Bug 2029247 - Vendor libwebrtc from 97f5e77203

Upstream commit: https://webrtc.googlesource.com/src/+/97f5e77203ad311e34f926ef311e70497bc2e3a8
    Remove some unused private functions.

    This covers peer_connection.h and also fixes one in rtp_transceiver.h.
    These are likely leftovers from previous refactorings.

    Bug: None
    Change-Id: I8ad28d8c54085a8d95ef40c520b4901ae06135f6
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/442161
    Auto-Submit: Harald Alvestrand <hta@webrtc.org>
    Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
    Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#46665}

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 2026-04-03T23:25:53.661752+00:00.
+libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2026-04-03T23:26:56.831213+00:00.
 # base of lastest vendoring
-7d3af2ee95
+97f5e77203

third_party/libwebrtc/pc/peer_connection.cc

@@ -1233,11 +1233,6 @@ PeerConnection::AddTransceiver(MediaType media_type,
   return scoped_refptr<RtpTransceiverInterface>(transceiver);
 }
 
-void PeerConnection::OnNegotiationNeeded() {
-  RTC_DCHECK_RUN_ON(signaling_thread());
-  RTC_DCHECK(!IsClosed());
-  sdp_handler_->UpdateNegotiationNeeded();
-}
 
 scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
     const std::string& kind,
@@ -2429,21 +2424,6 @@ bool PeerConnection::GetSslRole(const std::string& content_name,
   return false;
 }
 
-bool PeerConnection::GetTransportDescription(
-    const SessionDescription* description,
-    const std::string& content_name,
-    TransportDescription* tdesc) {
-  if (!description || !tdesc) {
-    return false;
-  }
-  const TransportInfo* transport_info =
-      description->GetTransportInfoByName(content_name);
-  if (!transport_info) {
-    return false;
-  }
-  *tdesc = transport_info->description;
-  return true;
-}
 
 std::vector<DataChannelStats> PeerConnection::GetDataChannelStats() const {
   RTC_DCHECK_RUN_ON(network_thread());

third_party/libwebrtc/pc/peer_connection.h

@@ -67,7 +67,6 @@
 #include "p2p/base/ice_transport_internal.h"
 #include "p2p/base/port.h"
 #include "p2p/base/port_allocator.h"
-#include "p2p/base/transport_description.h"
 #include "p2p/dtls/dtls_transport_factory.h"
 #include "pc/channel_interface.h"
 #include "pc/codec_vendor.h"
@@ -526,8 +525,6 @@ class PeerConnection : public PeerConnectionInternal,
   void OnSelectedCandidatePairChanged(const CandidatePairChangeEvent& event)
       RTC_RUN_ON(signaling_thread());
 
-  void OnNegotiationNeeded();
-
   const JsepTransportController* transport_controller_s() const
       RTC_RUN_ON(signaling_thread()) {
     return transport_controller_copy_;
@@ -562,12 +559,6 @@ class PeerConnection : public PeerConnectionInternal,
   // This function should only be called from the worker thread.
   void StopRtcEventLog_w();
 
-  // Returns true and the TransportInfo of the given `content_name`
-  // from `description`. Returns false if it's not available.
-  static bool GetTransportDescription(const SessionDescription* description,
-                                      const std::string& content_name,
-                                      TransportDescription* info);
-
   // Returns the media index for a local ice candidate given the content name.
   // Returns false if the local session description does not have a media
   // content called  `content_name`.

third_party/libwebrtc/pc/rtp_transceiver.cc

@@ -439,7 +439,7 @@ RTCError RtpTransceiver::CreateChannel(
     absl::AnyInvocable<RtpTransportInternal*(absl::string_view) &&>
         transport_lookup) {
   RTC_DCHECK_RUN_ON(thread_);
-  RTC_DCHECK(!channel());
+  RTC_DCHECK(!channel_);
   RTC_DCHECK(!mid_ || mid_.value() == mid);
   RTC_DCHECK(!stopped_);
 

third_party/libwebrtc/pc/rtp_transceiver.h

@@ -374,10 +374,6 @@ class RtpTransceiver : public RtpTransceiverInterface {
   VoiceMediaReceiveChannelInterface* voice_media_receive_channel();
 
  private:
-  // Returns the Voice/VideoChannel set for this transceiver. May be null if
-  // the transceiver is not in the currently set local/remote description.
-  ChannelInterface* channel() const { return channel_.get(); }
-
   MediaEngineInterface* media_engine() RTC_RUN_ON(context()->worker_thread());
   ConnectionContext* context() const { return context_; }
   CodecVendor& codec_vendor() {