Merging upstream version 126.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

5 files changed, 25 insertions, 33 deletions

diff --git a/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc
index 40af78cdac..0e895c520b 100644
--- a/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc
+++ b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc
@@ -154,8 +154,7 @@ void AndroidCallClient::CreatePeerConnectionFactory() {
   pcf_deps.worker_thread = worker_thread_.get();
   pcf_deps.signaling_thread = signaling_thread_.get();
   pcf_deps.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
-  pcf_deps.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>(
-      pcf_deps.task_queue_factory.get());
+  pcf_deps.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>();
 
   pcf_deps.video_encoder_factory =
       std::make_unique<webrtc::InternalEncoderFactory>();
diff --git a/third_party/libwebrtc/examples/androidvoip/BUILD.gn b/third_party/libwebrtc/examples/androidvoip/BUILD.gn
index cea05ea128..d390815406 100644
--- a/third_party/libwebrtc/examples/androidvoip/BUILD.gn
+++ b/third_party/libwebrtc/examples/androidvoip/BUILD.gn
@@ -71,7 +71,7 @@ if (is_android) {
       "//api/task_queue:default_task_queue_factory",
       "//api/voip:voip_api",
       "//api/voip:voip_engine_factory",
-      "//rtc_base/third_party/sigslot:sigslot",
+      "//rtc_base/network:received_packet",
       "//sdk/android:native_api_audio_device_module",
       "//sdk/android:native_api_base",
       "//sdk/android:native_api_jni",
diff --git a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc
index 8a0a3badb9..69327990e0 100644
--- a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc
+++ b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc
@@ -313,8 +313,10 @@ void AndroidVoipClient::StartSession(JNIEnv* env) {
                                             /*isSuccessful=*/false);
     return;
   }
-  rtp_socket_->SignalReadPacket.connect(
-      this, &AndroidVoipClient::OnSignalReadRTPPacket);
+  rtp_socket_->RegisterReceivedPacketCallback(
+      [&](rtc::AsyncPacketSocket* socket, const rtc::ReceivedPacket& packet) {
+        OnSignalReadRTPPacket(socket, packet);
+      });
 
   rtcp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(),
                                                  rtcp_local_address_));
@@ -324,8 +326,10 @@ void AndroidVoipClient::StartSession(JNIEnv* env) {
                                             /*isSuccessful=*/false);
     return;
   }
-  rtcp_socket_->SignalReadPacket.connect(
-      this, &AndroidVoipClient::OnSignalReadRTCPPacket);
+  rtcp_socket_->RegisterReceivedPacketCallback(
+      [&](rtc::AsyncPacketSocket* socket, const rtc::ReceivedPacket& packet) {
+        OnSignalReadRTCPPacket(socket, packet);
+      });
   Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_,
                                           /*isSuccessful=*/true);
 }
@@ -467,12 +471,11 @@ void AndroidVoipClient::ReadRTPPacket(const std::vector<uint8_t>& packet_copy) {
   RTC_CHECK(result == webrtc::VoipResult::kOk);
 }
 
-void AndroidVoipClient::OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket,
-                                              const char* rtp_packet,
-                                              size_t size,
-                                              const rtc::SocketAddress& addr,
-                                              const int64_t& timestamp) {
-  std::vector<uint8_t> packet_copy(rtp_packet, rtp_packet + size);
+void AndroidVoipClient::OnSignalReadRTPPacket(
+    rtc::AsyncPacketSocket* socket,
+    const rtc::ReceivedPacket& packet) {
+  std::vector<uint8_t> packet_copy(packet.payload().begin(),
+                                   packet.payload().end());
   voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] {
     ReadRTPPacket(packet_copy);
   });
@@ -492,12 +495,11 @@ void AndroidVoipClient::ReadRTCPPacket(
   RTC_CHECK(result == webrtc::VoipResult::kOk);
 }
 
-void AndroidVoipClient::OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket,
-                                               const char* rtcp_packet,
-                                               size_t size,
-                                               const rtc::SocketAddress& addr,
-                                               const int64_t& timestamp) {
-  std::vector<uint8_t> packet_copy(rtcp_packet, rtcp_packet + size);
+void AndroidVoipClient::OnSignalReadRTCPPacket(
+    rtc::AsyncPacketSocket* socket,
+    const rtc::ReceivedPacket& packet) {
+  std::vector<uint8_t> packet_copy(packet.payload().begin(),
+                                   packet.payload().end());
   voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] {
     ReadRTCPPacket(packet_copy);
   });
diff --git a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h
index e2f1c64590..1d9a13b29d 100644
--- a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h
+++ b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h
@@ -23,8 +23,8 @@
 #include "api/voip/voip_engine.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/async_udp_socket.h"
+#include "rtc_base/network/received_packet.h"
 #include "rtc_base/socket_address.h"
-#include "rtc_base/third_party/sigslot/sigslot.h"
 #include "rtc_base/thread.h"
 #include "sdk/android/native_api/jni/scoped_java_ref.h"
 
@@ -40,8 +40,7 @@ namespace webrtc_examples {
 // with consistent thread usage requirement with ProcessThread used
 // within VoipEngine, as well as providing asynchronicity to the
 // caller. AndroidVoipClient is meant to be used by Java through JNI.
-class AndroidVoipClient : public webrtc::Transport,
-                          public sigslot::has_slots<> {
+class AndroidVoipClient : public webrtc::Transport {
  public:
   // Returns a pointer to an AndroidVoipClient object. Clients should
   // use this factory method to create AndroidVoipClient objects. The
@@ -122,17 +121,10 @@ class AndroidVoipClient : public webrtc::Transport,
                const webrtc::PacketOptions& options) override;
   bool SendRtcp(rtc::ArrayView<const uint8_t> packet) override;
 
-  // Slots for sockets to connect to.
   void OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket,
-                             const char* rtp_packet,
-                             size_t size,
-                             const rtc::SocketAddress& addr,
-                             const int64_t& timestamp);
+                             const rtc::ReceivedPacket& packet);
   void OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket,
-                              const char* rtcp_packet,
-                              size_t size,
-                              const rtc::SocketAddress& addr,
-                              const int64_t& timestamp);
+                              const rtc::ReceivedPacket& packet);
 
  private:
   AndroidVoipClient(JNIEnv* env,
diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm
index 996c6a9c7f..2601beed71 100644
--- a/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm
@@ -126,8 +126,7 @@ void ObjCCallClient::CreatePeerConnectionFactory() {
       [[RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) alloc] init]);
   dependencies.audio_processing = webrtc::AudioProcessingBuilder().Create();
   webrtc::EnableMedia(dependencies);
-  dependencies.event_log_factory =
-      std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
+  dependencies.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>();
   pcf_ = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
   RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_.get();
 }