Bug 1860685 - Vendor libwebrtc from ff281aa328

Upstream commit: https://webrtc.googlesource.com/src/+/ff281aa3280bc57247a7da96c9bb3711b92a4f5d
    Convert signals in rtp_transport_internal.h to CallbackList

    Bug: webrtc:11943
    Change-Id: I8e0839363712d9d8b49c2f6cbdb5f3ac59d79219
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318882
    Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
    Commit-Queue: Harald Alvestrand <hta@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#40700}

third_party/libwebrtc/README.moz-ff-commit

@@ -25806,3 +25806,6 @@ cbaf91bcf0
 # MOZ_LIBWEBRTC_SRC=/home/bcampen/checkouts/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 96de4d63e3
+# MOZ_LIBWEBRTC_SRC=/home/bcampen/checkouts/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+ff281aa328

third_party/libwebrtc/README.mozilla

@@ -17228,3 +17228,5 @@ libwebrtc updated from /home/bcampen/checkouts/elm/.moz-fast-forward/moz-libwebr
 libwebrtc updated from /home/bcampen/checkouts/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-10-27T21:05:37.244589.
 # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/bcampen/checkouts/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
 libwebrtc updated from /home/bcampen/checkouts/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-10-27T21:06:33.438675.
+# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/bcampen/checkouts/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
+libwebrtc updated from /home/bcampen/checkouts/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-10-27T21:07:26.514345.

third_party/libwebrtc/pc/BUILD.gn

@@ -475,9 +475,9 @@ rtc_source_set("rtp_transport_internal") {
     ":session_description",
     "../call:rtp_receiver",
     "../p2p:rtc_p2p",
+    "../rtc_base:callback_list",
     "../rtc_base:network_route",
     "../rtc_base:ssl",
-    "../rtc_base/third_party/sigslot",
   ]
 }
 

third_party/libwebrtc/pc/channel.cc

@@ -166,14 +166,17 @@ bool BaseChannel::ConnectToRtpTransport_n() {
   if (!rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this)) {
     return false;
   }
-  rtp_transport_->SignalReadyToSend.connect(
-      this, &BaseChannel::OnTransportReadyToSend);
-  rtp_transport_->SignalNetworkRouteChanged.connect(
-      this, &BaseChannel::OnNetworkRouteChanged);
-  rtp_transport_->SignalWritableState.connect(this,
-                                              &BaseChannel::OnWritableState);
-  rtp_transport_->SignalSentPacket.connect(this,
-                                           &BaseChannel::SignalSentPacket_n);
+  rtp_transport_->SubscribeReadyToSend(
+      this, [this](bool ready) { OnTransportReadyToSend(ready); });
+  rtp_transport_->SubscribeNetworkRouteChanged(
+      this, [this](absl::optional<rtc::NetworkRoute> route) {
+        OnNetworkRouteChanged(route);
+      });
+  rtp_transport_->SubscribeWritableState(
+      this, [this](bool state) { OnWritableState(state); });
+  rtp_transport_->SubscribeSentPacket(
+      this,
+      [this](const rtc::SentPacket& packet) { SignalSentPacket_n(packet); });
   return true;
 }
 
@@ -181,10 +184,10 @@ void BaseChannel::DisconnectFromRtpTransport_n() {
   RTC_DCHECK(rtp_transport_);
   RTC_DCHECK(media_send_channel());
   rtp_transport_->UnregisterRtpDemuxerSink(this);
-  rtp_transport_->SignalReadyToSend.disconnect(this);
-  rtp_transport_->SignalNetworkRouteChanged.disconnect(this);
-  rtp_transport_->SignalWritableState.disconnect(this);
-  rtp_transport_->SignalSentPacket.disconnect(this);
+  rtp_transport_->UnsubscribeReadyToSend(this);
+  rtp_transport_->UnsubscribeNetworkRouteChanged(this);
+  rtp_transport_->UnsubscribeWritableState(this);
+  rtp_transport_->UnsubscribeSentPacket(this);
   rtp_transport_ = nullptr;
   media_send_channel()->SetInterface(nullptr);
   media_receive_channel()->SetInterface(nullptr);

third_party/libwebrtc/pc/channel.h

@@ -45,7 +45,6 @@
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/socket.h"
-#include "rtc_base/third_party/sigslot/sigslot.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/unique_id_generator.h"
@@ -69,8 +68,6 @@ class VideoChannel;
 class VoiceChannel;
 
 class BaseChannel : public ChannelInterface,
-                    // TODO(tommi): Remove has_slots inheritance.
-                    public sigslot::has_slots<>,
                     // TODO(tommi): Consider implementing these interfaces
                     // via composition.
                     public MediaChannelNetworkInterface,

third_party/libwebrtc/pc/dtls_srtp_transport_unittest.cc

@@ -77,17 +77,23 @@ class DtlsSrtpTransportTest : public ::testing::Test,
     dtls_srtp_transport2_ =
         MakeDtlsSrtpTransport(rtp_dtls2, rtcp_dtls2, rtcp_mux_enabled);
 
-    dtls_srtp_transport1_->SignalRtcpPacketReceived.connect(
+    dtls_srtp_transport1_->SubscribeRtcpPacketReceived(
         &transport_observer1_,
-        &webrtc::TransportObserver::OnRtcpPacketReceived);
-    dtls_srtp_transport1_->SignalReadyToSend.connect(
-        &transport_observer1_, &webrtc::TransportObserver::OnReadyToSend);
+        [this](rtc::CopyOnWriteBuffer* buffer, int64_t packet_time_ms) {
+          transport_observer1_.OnRtcpPacketReceived(buffer, packet_time_ms);
+        });
+    dtls_srtp_transport1_->SubscribeReadyToSend(
+        &transport_observer1_,
+        [this](bool ready) { transport_observer1_.OnReadyToSend(ready); });
 
-    dtls_srtp_transport2_->SignalRtcpPacketReceived.connect(
+    dtls_srtp_transport2_->SubscribeRtcpPacketReceived(
         &transport_observer2_,
-        &webrtc::TransportObserver::OnRtcpPacketReceived);
-    dtls_srtp_transport2_->SignalReadyToSend.connect(
-        &transport_observer2_, &webrtc::TransportObserver::OnReadyToSend);
+        [this](rtc::CopyOnWriteBuffer* buffer, int64_t packet_time_ms) {
+          transport_observer2_.OnRtcpPacketReceived(buffer, packet_time_ms);
+        });
+    dtls_srtp_transport2_->SubscribeReadyToSend(
+        &transport_observer2_,
+        [this](bool ready) { transport_observer2_.OnReadyToSend(ready); });
     webrtc::RtpDemuxerCriteria demuxer_criteria;
     // 0x00 is the payload type used in kPcmuFrame.
     demuxer_criteria.payload_types() = {0x00};

third_party/libwebrtc/pc/jsep_transport_controller.cc

@@ -1090,10 +1090,16 @@ RTCError JsepTransportController::MaybeCreateJsepTransport(
             UpdateAggregateStates_n();
           });
 
-  jsep_transport->rtp_transport()->SignalRtcpPacketReceived.connect(
-      this, &JsepTransportController::OnRtcpPacketReceived_n);
-  jsep_transport->rtp_transport()->SignalUnDemuxableRtpPacketReceived.connect(
-      this, &JsepTransportController::OnUnDemuxableRtpPacketReceived_n);
+  jsep_transport->rtp_transport()->SubscribeRtcpPacketReceived(
+      this, [this](rtc::CopyOnWriteBuffer* buffer, int64_t packet_time_ms) {
+        RTC_DCHECK_RUN_ON(network_thread_);
+        OnRtcpPacketReceived_n(buffer, packet_time_ms);
+      });
+  jsep_transport->rtp_transport()->SetUnDemuxableRtpPacketReceivedHandler(
+      [this](webrtc::RtpPacketReceived& packet) {
+        RTC_DCHECK_RUN_ON(network_thread_);
+        OnUnDemuxableRtpPacketReceived_n(packet);
+      });
 
   transports_.RegisterTransport(content_info.name, std::move(jsep_transport));
   UpdateAggregateStates_n();

third_party/libwebrtc/pc/rtp_transport.cc

@@ -59,7 +59,7 @@ void RtpTransport::SetRtpPacketTransport(
     rtp_packet_transport_->SignalWritableState.disconnect(this);
     rtp_packet_transport_->SignalSentPacket.disconnect(this);
     // Reset the network route of the old transport.
-    SignalNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
+    SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
   }
   if (new_packet_transport) {
     new_packet_transport->SignalReadyToSend.connect(
@@ -73,7 +73,7 @@ void RtpTransport::SetRtpPacketTransport(
     new_packet_transport->SignalSentPacket.connect(this,
                                                    &RtpTransport::OnSentPacket);
     // Set the network route for the new transport.
-    SignalNetworkRouteChanged(new_packet_transport->network_route());
+    SendNetworkRouteChanged(new_packet_transport->network_route());
   }
 
   rtp_packet_transport_ = new_packet_transport;
@@ -95,7 +95,7 @@ void RtpTransport::SetRtcpPacketTransport(
     rtcp_packet_transport_->SignalWritableState.disconnect(this);
     rtcp_packet_transport_->SignalSentPacket.disconnect(this);
     // Reset the network route of the old transport.
-    SignalNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
+    SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
   }
   if (new_packet_transport) {
     new_packet_transport->SignalReadyToSend.connect(
@@ -109,7 +109,7 @@ void RtpTransport::SetRtcpPacketTransport(
     new_packet_transport->SignalSentPacket.connect(this,
                                                    &RtpTransport::OnSentPacket);
     // Set the network route for the new transport.
-    SignalNetworkRouteChanged(new_packet_transport->network_route());
+    SendNetworkRouteChanged(new_packet_transport->network_route());
   }
   rtcp_packet_transport_ = new_packet_transport;
 
@@ -195,7 +195,7 @@ void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet,
   if (!rtp_demuxer_.OnRtpPacket(parsed_packet)) {
     RTC_LOG(LS_VERBOSE) << "Failed to demux RTP packet: "
                         << RtpDemuxer::DescribePacket(parsed_packet);
-    SignalUnDemuxableRtpPacketReceived(parsed_packet);
+    NotifyUnDemuxableRtpPacketReceived(parsed_packet);
   }
 }
 
@@ -212,21 +212,21 @@ void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
 
 void RtpTransport::OnNetworkRouteChanged(
     absl::optional<rtc::NetworkRoute> network_route) {
-  SignalNetworkRouteChanged(network_route);
+  SendNetworkRouteChanged(network_route);
 }
 
 void RtpTransport::OnWritableState(
     rtc::PacketTransportInternal* packet_transport) {
   RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
              packet_transport == rtcp_packet_transport_);
-  SignalWritableState(IsTransportWritable());
+  SendWritableState(IsTransportWritable());
 }
 
 void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
                                 const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
              packet_transport == rtcp_packet_transport_);
-  SignalSentPacket(sent_packet);
+  SendSentPacket(sent_packet);
 }
 
 void RtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
@@ -236,7 +236,7 @@ void RtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
 
 void RtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
                                         int64_t packet_time_us) {
-  SignalRtcpPacketReceived(&packet, packet_time_us);
+  SendRtcpPacketReceived(&packet, packet_time_us);
 }
 
 void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
@@ -286,7 +286,7 @@ void RtpTransport::MaybeSignalReadyToSend() {
       rtp_ready_to_send_ && (rtcp_ready_to_send_ || rtcp_mux_enabled_);
   if (ready_to_send != ready_to_send_) {
     ready_to_send_ = ready_to_send;
-    SignalReadyToSend(ready_to_send);
+    SendReadyToSend(ready_to_send);
   }
 }
 

third_party/libwebrtc/pc/rtp_transport_internal.h

@@ -12,13 +12,14 @@
 #define PC_RTP_TRANSPORT_INTERNAL_H_
 
 #include <string>
+#include <utility>
 
 #include "call/rtp_demuxer.h"
 #include "p2p/base/ice_transport_internal.h"
 #include "pc/session_description.h"
+#include "rtc_base/callback_list.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/ssl_stream_adapter.h"
-#include "rtc_base/third_party/sigslot/sigslot.h"
 
 namespace rtc {
 class CopyOnWriteBuffer;
@@ -50,27 +51,59 @@ class RtpTransportInternal : public sigslot::has_slots<> {
   // Called whenever a transport's ready-to-send state changes. The argument
   // is true if all used transports are ready to send. This is more specific
   // than just "writable"; it means the last send didn't return ENOTCONN.
-  sigslot::signal1<bool> SignalReadyToSend;
+  void SubscribeReadyToSend(const void* tag,
+                            absl::AnyInvocable<void(bool)> callback) {
+    callback_list_ready_to_send_.AddReceiver(tag, std::move(callback));
+  }
+  void UnsubscribeReadyToSend(const void* tag) {
+    callback_list_ready_to_send_.RemoveReceivers(tag);
+  }
 
   // Called whenever an RTCP packet is received. There is no equivalent signal
   // for demuxable RTP packets because they would be forwarded to the
   // BaseChannel through the RtpDemuxer callback.
-  sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived;
+  void SubscribeRtcpPacketReceived(
+      const void* tag,
+      absl::AnyInvocable<void(rtc::CopyOnWriteBuffer*, int64_t)> callback) {
+    callback_list_rtcp_packet_received_.AddReceiver(tag, std::move(callback));
+  }
+  // There doesn't seem to be a need to unsubscribe from this signal.
 
   // Called whenever a RTP packet that can not be demuxed by the transport is
   // received.
-  sigslot::signal<const webrtc::RtpPacketReceived&>
-      SignalUnDemuxableRtpPacketReceived;
+  void SetUnDemuxableRtpPacketReceivedHandler(
+      absl::AnyInvocable<void(webrtc::RtpPacketReceived&)> callback) {
+    callback_undemuxable_rtp_packet_received_ = std::move(callback);
+  }
 
   // Called whenever the network route of the P2P layer transport changes.
   // The argument is an optional network route.
-  sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
+  void SubscribeNetworkRouteChanged(
+      const void* tag,
+      absl::AnyInvocable<void(absl::optional<rtc::NetworkRoute>)> callback) {
+    callback_list_network_route_changed_.AddReceiver(tag, std::move(callback));
+  }
+  void UnsubscribeNetworkRouteChanged(const void* tag) {
+    callback_list_network_route_changed_.RemoveReceivers(tag);
+  }
 
   // Called whenever a transport's writable state might change. The argument is
   // true if the transport is writable, otherwise it is false.
-  sigslot::signal1<bool> SignalWritableState;
-
-  sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
+  void SubscribeWritableState(const void* tag,
+                              absl::AnyInvocable<void(bool)> callback) {
+    callback_list_writable_state_.AddReceiver(tag, std::move(callback));
+  }
+  void UnsubscribeWritableState(const void* tag) {
+    callback_list_writable_state_.RemoveReceivers(tag);
+  }
+  void SubscribeSentPacket(
+      const void* tag,
+      absl::AnyInvocable<void(const rtc::SentPacket&)> callback) {
+    callback_list_sent_packet_.AddReceiver(tag, std::move(callback));
+  }
+  void UnsubscribeSentPacket(const void* tag) {
+    callback_list_sent_packet_.RemoveReceivers(tag);
+  }
 
   virtual bool IsWritable(bool rtcp) const = 0;
 
@@ -103,6 +136,37 @@ class RtpTransportInternal : public sigslot::has_slots<> {
                                       RtpPacketSinkInterface* sink) = 0;
 
   virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
+
+ protected:
+  void SendReadyToSend(bool arg) { callback_list_ready_to_send_.Send(arg); }
+  void SendRtcpPacketReceived(rtc::CopyOnWriteBuffer* buffer,
+                              int64_t packet_time_us) {
+    callback_list_rtcp_packet_received_.Send(buffer, packet_time_us);
+  }
+  void NotifyUnDemuxableRtpPacketReceived(RtpPacketReceived& packet) {
+    callback_undemuxable_rtp_packet_received_(packet);
+  }
+  void SendNetworkRouteChanged(absl::optional<rtc::NetworkRoute> route) {
+    callback_list_network_route_changed_.Send(route);
+  }
+  void SendWritableState(bool state) {
+    callback_list_writable_state_.Send(state);
+  }
+  void SendSentPacket(const rtc::SentPacket& packet) {
+    callback_list_sent_packet_.Send(packet);
+  }
+
+ private:
+  CallbackList<bool> callback_list_ready_to_send_;
+  CallbackList<rtc::CopyOnWriteBuffer*, int64_t>
+      callback_list_rtcp_packet_received_;
+  absl::AnyInvocable<void(webrtc::RtpPacketReceived&)>
+      callback_undemuxable_rtp_packet_received_ =
+          [](RtpPacketReceived& packet) {};
+  CallbackList<absl::optional<rtc::NetworkRoute>>
+      callback_list_network_route_changed_;
+  CallbackList<bool> callback_list_writable_state_;
+  CallbackList<const rtc::SentPacket&> callback_list_sent_packet_;
 };
 
 }  // namespace webrtc

third_party/libwebrtc/pc/rtp_transport_unittest.cc

@@ -30,9 +30,12 @@ class SignalObserver : public sigslot::has_slots<> {
  public:
   explicit SignalObserver(RtpTransport* transport) {
     transport_ = transport;
-    transport->SignalReadyToSend.connect(this, &SignalObserver::OnReadyToSend);
-    transport->SignalNetworkRouteChanged.connect(
-        this, &SignalObserver::OnNetworkRouteChanged);
+    transport->SubscribeReadyToSend(
+        this, [this](bool ready) { OnReadyToSend(ready); });
+    transport->SubscribeNetworkRouteChanged(
+        this, [this](absl::optional<rtc::NetworkRoute> route) {
+          OnNetworkRouteChanged(route);
+        });
     if (transport->rtp_packet_transport()) {
       transport->rtp_packet_transport()->SignalSentPacket.connect(
           this, &SignalObserver::OnSentPacket);

third_party/libwebrtc/pc/srtp_transport.cc

@@ -244,7 +244,7 @@ void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
     return;
   }
   packet.SetSize(len);
-  SignalRtcpPacketReceived(&packet, packet_time_us);
+  SendRtcpPacketReceived(&packet, packet_time_us);
 }
 
 void SrtpTransport::OnNetworkRouteChanged(
@@ -257,12 +257,12 @@ void SrtpTransport::OnNetworkRouteChanged(
     }
     network_route->packet_overhead += srtp_overhead;
   }
-  SignalNetworkRouteChanged(network_route);
+  SendNetworkRouteChanged(network_route);
 }
 
 void SrtpTransport::OnWritableState(
     rtc::PacketTransportInternal* packet_transport) {
-  SignalWritableState(IsWritable(/*rtcp=*/false) && IsWritable(/*rtcp=*/true));
+  SendWritableState(IsWritable(/*rtcp=*/false) && IsWritable(/*rtcp=*/true));
 }
 
 bool SrtpTransport::SetRtpParams(int send_crypto_suite,
@@ -515,7 +515,7 @@ void SrtpTransport::MaybeUpdateWritableState() {
   // Only fire the signal if the writable state changes.
   if (writable_ != writable) {
     writable_ = writable;
-    SignalWritableState(writable_);
+    SendWritableState(writable_);
   }
 }
 

third_party/libwebrtc/pc/srtp_transport_unittest.cc

@@ -66,10 +66,16 @@ class SrtpTransportTest : public ::testing::Test, public sigslot::has_slots<> {
     srtp_transport1_->SetRtpPacketTransport(rtp_packet_transport1_.get());
     srtp_transport2_->SetRtpPacketTransport(rtp_packet_transport2_.get());
 
-    srtp_transport1_->SignalRtcpPacketReceived.connect(
-        &rtp_sink1_, &TransportObserver::OnRtcpPacketReceived);
-    srtp_transport2_->SignalRtcpPacketReceived.connect(
-        &rtp_sink2_, &TransportObserver::OnRtcpPacketReceived);
+    srtp_transport1_->SubscribeRtcpPacketReceived(
+        &rtp_sink1_,
+        [this](rtc::CopyOnWriteBuffer* buffer, int64_t packet_time_ms) {
+          rtp_sink1_.OnRtcpPacketReceived(buffer, packet_time_ms);
+        });
+    srtp_transport2_->SubscribeRtcpPacketReceived(
+        &rtp_sink2_,
+        [this](rtc::CopyOnWriteBuffer* buffer, int64_t packet_time_ms) {
+          rtp_sink2_.OnRtcpPacketReceived(buffer, packet_time_ms);
+        });
 
     RtpDemuxerCriteria demuxer_criteria;
     // 0x00 is the payload type used in kPcmuFrame.

third_party/libwebrtc/pc/test/rtp_transport_test_util.h

@@ -14,24 +14,26 @@
 #include "call/rtp_packet_sink_interface.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "pc/rtp_transport_internal.h"
-#include "rtc_base/third_party/sigslot/sigslot.h"
 
 namespace webrtc {
 
 // Used to handle the signals when the RtpTransport receives an RTP/RTCP packet.
 // Used in Rtp/Srtp/DtlsTransport unit tests.
-class TransportObserver : public RtpPacketSinkInterface,
-                          public sigslot::has_slots<> {
+class TransportObserver : public RtpPacketSinkInterface {
  public:
   TransportObserver() {}
 
   explicit TransportObserver(RtpTransportInternal* rtp_transport) {
-    rtp_transport->SignalRtcpPacketReceived.connect(
-        this, &TransportObserver::OnRtcpPacketReceived);
-    rtp_transport->SignalReadyToSend.connect(this,
-                                             &TransportObserver::OnReadyToSend);
-    rtp_transport->SignalUnDemuxableRtpPacketReceived.connect(
-        this, &TransportObserver::OnUndemuxableRtpPacket);
+    rtp_transport->SubscribeRtcpPacketReceived(
+        this, [this](rtc::CopyOnWriteBuffer* buffer, int64_t packet_time_ms) {
+          OnRtcpPacketReceived(buffer, packet_time_ms);
+        });
+    rtp_transport->SubscribeReadyToSend(
+        this, [this](bool arg) { OnReadyToSend(arg); });
+    rtp_transport->SetUnDemuxableRtpPacketReceivedHandler(
+        [this](webrtc::RtpPacketReceived& packet) {
+          OnUndemuxableRtpPacket(packet);
+        });
   }
 
   // RtpPacketInterface override.