma_metrics.h b/api/uma_metrics.h
-index 53511ef762..d7946885c7 100644
---- a/api/uma_metrics.h
-+++ b/api/uma_metrics.h
-@@ -48,7 +48,7 @@ enum PeerConnectionMetricsName {
- enum IceCandidatePairType {
- // HostHost is deprecated. It was replaced with the set of types at the bottom
- // to report private or public host IP address.
-- kIceCandidatePairHostHost [[deprecated]] = 0,
-+ kIceCandidatePairHostHost = 0,
- kIceCandidatePairHostSrflx = 1,
- kIceCandidatePairHostRelay = 2,
- kIceCandidatePairHostPrflx = 3,
-diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
-index 8a3651566a..ce668bad1f 100644
---- a/modules/audio_processing/audio_processing_unittest.cc
-+++ b/modules/audio_processing/audio_processing_unittest.cc
-@@ -2875,10 +2875,7 @@ TEST(ApmStatistics, DoNotReportVoiceDetectedStat) {
- StreamConfig(frame.sample_rate_hz, frame.num_channels()),
- frame.data.data()),
- 0);
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- EXPECT_FALSE(apm->GetStatistics().voice_detected.has_value());
--#pragma clang diagnostic pop
+ IceTransportInternal::IceTransportInternal()
+- : role_conflict_trampoline_(this),
++ : candidate_gathered_trampoline_(this),
++ role_conflict_trampoline_(this),
+ ice_transport_state_changed_trampoline_(this),
+ destroyed_trampoline_(this) {}
+
+@@ -265,7 +266,7 @@ void IceTransportInternal::RemoveGatheringStateCallback(
+ void IceTransportInternal::SubscribeCandidateGathered(
+ absl::AnyInvocable<void(IceTransportInternal*, const Candidate&)>
+ callback) {
+- candidate_gathered_callbacks_.AddReceiver(std::move(callback));
++ candidate_gathered_trampoline_.Subscribe(std::move(callback));
}
- TEST(ApmStatistics, GetStatisticsReportsNoEchoDetectorStatsWhenDisabled) {
-diff --git a/modules/audio_processing/include/audio_frame_proxies.cc b/modules/audio_processing/include/audio_frame_proxies.cc
-index 84523250a5..97d81e6b85 100644
---- a/modules/audio_processing/include/audio_frame_proxies.cc
-+++ b/modules/audio_processing/include/audio_frame_proxies.cc
-@@ -31,15 +31,11 @@ int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
-
- AudioProcessingStats stats = ap->GetStatistics();
-
-- // TODO: https://issues.webrtc.org/42221314 - remove when deleted
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- if (stats.voice_detected) {
- frame->vad_activity_ = *stats.voice_detected
- ? AudioFrame::VADActivity::kVadActive
- : AudioFrame::VADActivity::kVadPassive;
- }
--#pragma clang diagnostic pop
-
- return result;
- }
-diff --git a/pc/data_channel_unittest.cc b/pc/data_channel_unittest.cc
-index dca2d6ece9..603b7f91d6 100644
---- a/pc/data_channel_unittest.cc
-+++ b/pc/data_channel_unittest.cc
-@@ -164,11 +164,8 @@ TEST_F(SctpDataChannelTest, VerifyConfigurationGetters) {
- // Note that the `init_.reliable` field is deprecated, so we directly set
- // it here to match spec behavior for purposes of checking the `reliable()`
- // getter.
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- init_.reliable = (!init_.maxRetransmits && !init_.maxRetransmitTime);
- EXPECT_EQ(channel_->reliable(), init_.reliable);
--#pragma clang diagnostic pop
- EXPECT_EQ(channel_->ordered(), init_.ordered);
- EXPECT_EQ(channel_->negotiated(), init_.negotiated);
- EXPECT_EQ(channel_->priority(), PriorityValue(Priority::kLow));
-diff --git a/pc/dtmf_sender.cc b/pc/dtmf_sender.cc
-index 7890e58841..703fe9b459 100644
---- a/pc/dtmf_sender.cc
-+++ b/pc/dtmf_sender.cc
-@@ -191,10 +191,7 @@ void DtmfSender::DoInsertDtmf() {
- // Fire a “OnToneChange” event with an empty string and stop.
- if (observer_) {
- observer_->OnToneChange(std::string(), tones_);
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- observer_->OnToneChange(std::string());
--#pragma clang diagnostic pop
- }
- return;
- } else {
-@@ -232,10 +229,7 @@ void DtmfSender::DoInsertDtmf() {
- if (observer_) {
- observer_->OnToneChange(tones_.substr(first_tone_pos, 1),
- tones_.substr(first_tone_pos + 1));
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- observer_->OnToneChange(tones_.substr(first_tone_pos, 1));
--#pragma clang diagnostic pop
+ void IceTransportInternal::SubscribeRoleConflict(
+diff --git a/p2p/base/ice_transport_internal.h b/p2p/base/ice_transport_internal.h
+index 69b8214fe1..0e233df6a5 100644
+--- a/p2p/base/ice_transport_internal.h
++++ b/p2p/base/ice_transport_internal.h
+@@ -352,9 +352,11 @@ class RTC_EXPORT IceTransportInternal : public PacketTransportInternal {
+ void RemoveGatheringStateCallback(const void* removal_tag);
+
+ // Handles sending and receiving of candidates.
++ sigslot::signal2<IceTransportInternal*, const Candidate&>
++ SignalCandidateGathered;
+ void NotifyCandidateGathered(IceTransportInternal* transport,
+ const Candidate& candidate) {
+- candidate_gathered_callbacks_.Send(transport, candidate);
++ SignalCandidateGathered(transport, candidate);
}
-
- // Erase the unrecognized characters plus the tone that's just processed.
-diff --git a/pc/dtmf_sender_unittest.cc b/pc/dtmf_sender_unittest.cc
-index 9c62cc151e..deb18fe769 100644
---- a/pc/dtmf_sender_unittest.cc
-+++ b/pc/dtmf_sender_unittest.cc
-@@ -200,6 +200,8 @@ class DtmfSenderTest : public ::testing::Test {
- const std::vector<std::string>& tones = observer_->tones();
- // The observer will get an empty string at the end.
- EXPECT_EQ(tones_ref.size() + 1, tones.size());
-+ EXPECT_EQ(observer_->tones(),
-+ observer_->tones_from_single_argument_callback());
- EXPECT_TRUE(tones.back().empty());
- EXPECT_TRUE(observer_->tones_remaining().empty());
- std::string::const_iterator it_ref = tones_ref.begin();
-diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
-index 7a8d0860ff..2d5deb8435 100644
---- a/pc/peer_connection_integrationtest.cc
-+++ b/pc/peer_connection_integrationtest.cc
-@@ -2136,12 +2136,9 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) {
- EXPECT_METRIC_EQ(0, num_best_ipv6);
- }
-
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- EXPECT_METRIC_EQ(
- 0, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
- kIceCandidatePairHostHost));
--#pragma clang diagnostic pop
- EXPECT_METRIC_EQ(
- 1, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
- kIceCandidatePairHostPublicHostPublic));
-diff --git a/pc/sctp_data_channel.cc b/pc/sctp_data_channel.cc
-index 18a8e8ab6b..7cca341408 100644
---- a/pc/sctp_data_channel.cc
-+++ b/pc/sctp_data_channel.cc
-@@ -84,11 +84,8 @@ BYPASS_PROXY_METHOD2(void,
- END_PROXY_MAP(DataChannel)
- } // namespace
-
--#pragma clang diagnostic push
--#pragma clang diagnostic ignored "-Wdeprecated-declarations"
- InternalDataChannelInit::InternalDataChannelInit(const DataChannelInit& base)
- : DataChannelInit(base), open_handshake_role(kOpener) {
--#pragma clang diagnostic pop
- // If the channel is externally negotiated, do not send the OPEN message.
- if (base.negotiated) {
- open_handshake_role = kNone;
-diff --git a/sdk/objc/unittests/RTCIceCandidateTest.mm b/sdk/objc/unittests/RTCIceCandidateTest.mm
-index 268d330815..c000be5a1b 100644
---- a/sdk/objc/unittests/RTCIceCandidateTest.mm
-+++ b/sdk/objc/unittests/RTCIceCandidateTest.mm
-@@ -39,7 +39,8 @@
- EXPECT_EQ("audio", nativeCandidate->sdp_mid());
- EXPECT_EQ(0, nativeCandidate->sdp_mline_index());
-
-- std::string sdpString = nativeCandidate->ToString();
-+ std::string sdpString;
-+ nativeCandidate->ToString(&sdpString);
- EXPECT_EQ(sdp.stdString, sdpString);
- }
-
+ void SubscribeCandidateGathered(
+ absl::AnyInvocable<void(IceTransportInternal*, const Candidate&)>
+@@ -464,8 +466,9 @@ class RTC_EXPORT IceTransportInternal : public PacketTransportInternal {
+ candidate_pair_change_callback_;
+
+ private:
+- CallbackList<IceTransportInternal*, const Candidate&>
+- candidate_gathered_callbacks_;
++ SignalTrampoline<IceTransportInternal,
++ &IceTransportInternal::SignalCandidateGathered>
++ candidate_gathered_trampoline_;
+ SignalTrampoline<IceTransportInternal,
+ &IceTransportInternal::SignalRoleConflict>
+ role_conflict_trampoline_;
+diff --git a/rtc_base/sigslot_trampoline.h b/rtc_base/sigslot_trampoline.h
+index a399e4177f..590e73d97b 100644
+--- a/rtc_base/sigslot_trampoline.h
++++ b/rtc_base/sigslot_trampoline.h
+@@ -51,18 +51,6 @@ namespace webrtc {
+ // my_class_object.SubscibeMyNamedEvent(
+ // SafeInvocable(target.safety_flag_.flag(),
+ // [target] { target.function(); }
+-// WHEN REMOVING THE SIGNAL
+-// Make a CL that will:
+-// - Delete the SignalMyNamedEvent signal
+-// - Change my_named_event_trampoline_ to my_named_event_callbacks,
+-// of type CallbackList<function arguments>
+-// - Change SubscribeMyNamedEvent to call my_named_event_callbacks.AddReceiver
+-// - Change NotifyMyNamedEvent to call my_named_event_callbacks_.Send
+-// - If UnsubscribeMyNamedEvent exists, change that also.
+-// - Delete the initialization of my_named_event_trampoline_ from the
+-// relevant constructor.
+-// Sending this through the bots will flush out remaining references to
+-// SignalMyNamedEvent.
+ namespace internal {
+ template <typename MemberPtrT>
+ struct member_pointer_traits;
diff --git a/third_party/libwebrtc/moz-patch-stack/p0002.patch b/third_party/libwebrtc/moz-patch-stack/p0002.patch
@@ -1,331 +1,425 @@
-From: Palak Agarwal <agpalak@google.com>
-Date: Tue, 7 Oct 2025 13:45:52 +0200
-Subject: (cherry-pick-branch-heads/7390) [M141] Add logging while creating
- {Window|Screen}Capturer
+From: Daniel Baker <dbaker@mozilla.com>
+Date: Mon, 1 Dec 2025 19:28:40 -0700
+Subject: (tmp-cherry-pick) Revert "Mark deprecated features with
+ [[deprecated]]" (7e1a12d6ff)
-(cherry picked from commit 6908505ae0eba5d530ad0bb4b37d4654a4f36c95)
+This reverts commit 3cf5dc1bece892a9c6e359bfa9931a88f896df12.
-Fixed: chromium:450184498
-Bug: chromium:448881311
-Change-Id: I0f043f58bf831e6822451cac99a8972a054ffdf7
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/414800
-Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
-Reviewed-by: Harald Alvestrand <hta@webrtc.org>
-Commit-Queue: Palak Agarwal <agpalak@google.com>
-Cr-Original-Commit-Position: refs/heads/main@{#45856}
-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/415420
-Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
-Cr-Commit-Position: refs/branch-heads/7390@{#3}
-Cr-Branched-From: 2f553bf8d573c90176d51559b76dfc836996c8f5-refs/heads/main@{#45520}
+Reason for revert: Broke downstream project (Nearby)
+
+Original change's description:
+> Mark deprecated features with [[deprecated]]
+>
+> This was the result of asking Gemini
+> "Consider the interfaces in @api. There are comments that indicate that certain methods are deprecated, but the methods don't have [[deprecated]] on them. Make a CL that adds [[deprecated]] to those methods"
+>
+> followed by a few rounds of reverting and working around errors.
+>
+> Bug: None
+> Change-Id: I5d7b1f89fff01277ce96e940aa7340cdfcb728cc
+> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407441
+> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
+> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
+> Cr-Commit-Position: refs/heads/main@{#45623}
+
+Bug: None
+No-Presubmit: true
+No-Tree-Checks: true
+No-Try: true
+Change-Id: Ib5c58e36e94b915a01514455c8a2ad561b61a5c9
+Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/409000
+Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
+Commit-Queue: Harald Alvestrand <hta@webrtc.org>
+Cr-Commit-Position: refs/heads/main@{#45629}
---
- modules/desktop_capture/desktop_capturer.cc | 15 +++++++++++++++
- .../linux/x11/screen_capturer_x11.cc | 6 ++++++
- .../linux/x11/window_capturer_x11.cc | 4 ++++
- modules/desktop_capture/screen_capturer_darwin.mm | 7 +++++++
- .../desktop_capture/screen_capturer_fuchsia.cc | 3 +++
- modules/desktop_capture/screen_capturer_linux.cc | 4 ++++
- modules/desktop_capture/screen_capturer_null.cc | 4 ++++
- modules/desktop_capture/screen_capturer_win.cc | 6 ++++++
- modules/desktop_capture/window_capturer_linux.cc | 4 ++++
- modules/desktop_capture/window_capturer_mac.mm | 2 ++
- modules/desktop_capture/window_capturer_null.cc | 3 +++
- modules/desktop_capture/window_capturer_win.cc | 7 +++++++
- 12 files changed, 65 insertions(+)
+ api/audio/audio_processing_statistics.cc | 5 -----
+ api/audio/audio_processing_statistics.h | 5 +----
+ api/call/bitrate_allocation.h | 1 -
+ api/candidate.cc | 5 -----
+ api/candidate.h | 4 +++-
+ api/data_channel_interface.h | 5 +----
+ api/dtls_transport_interface.h | 15 +++++++--------
+ api/dtmf_sender_interface.h | 3 +--
+ api/jsep.h | 3 ++-
+ api/peer_connection_interface.h | 4 ++--
+ api/rtp_receiver_interface.h | 5 +++--
+ api/rtp_sender_interface.h | 5 +++--
+ api/uma_metrics.h | 2 +-
+ .../audio_processing/audio_processing_unittest.cc | 3 ---
+ .../include/audio_frame_proxies.cc | 4 ----
+ pc/data_channel_unittest.cc | 3 ---
+ pc/dtmf_sender.cc | 6 ------
+ pc/dtmf_sender_unittest.cc | 2 ++
+ pc/peer_connection_integrationtest.cc | 3 ---
+ pc/sctp_data_channel.cc | 3 ---
+ sdk/objc/unittests/RTCIceCandidateTest.mm | 3 ++-
+ 21 files changed, 28 insertions(+), 61 deletions(-)
-diff --git a/modules/desktop_capture/desktop_capturer.cc b/modules/desktop_capture/desktop_capturer.cc
-index ae9aba26ee..bf63f73178 100644
---- a/modules/desktop_capture/desktop_capturer.cc
-+++ b/modules/desktop_capture/desktop_capturer.cc
-@@ -21,6 +21,7 @@
- #include "modules/desktop_capture/desktop_capturer_differ_wrapper.h"
- #include "modules/desktop_capture/desktop_geometry.h"
- #include "modules/desktop_capture/shared_memory.h"
-+#include "rtc_base/logging.h"
- #include "system_wrappers/include/metrics.h"
+diff --git a/api/audio/audio_processing_statistics.cc b/api/audio/audio_processing_statistics.cc
+index b6e92eaa12..90da7e8874 100644
+--- a/api/audio/audio_processing_statistics.cc
++++ b/api/audio/audio_processing_statistics.cc
+@@ -14,13 +14,8 @@ namespace webrtc {
- #if defined(WEBRTC_WIN)
-@@ -81,18 +82,26 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateWindowCapturer(
- #if defined(RTC_ENABLE_WIN_WGC)
- if (options.allow_wgc_window_capturer() &&
- IsWgcSupported(CaptureType::kWindow)) {
-+ RTC_LOG(LS_INFO) << "video capture: DesktopCapturer::CreateWindowCapturer "
-+ "creates DesktopCapturer of type WgcCapturerWin";
- return WgcCapturerWin::CreateRawWindowCapturer(options);
- }
- #endif // defined(RTC_ENABLE_WIN_WGC)
+ AudioProcessingStats::AudioProcessingStats() = default;
- #if defined(WEBRTC_WIN)
- if (options.allow_cropping_window_capturer()) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateWindowCapturer "
-+ "creates DesktopCapturer of type CroppingWindowCapturerWin";
- return CroppingWindowCapturer::CreateCapturer(options);
- }
- #endif // defined(WEBRTC_WIN)
+-// TODO: https://issues.webrtc.org/42221314 - remove pragma when deprecated
+-// field `voice_detected` is removed.
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ AudioProcessingStats::AudioProcessingStats(const AudioProcessingStats& other) =
+ default;
+-#pragma clang diagnostic pop
- std::unique_ptr<DesktopCapturer> capturer = CreateRawWindowCapturer(options);
- if (capturer && options.detect_updated_region()) {
-+ RTC_LOG(LS_INFO) << "video capture: DesktopCapturer::CreateWindowCapturer "
-+ "creates DesktopCapturer of type "
-+ "DesktopCapturerDifferWrapper over a base capturer";
- capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
- }
-
-@@ -105,12 +114,18 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
- #if defined(RTC_ENABLE_WIN_WGC)
- if (options.allow_wgc_screen_capturer() &&
- IsWgcSupported(CaptureType::kScreen)) {
-+ RTC_LOG(LS_INFO) << "video capture: DesktopCapturer::CreateScreenCapturer "
-+ "creates DesktopCapturer of type WgcCapturerWin";
- return WgcCapturerWin::CreateRawScreenCapturer(options);
- }
- #endif // defined(RTC_ENABLE_WIN_WGC)
+ AudioProcessingStats::~AudioProcessingStats() = default;
- std::unique_ptr<DesktopCapturer> capturer = CreateRawScreenCapturer(options);
- if (capturer && options.detect_updated_region()) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateScreenCapturer creates "
-+ "DesktopCapturer of type DesktopCapturerDifferWrapper over a base "
-+ "capturer";
- capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
- }
+diff --git a/api/audio/audio_processing_statistics.h b/api/audio/audio_processing_statistics.h
+index 897af0c2c3..d6f8b6e61c 100644
+--- a/api/audio/audio_processing_statistics.h
++++ b/api/audio/audio_processing_statistics.h
+@@ -20,8 +20,6 @@
+ namespace webrtc {
+ // This version of the stats uses Optionals, it will replace the regular
+ // AudioProcessingStatistics struct.
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ struct RTC_EXPORT AudioProcessingStats {
+ AudioProcessingStats();
+ AudioProcessingStats(const AudioProcessingStats& other);
+@@ -33,7 +31,7 @@ struct RTC_EXPORT AudioProcessingStats {
+ // It is conservative in flagging audio as speech, with low likelihood of
+ // incorrectly flagging a frame as voice.
+ // Only reported if voice detection is enabled in AudioProcessing::Config.
+- [[deprecated("bugs.webrtc.org/11226")]] std::optional<bool> voice_detected;
++ std::optional<bool> voice_detected;
-diff --git a/modules/desktop_capture/linux/x11/screen_capturer_x11.cc b/modules/desktop_capture/linux/x11/screen_capturer_x11.cc
-index 90dd797cb3..f10dcabc7e 100644
---- a/modules/desktop_capture/linux/x11/screen_capturer_x11.cc
-+++ b/modules/desktop_capture/linux/x11/screen_capturer_x11.cc
-@@ -511,8 +511,14 @@ std::unique_ptr<DesktopCapturer> ScreenCapturerX11::CreateRawScreenCapturer(
- if (!options.x_display())
- return nullptr;
+ // AEC Statistics.
+ // ERL = 10log_10(P_far / P_echo)
+@@ -64,7 +62,6 @@ struct RTC_EXPORT AudioProcessingStats {
+ // call to `GetStatistics()`.
+ std::optional<int32_t> delay_ms;
+ };
+-#pragma clang diagnostic pop
-+ RTC_LOG(LS_INFO)
-+ << "video capture: ScreenCapturerX11::CreateRawScreenCapturer creates "
-+ "DesktopCapturer of type ScreenCapturerX11";
- std::unique_ptr<ScreenCapturerX11> capturer(new ScreenCapturerX11());
- if (!capturer->Init(options)) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: ScreenCapturerX11::CreateRawScreenCapturer "
-+ "DesktopCapturer is null because it can not be initiated";
- return nullptr;
- }
+ } // namespace webrtc
-diff --git a/modules/desktop_capture/linux/x11/window_capturer_x11.cc b/modules/desktop_capture/linux/x11/window_capturer_x11.cc
-index 8e592ff2d9..9505205b76 100644
---- a/modules/desktop_capture/linux/x11/window_capturer_x11.cc
-+++ b/modules/desktop_capture/linux/x11/window_capturer_x11.cc
-@@ -247,6 +247,10 @@ std::unique_ptr<DesktopCapturer> WindowCapturerX11::CreateRawWindowCapturer(
- const DesktopCaptureOptions& options) {
- if (!options.x_display())
- return nullptr;
-+
-+ RTC_LOG(LS_INFO)
-+ << "video capture: WindowCapturerX11::CreateRawWindowCapturer creates "
-+ "DesktopCapturer of type WindowCapturerX11";
- return std::unique_ptr<DesktopCapturer>(new WindowCapturerX11(options));
+diff --git a/api/call/bitrate_allocation.h b/api/call/bitrate_allocation.h
+index 20f8b69b13..e0557504d6 100644
+--- a/api/call/bitrate_allocation.h
++++ b/api/call/bitrate_allocation.h
+@@ -28,7 +28,6 @@ struct BitrateAllocationUpdate {
+ // Predicted round trip time.
+ TimeDelta round_trip_time = TimeDelta::PlusInfinity();
+ // `bwe_period` is deprecated.
+- // TODO: https://issues.webrtc.org/442860748 - complete removal.
+ TimeDelta bwe_period = TimeDelta::PlusInfinity();
+ // Congestion window pushback bitrate reduction fraction. Used in
+ // VideoStreamEncoder to reduce the bitrate by the given fraction
+diff --git a/api/candidate.cc b/api/candidate.cc
+index a92f00e6ab..bb12f06a3b 100644
+--- a/api/candidate.cc
++++ b/api/candidate.cc
+@@ -195,11 +195,6 @@ bool Candidate::operator!=(const Candidate& o) const {
+ return !(*this == o);
}
-diff --git a/modules/desktop_capture/screen_capturer_darwin.mm b/modules/desktop_capture/screen_capturer_darwin.mm
-index 95a877c45d..3f14ddfe58 100644
---- a/modules/desktop_capture/screen_capturer_darwin.mm
-+++ b/modules/desktop_capture/screen_capturer_darwin.mm
-@@ -12,6 +12,7 @@
+-Candidate Candidate::ToSanitizedCopy(bool use_hostname_address,
+- bool filter_related_address) const {
+- return ToSanitizedCopy(use_hostname_address, filter_related_address, false);
+-}
+-
+ Candidate Candidate::ToSanitizedCopy(bool use_hostname_address,
+ bool filter_related_address,
+ bool filter_ufrag) const {
+diff --git a/api/candidate.h b/api/candidate.h
+index 7e2b13838f..c6f16a50f7 100644
+--- a/api/candidate.h
++++ b/api/candidate.h
+@@ -236,7 +236,9 @@ class RTC_EXPORT Candidate {
+ // any remote ice parameters have been set.
+ [[deprecated("Use variant with filter_ufrag")]] Candidate ToSanitizedCopy(
+ bool use_hostname_address,
+- bool filter_related_address) const;
++ bool filter_related_address) const {
++ return ToSanitizedCopy(use_hostname_address, filter_related_address, false);
++ }
+ Candidate ToSanitizedCopy(bool use_hostname_address,
+ bool filter_related_address,
+ bool filter_ufrag) const;
+diff --git a/api/data_channel_interface.h b/api/data_channel_interface.h
+index e9c0c78750..4a8299a0b7 100644
+--- a/api/data_channel_interface.h
++++ b/api/data_channel_interface.h
+@@ -32,12 +32,10 @@ namespace webrtc {
- #include "modules/desktop_capture/mac/screen_capturer_mac.h"
- #include "modules/desktop_capture/mac/screen_capturer_sck.h"
-+#include "rtc_base/logging.h"
+ // C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelinit
+ // TODO(deadbeef): Use std::optional for the "-1 if unset" things.
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ struct DataChannelInit {
+ // Deprecated. Reliability is assumed, and channel will be unreliable if
+ // maxRetransmitTime or MaxRetransmits is set.
+- [[deprecated]] bool reliable = false;
++ bool reliable = false;
- namespace webrtc {
-
-@@ -27,10 +28,16 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
- std::unique_ptr<DesktopCapturer> sck_capturer =
- CreateScreenCapturerSck(options);
- if (sck_capturer) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateRawScreenCapturer creates "
-+ "DesktopCapturer of type ScreenCapturerSck";
- return sck_capturer;
- }
- }
+ // True if ordered delivery is required.
+ bool ordered = true;
+@@ -71,7 +69,6 @@ struct DataChannelInit {
+ // https://w3c.github.io/webrtc-priority/#new-rtcdatachannelinit-member
+ std::optional<PriorityValue> priority;
+ };
+-#pragma clang diagnostic pop
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateRawScreenCapturer creates "
-+ "DesktopCapturer of type ScreenCapturerMac";
- auto capturer =
- std::make_unique<ScreenCapturerMac>(options.configuration_monitor(),
- options.detect_updated_region(),
-diff --git a/modules/desktop_capture/screen_capturer_fuchsia.cc b/modules/desktop_capture/screen_capturer_fuchsia.cc
-index c3f51ef0fc..2a6c5e753d 100644
---- a/modules/desktop_capture/screen_capturer_fuchsia.cc
-+++ b/modules/desktop_capture/screen_capturer_fuchsia.cc
-@@ -56,6 +56,9 @@ size_t RoundUpToMultiple(size_t value, size_t multiple) {
+ // At the JavaScript level, data can be passed in as a string or a blob, so
+ // this structure's `binary` flag tells whether the data should be interpreted
+diff --git a/api/dtls_transport_interface.h b/api/dtls_transport_interface.h
+index 3b11015bc0..79099f43a2 100644
+--- a/api/dtls_transport_interface.h
++++ b/api/dtls_transport_interface.h
+@@ -55,14 +55,13 @@ class RTC_EXPORT DtlsTransportInformation {
+ std::optional<int> srtp_cipher_suite,
+ std::unique_ptr<SSLCertChain> remote_ssl_certificates,
+ std::optional<int> ssl_group_id);
+- [[deprecated("Use version with role parameter")]] ABSL_DEPRECATED(
+- "Use version with role parameter")
+- DtlsTransportInformation(
+- DtlsTransportState state,
+- std::optional<int> tls_version,
+- std::optional<int> ssl_cipher_suite,
+- std::optional<int> srtp_cipher_suite,
+- std::unique_ptr<SSLCertChain> remote_ssl_certificates);
++ ABSL_DEPRECATED("Use version with role parameter")
++ DtlsTransportInformation(
++ DtlsTransportState state,
++ std::optional<int> tls_version,
++ std::optional<int> ssl_cipher_suite,
++ std::optional<int> srtp_cipher_suite,
++ std::unique_ptr<SSLCertChain> remote_ssl_certificates);
- std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
- const DesktopCaptureOptions& options) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateRawScreenCapturer creates "
-+ "DesktopCapturer of type ScreenCapturerFuchsia";
- std::unique_ptr<ScreenCapturerFuchsia> capturer(new ScreenCapturerFuchsia());
- return capturer;
- }
-diff --git a/modules/desktop_capture/screen_capturer_linux.cc b/modules/desktop_capture/screen_capturer_linux.cc
-index 94726750c5..f25e08fb59 100644
---- a/modules/desktop_capture/screen_capturer_linux.cc
-+++ b/modules/desktop_capture/screen_capturer_linux.cc
-@@ -13,6 +13,7 @@
- #include "modules/desktop_capture/desktop_capture_options.h"
- #include "modules/desktop_capture/desktop_capture_types.h"
- #include "modules/desktop_capture/desktop_capturer.h"
-+#include "rtc_base/logging.h"
+ // Copy and assign
+ DtlsTransportInformation(const DtlsTransportInformation& c);
+diff --git a/api/dtmf_sender_interface.h b/api/dtmf_sender_interface.h
+index ba1948f430..3b184832c0 100644
+--- a/api/dtmf_sender_interface.h
++++ b/api/dtmf_sender_interface.h
+@@ -32,8 +32,7 @@ class DtmfSenderObserverInterface {
+ // DEPRECATED: Older API without tone buffer.
+ // TODO(bugs.webrtc.org/9725): Remove old API and default implementation
+ // when old callers are gone.
+- [[deprecated("Use OnToneChange with tone_buffer")]] virtual void OnToneChange(
+- const std::string& /* tone */) {}
++ virtual void OnToneChange(const std::string& /* tone */) {}
- #if defined(WEBRTC_USE_PIPEWIRE)
- #include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
-@@ -29,6 +30,9 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
- const DesktopCaptureOptions& options) {
- #if defined(WEBRTC_USE_PIPEWIRE)
- if (options.allow_pipewire() && BaseCapturerPipeWire::IsSupported()) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateRawScreenCapturer creates "
-+ "DesktopCapturer of type BaseCapturerPipeWire";
- return std::make_unique<BaseCapturerPipeWire>(options,
- CaptureType::kScreen);
- }
-diff --git a/modules/desktop_capture/screen_capturer_null.cc b/modules/desktop_capture/screen_capturer_null.cc
-index aa6d4991a0..21b2f46c4d 100644
---- a/modules/desktop_capture/screen_capturer_null.cc
-+++ b/modules/desktop_capture/screen_capturer_null.cc
-@@ -11,12 +11,16 @@
- #include <memory>
+ protected:
+ virtual ~DtmfSenderObserverInterface() = default;
+diff --git a/api/jsep.h b/api/jsep.h
+index df0ebdcc43..7495535e31 100644
+--- a/api/jsep.h
++++ b/api/jsep.h
+@@ -89,7 +89,8 @@ class RTC_EXPORT IceCandidate final {
+ std::string ToString() const;
- #include "modules/desktop_capture/desktop_capturer.h"
-+#include "rtc_base/logging.h"
+ // TODO: webrtc:406795492 - Deprecate and remove this method.
+- [[deprecated("Use ToString()")]] bool ToString(std::string* out) const {
++ // [[deprecated("Use ToString()")]]
++ bool ToString(std::string* out) const {
+ if (!out)
+ return false;
+ *out = ToString();
+diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
+index aea35c2514..15942a40e1 100644
+--- a/api/peer_connection_interface.h
++++ b/api/peer_connection_interface.h
+@@ -1132,8 +1132,8 @@ class RTC_EXPORT PeerConnectionInterface : public RefCountInterface {
+ // to the MID of the m= section that generated the candidate.
+ // TODO(bugs.webrtc.org/8395): Use IceCandidate instead of
+ // Candidate, which would avoid the transport_name oddity.
+- [[deprecated("Use IceCandidate version")]] virtual bool RemoveIceCandidates(
+- const std::vector<Candidate>& candidates);
++ [[deprecated("Use IceCandidate version")]]
++ virtual bool RemoveIceCandidates(const std::vector<Candidate>& candidates);
- namespace webrtc {
+ // SetBitrate limits the bandwidth allocated for all RTP streams sent by
+ // this PeerConnection. Other limitations might affect these limits and
+diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
+index 8e301ee575..bf87f463c3 100644
+--- a/api/rtp_receiver_interface.h
++++ b/api/rtp_receiver_interface.h
+@@ -116,8 +116,9 @@ class RTC_EXPORT RtpReceiverInterface : public RefCountInterface,
+ // Sets a frame transformer between the depacketizer and the decoder to enable
+ // client code to transform received frames according to their own processing
+ // logic.
+- [[deprecated("Use SetFrameTransformer")]] virtual void
+- SetDepacketizerToDecoderFrameTransformer(
++ // TODO: bugs.webrtc.org/15929 - add [[deprecated("Use SetFrameTransformer")]]
++ // when usage in Chrome is removed
++ virtual void SetDepacketizerToDecoderFrameTransformer(
+ scoped_refptr<FrameTransformerInterface> frame_transformer) {
+ SetFrameTransformer(std::move(frame_transformer));
+ }
+diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
+index db04b9580c..2a7907dc4e 100644
+--- a/api/rtp_sender_interface.h
++++ b/api/rtp_sender_interface.h
+@@ -118,8 +118,9 @@ class RTC_EXPORT RtpSenderInterface : public RefCountInterface,
+ // user. This can be used to update the state of the object.
+ virtual scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const = 0;
- // static
- std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
- const DesktopCaptureOptions& options) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateRawScreenCapturer creates null "
-+ "DesktopCapturer";
- return nullptr;
+- [[deprecated("Use SetFrameTransformer")]] virtual void
+- SetEncoderToPacketizerFrameTransformer(
++ // TODO: bugs.webrtc.org/15929 - add [[deprecated("Use SetFrameTransformer")]]
++ // when usage in Chrome is removed
++ virtual void SetEncoderToPacketizerFrameTransformer(
+ scoped_refptr<FrameTransformerInterface> frame_transformer) {
+ SetFrameTransformer(std::move(frame_transformer));
+ }
+diff --git a/api/uma_metrics.h b/api/uma_metrics.h
+index 53511ef762..d7946885c7 100644
+--- a/api/uma_metrics.h
++++ b/api/uma_metrics.h
+@@ -48,7 +48,7 @@ enum PeerConnectionMetricsName {
+ enum IceCandidatePairType {
+ // HostHost is deprecated. It was replaced with the set of types at the bottom
+ // to report private or public host IP address.
+- kIceCandidatePairHostHost [[deprecated]] = 0,
++ kIceCandidatePairHostHost = 0,
+ kIceCandidatePairHostSrflx = 1,
+ kIceCandidatePairHostRelay = 2,
+ kIceCandidatePairHostPrflx = 3,
+diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
+index 8a3651566a..ce668bad1f 100644
+--- a/modules/audio_processing/audio_processing_unittest.cc
++++ b/modules/audio_processing/audio_processing_unittest.cc
+@@ -2875,10 +2875,7 @@ TEST(ApmStatistics, DoNotReportVoiceDetectedStat) {
+ StreamConfig(frame.sample_rate_hz, frame.num_channels()),
+ frame.data.data()),
+ 0);
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ EXPECT_FALSE(apm->GetStatistics().voice_detected.has_value());
+-#pragma clang diagnostic pop
}
-diff --git a/modules/desktop_capture/screen_capturer_win.cc b/modules/desktop_capture/screen_capturer_win.cc
-index a5f2052979..6620888f18 100644
---- a/modules/desktop_capture/screen_capturer_win.cc
-+++ b/modules/desktop_capture/screen_capturer_win.cc
-@@ -19,6 +19,7 @@
- #include "modules/desktop_capture/win/dxgi_duplicator_controller.h"
- #include "modules/desktop_capture/win/screen_capturer_win_directx.h"
- #include "modules/desktop_capture/win/screen_capturer_win_gdi.h"
-+#include "rtc_base/logging.h"
-
- namespace webrtc {
+ TEST(ApmStatistics, GetStatisticsReportsNoEchoDetectorStatsWhenDisabled) {
+diff --git a/modules/audio_processing/include/audio_frame_proxies.cc b/modules/audio_processing/include/audio_frame_proxies.cc
+index 84523250a5..97d81e6b85 100644
+--- a/modules/audio_processing/include/audio_frame_proxies.cc
++++ b/modules/audio_processing/include/audio_frame_proxies.cc
+@@ -31,15 +31,11 @@ int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
-@@ -26,6 +27,9 @@ namespace {
+ AudioProcessingStats stats = ap->GetStatistics();
- std::unique_ptr<DesktopCapturer> CreateScreenCapturerWinDirectx(
- const DesktopCaptureOptions& options) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateRawScreenCapturer creates "
-+ "DesktopCapturer of type ScreenCapturerWinDirectx";
- std::unique_ptr<DesktopCapturer> capturer(
- new ScreenCapturerWinDirectx(options));
- capturer.reset(new BlankDetectorDesktopCapturerWrapper(
-@@ -39,6 +43,8 @@ std::unique_ptr<DesktopCapturer> CreateScreenCapturerWinDirectx(
- std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
- const DesktopCaptureOptions& options) {
- // Default capturer if no options are enabled is GDI.
-+ RTC_LOG(LS_INFO) << "video capture: DesktopCapturer::CreateRawScreenCapturer "
-+ "creates DesktopCapturer of type ScreenCapturerWinGdi";
- std::unique_ptr<DesktopCapturer> capturer(new ScreenCapturerWinGdi(options));
+- // TODO: https://issues.webrtc.org/42221314 - remove when deleted
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ if (stats.voice_detected) {
+ frame->vad_activity_ = *stats.voice_detected
+ ? AudioFrame::VADActivity::kVadActive
+ : AudioFrame::VADActivity::kVadPassive;
+ }
+-#pragma clang diagnostic pop
- // If DirectX is enabled use it as main capturer with GDI as fallback.
-diff --git a/modules/desktop_capture/window_capturer_linux.cc b/modules/desktop_capture/window_capturer_linux.cc
-index f621a63e72..87ea3d5721 100644
---- a/modules/desktop_capture/window_capturer_linux.cc
-+++ b/modules/desktop_capture/window_capturer_linux.cc
-@@ -13,6 +13,7 @@
- #include "modules/desktop_capture/desktop_capture_options.h"
- #include "modules/desktop_capture/desktop_capture_types.h"
- #include "modules/desktop_capture/desktop_capturer.h"
-+#include "rtc_base/logging.h"
+ return result;
+ }
+diff --git a/pc/data_channel_unittest.cc b/pc/data_channel_unittest.cc
+index dca2d6ece9..603b7f91d6 100644
+--- a/pc/data_channel_unittest.cc
++++ b/pc/data_channel_unittest.cc
+@@ -164,11 +164,8 @@ TEST_F(SctpDataChannelTest, VerifyConfigurationGetters) {
+ // Note that the `init_.reliable` field is deprecated, so we directly set
+ // it here to match spec behavior for purposes of checking the `reliable()`
+ // getter.
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ init_.reliable = (!init_.maxRetransmits && !init_.maxRetransmitTime);
+ EXPECT_EQ(channel_->reliable(), init_.reliable);
+-#pragma clang diagnostic pop
+ EXPECT_EQ(channel_->ordered(), init_.ordered);
+ EXPECT_EQ(channel_->negotiated(), init_.negotiated);
+ EXPECT_EQ(channel_->priority(), PriorityValue(Priority::kLow));
+diff --git a/pc/dtmf_sender.cc b/pc/dtmf_sender.cc
+index 7890e58841..703fe9b459 100644
+--- a/pc/dtmf_sender.cc
++++ b/pc/dtmf_sender.cc
+@@ -191,10 +191,7 @@ void DtmfSender::DoInsertDtmf() {
+ // Fire a “OnToneChange” event with an empty string and stop.
+ if (observer_) {
+ observer_->OnToneChange(std::string(), tones_);
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ observer_->OnToneChange(std::string());
+-#pragma clang diagnostic pop
+ }
+ return;
+ } else {
+@@ -232,10 +229,7 @@ void DtmfSender::DoInsertDtmf() {
+ if (observer_) {
+ observer_->OnToneChange(tones_.substr(first_tone_pos, 1),
+ tones_.substr(first_tone_pos + 1));
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ observer_->OnToneChange(tones_.substr(first_tone_pos, 1));
+-#pragma clang diagnostic pop
+ }
- #if defined(WEBRTC_USE_PIPEWIRE)
- #include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
-@@ -29,6 +30,9 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
- const DesktopCaptureOptions& options) {
- #if defined(WEBRTC_USE_PIPEWIRE)
- if (options.allow_pipewire() && BaseCapturerPipeWire::IsSupported()) {
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateRawWindowCapturer creates "
-+ "DesktopCapturer of type BaseCapturerPipeWire";
- return std::make_unique<BaseCapturerPipeWire>(options,
- CaptureType::kWindow);
+ // Erase the unrecognized characters plus the tone that's just processed.
+diff --git a/pc/dtmf_sender_unittest.cc b/pc/dtmf_sender_unittest.cc
+index 9c62cc151e..deb18fe769 100644
+--- a/pc/dtmf_sender_unittest.cc
++++ b/pc/dtmf_sender_unittest.cc
+@@ -200,6 +200,8 @@ class DtmfSenderTest : public ::testing::Test {
+ const std::vector<std::string>& tones = observer_->tones();
+ // The observer will get an empty string at the end.
+ EXPECT_EQ(tones_ref.size() + 1, tones.size());
++ EXPECT_EQ(observer_->tones(),
++ observer_->tones_from_single_argument_callback());
+ EXPECT_TRUE(tones.back().empty());
+ EXPECT_TRUE(observer_->tones_remaining().empty());
+ std::string::const_iterator it_ref = tones_ref.begin();
+diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
+index 7a8d0860ff..2d5deb8435 100644
+--- a/pc/peer_connection_integrationtest.cc
++++ b/pc/peer_connection_integrationtest.cc
+@@ -2136,12 +2136,9 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) {
+ EXPECT_METRIC_EQ(0, num_best_ipv6);
}
-diff --git a/modules/desktop_capture/window_capturer_mac.mm b/modules/desktop_capture/window_capturer_mac.mm
-index a1e6157b87..731090d3fb 100644
---- a/modules/desktop_capture/window_capturer_mac.mm
-+++ b/modules/desktop_capture/window_capturer_mac.mm
-@@ -217,6 +217,8 @@ void WindowCapturerMac::CaptureFrame() {
- // static
- std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
- const DesktopCaptureOptions& options) {
-+ RTC_LOG(LS_INFO) << "video capture: DesktopCapturer::CreateRawWindowCapturer "
-+ "creates DesktopCapturer of type WindowCapturerMac";
- return std::unique_ptr<DesktopCapturer>(new WindowCapturerMac(
- options.full_screen_window_detector(), options.configuration_monitor()));
- }
-diff --git a/modules/desktop_capture/window_capturer_null.cc b/modules/desktop_capture/window_capturer_null.cc
-index 2061dc5248..384e186d20 100644
---- a/modules/desktop_capture/window_capturer_null.cc
-+++ b/modules/desktop_capture/window_capturer_null.cc
-@@ -12,6 +12,7 @@
- #include "modules/desktop_capture/desktop_capturer.h"
- #include "rtc_base/checks.h"
-+#include "rtc_base/logging.h"
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ EXPECT_METRIC_EQ(
+ 0, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
+ kIceCandidatePairHostHost));
+-#pragma clang diagnostic pop
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
+ kIceCandidatePairHostPublicHostPublic));
+diff --git a/pc/sctp_data_channel.cc b/pc/sctp_data_channel.cc
+index 18a8e8ab6b..7cca341408 100644
+--- a/pc/sctp_data_channel.cc
++++ b/pc/sctp_data_channel.cc
+@@ -84,11 +84,8 @@ BYPASS_PROXY_METHOD2(void,
+ END_PROXY_MAP(DataChannel)
+ } // namespace
- namespace webrtc {
+-#pragma clang diagnostic push
+-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+ InternalDataChannelInit::InternalDataChannelInit(const DataChannelInit& base)
+ : DataChannelInit(base), open_handshake_role(kOpener) {
+-#pragma clang diagnostic pop
+ // If the channel is externally negotiated, do not send the OPEN message.
+ if (base.negotiated) {
+ open_handshake_role = kNone;
+diff --git a/sdk/objc/unittests/RTCIceCandidateTest.mm b/sdk/objc/unittests/RTCIceCandidateTest.mm
+index 268d330815..c000be5a1b 100644
+--- a/sdk/objc/unittests/RTCIceCandidateTest.mm
++++ b/sdk/objc/unittests/RTCIceCandidateTest.mm
+@@ -39,7 +39,8 @@
+ EXPECT_EQ("audio", nativeCandidate->sdp_mid());
+ EXPECT_EQ(0, nativeCandidate->sdp_mline_index());
-@@ -65,6 +66,8 @@ void WindowCapturerNull::CaptureFrame() {
- // static
- std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
- const DesktopCaptureOptions& options) {
-+ RTC_LOG(LS_INFO) << "video capture: DesktopCapturer::CreateRawWindowCapturer "
-+ "creates DesktopCapturer of type WindowCapturerNull";
- return std::unique_ptr<DesktopCapturer>(new WindowCapturerNull());
+- std::string sdpString = nativeCandidate->ToString();
++ std::string sdpString;
++ nativeCandidate->ToString(&sdpString);
+ EXPECT_EQ(sdp.stdString, sdpString);
}
-diff --git a/modules/desktop_capture/window_capturer_win.cc b/modules/desktop_capture/window_capturer_win.cc
-index 115bc7d9b0..15ed7f5121 100644
---- a/modules/desktop_capture/window_capturer_win.cc
-+++ b/modules/desktop_capture/window_capturer_win.cc
-@@ -15,6 +15,7 @@
- #include "modules/desktop_capture/desktop_capturer.h"
- #include "modules/desktop_capture/rgba_color.h"
- #include "modules/desktop_capture/win/window_capturer_win_gdi.h"
-+#include "rtc_base/logging.h"
-
- #if defined(RTC_ENABLE_WIN_WGC)
- #include "modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h"
-@@ -28,6 +29,8 @@ namespace webrtc {
- // static
- std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
- const DesktopCaptureOptions& options) {
-+ RTC_LOG(LS_INFO) << "video capture: DesktopCapturer::CreateRawWindowCapturer "
-+ "creates DesktopCapturer of type WindowCapturerWinGdi";
- std::unique_ptr<DesktopCapturer> capturer(
- WindowCapturerWinGdi::CreateRawWindowCapturer(options));
- #if defined(RTC_ENABLE_WIN_WGC)
-@@ -36,6 +39,10 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
- // BlankDectector capturer will send an error when it detects a failed
- // GDI rendering, then Fallback capturer will try to capture it again with
- // WGC.
-+ RTC_LOG(LS_INFO)
-+ << "video capture: DesktopCapturer::CreateRawWindowCapturer creates "
-+ "DesktopCapturer of type FallbackDesktopCapturerWrapper which has a "
-+ "fallback capturer of type WgcCapturerWin";
- capturer = std::make_unique<BlankDetectorDesktopCapturerWrapper>(
- std::move(capturer), RgbaColor(0, 0, 0, 0),
- /*check_per_capture*/ true);
diff --git a/third_party/libwebrtc/moz-patch-stack/p0002.patch b/third_party/libwebrtc/moz-patch-stack/p0003.patch
