Merge branch 'fix/ros2_jazzy' into develop-piezo

https://github.com/EricVoll/webrtc_ros/tree/fix/ros2_jazzy で公開されているJazzy対応修正の取り込み

Author: HABU Miki

webrtc/build/get_gn

@@ -25,7 +25,7 @@ then
 	git clone https://gn.googlesource.com/gn third_party/gn
 	cd third_party/gn
 	git checkout 501b49a3ab4f0d099457b6e5b62c709a1d2311be
-	CC=gcc CXX=g++ LDFLAGS=-fuse-ld=gold python build/gen.py
+	CC=gcc CXX=g++ LDFLAGS=-fuse-ld=gold python3 build/gen.py
 	"$rootdir/depot_tools/ninja" -C out gn
 	cp out/gn .
 fi

webrtc_ros/CMakeLists.txt

@@ -11,6 +11,7 @@ find_package(webrtc_ros_msgs REQUIRED)
 find_package(webrtc REQUIRED)
 find_package(X11 REQUIRED)
 find_package(jsoncpp REQUIRED)
+find_package(OpenCV REQUIRED)
 
 
 ###########
@@ -23,7 +24,7 @@ include_directories(
   ${OpenCV_INCLUDE_DIRS}
   ${webrtc_ros_msgs_INCLUDE_DIRS}
   ${webrtc_INCLUDE_DIRS}
-  
+
 )
 add_definitions(${webrtc_DEFINITIONS})
 
@@ -56,15 +57,17 @@ ament_target_dependencies(
     image_transport
     rclcpp
     std_msgs
-    webrtc_ros_msgs
-    
+    webrtc_ros_msgs    
+
 )
 
 target_link_libraries(${PROJECT_NAME}_server_node 
   ${webrtc_LIBRARIES}
   webrtc
   jsoncpp_lib
   ${X11_LIBRARIES}
+  ${OpenCV_LIBS}
+  ament_index_cpp::ament_index_cpp
 )
 
 set_target_properties(${PROJECT_NAME}_server_node PROPERTIES COMPILE_OPTIONS "-std=c++17")

webrtc_ros/include/webrtc_ros/ros_video_capturer.h

@@ -37,7 +37,9 @@ class RosVideoCapturer :
 	bool remote() const override;
 
 private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(RosVideoCapturer);
+  RosVideoCapturer(const RosVideoCapturer&) = delete;
+  RosVideoCapturer& operator=(const RosVideoCapturer&) = delete;
+
   boost::shared_ptr<RosVideoCapturerImpl> impl_;
 };
 
@@ -56,7 +58,8 @@ class RosVideoCapturerImpl : public boost::enable_shared_from_this<RosVideoCaptu
   void Stop();
 
 private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(RosVideoCapturerImpl);
+  RosVideoCapturerImpl(const RosVideoCapturerImpl&) = delete;
+  RosVideoCapturerImpl& operator=(const RosVideoCapturerImpl&) = delete;
 
   ImageTransportFactory it_;
   const std::string topic_, transport_;

webrtc_ros/include/webrtc_ros/ros_video_renderer.h

@@ -16,7 +16,8 @@ class RosVideoRenderer :
   virtual void OnFrame(const webrtc::VideoFrame& frame) override;
 
 private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(RosVideoRenderer);
+  RosVideoRenderer(const RosVideoRenderer&) = delete;
+  RosVideoRenderer& operator=(const RosVideoRenderer&) = delete;
   std::shared_ptr<image_transport::ImageTransport> it_;
   const std::string topic_;
   image_transport::Publisher pub_;

webrtc_ros/src/ros_video_capturer.cpp

@@ -1,8 +1,7 @@
 #include "webrtc_ros/ros_video_capturer.h"
-#include "webrtc/rtc_base/bind.h"
 
 #include <rclcpp/rclcpp.hpp>
-#include <cv_bridge/cv_bridge.h>
+#include <cv_bridge/cv_bridge.hpp>
 #include <boost/enable_shared_from_this.hpp>
 
 namespace webrtc_ros

webrtc_ros/src/ros_video_renderer.cpp

@@ -1,6 +1,6 @@
 #include <webrtc_ros/ros_video_renderer.h>
 #include <rclcpp/rclcpp.hpp>
-#include <cv_bridge/cv_bridge.h>
+#include <cv_bridge/cv_bridge.hpp>
 #include <webrtc/3rdparty/libyuv/convert_from.h>
 
 namespace webrtc_ros

webrtc_ros/src/webrtc_client.cpp

@@ -5,7 +5,6 @@
 #include <webrtc_ros/ice_candidate_message.h>
 //#include "talk/media/devices/devicemanager.h"
 #include <webrtc/api/video/video_source_interface.h>
-#include <webrtc/rtc_base/bind.h>
 #include <webrtc_ros/ros_video_capturer.h>
 #include <webrtc_ros_msgs/srv/get_ice_servers.hpp>
 
@@ -177,8 +176,7 @@ class MessageHandlerImpl : public MessageHandler {
   {
     WebrtcClientPtr _this = weak_this_.lock();
     if (_this)
-      _this->signaling_thread_->Invoke<void>(RTC_FROM_HERE, rtc::Bind(&WebrtcClient::handle_message,
-						       _this.get(), type, raw));
+      _this->signaling_thread_->BlockingCall([t = _this.get(), type, raw] { t->handle_message(type, raw); });
   }
 private:
   WebrtcClientWeakPtr weak_this_;
@@ -285,7 +283,7 @@ void WebrtcClient::handle_message(MessageHandler::Type type, const std::string&
 
                     rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = peer_connection_factory_->CreateLocalMediaStream(stream_id);
 
-                    if (!peer_connection_->AddStream(stream))
+                    if (!peer_connection_->AddStream(stream.get()))
                     {
                       RCLCPP_WARN(nh_->get_logger(), "Adding stream to PeerConnection failed");
                 continue;
@@ -300,7 +298,7 @@ void WebrtcClient::handle_message(MessageHandler::Type type, const std::string&
                 RCLCPP_WARN_STREAM(nh_->get_logger(), "Stream not found with id: " << stream_id);
                 continue;
               }
-                    peer_connection_->RemoveStream(stream);
+                    peer_connection_->RemoveStream(stream.get());
             }
             else if(action.type == ConfigureAction::kAddVideoTrackActionName) {
               FIND_PROPERTY_OR_CONTINUE("stream_id", stream_id);
@@ -326,7 +324,7 @@ void WebrtcClient::handle_message(MessageHandler::Type type, const std::string&
                       rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
                         peer_connection_factory_->CreateVideoTrack(
                           track_id,
-                          capturer));
+                          capturer.get()));
                       stream->AddTrack(video_track);
                       capturer->Start();
               }
@@ -356,10 +354,9 @@ void WebrtcClient::handle_message(MessageHandler::Type type, const std::string&
               if(audio_type == "local") {
                 cricket::AudioOptions options;
                 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-                  peer_connection_factory_->CreateAudioTrack(
-                    track_id,
-              peer_connection_factory_->CreateAudioSource(options)));
-                      stream->AddTrack(audio_track);
+                    peer_connection_factory_->CreateAudioTrack(track_id, peer_connection_factory_->CreateAudioSource(options).get())
+                );
+                stream->AddTrack(audio_track);
               }
               else {
                 RCLCPP_WARN_STREAM(nh_->get_logger(), "Unknown video source type: " << audio_type);
@@ -416,7 +413,7 @@ void WebrtcClient::handle_message(MessageHandler::Type type, const std::string&
 
       RCLCPP_DEBUG_STREAM(nh_->get_logger(), "Received remote description: " << message.sdp);
       rtc::scoped_refptr<DummySetSessionDescriptionObserver> dummy_set_description_observer(new rtc::RefCountedObject<DummySetSessionDescriptionObserver>());
-      peer_connection_->SetRemoteDescription(dummy_set_description_observer, session_description);
+      peer_connection_->SetRemoteDescription(dummy_set_description_observer.get(), session_description);
     }
     else if (IceCandidateMessage::isIceCandidate(message_json))
     {
@@ -465,7 +462,7 @@ void WebrtcClient::handle_message(MessageHandler::Type type, const std::string&
 void WebrtcClient::OnSessionDescriptionSuccess(webrtc::SessionDescriptionInterface* description)
 {
   rtc::scoped_refptr<DummySetSessionDescriptionObserver> dummy_set_description_observer(new rtc::RefCountedObject<DummySetSessionDescriptionObserver>());
-  peer_connection_->SetLocalDescription(dummy_set_description_observer, description);
+  peer_connection_->SetLocalDescription(dummy_set_description_observer.get(), description);
 
   SdpMessage message;
   if (message.fromSessionDescription(*description))

webrtc_ros/src/webrtc_ros_server.cpp

@@ -2,15 +2,16 @@
 #include <webrtc_ros/webrtc_ros_server.h>
 #include "webrtc/rtc_base/ssl_adapter.h"
 
-#include "webrtc/rtc_base/bind.h"
-
 namespace webrtc_ros
 {
 
 MessageHandler* WebrtcRosServer_handle_new_signaling_channel(void* _this, SignalingChannel *channel)
 {
-    return ((WebrtcRosServer*) _this)->signaling_thread_->Invoke<MessageHandler*>(RTC_FROM_HERE, rtc::Bind(&WebrtcRosServer::handle_new_signaling_channel,
-            (WebrtcRosServer*)_this, channel));
+    MessageHandler* result;
+    ((WebrtcRosServer*) _this)->signaling_thread_->BlockingCall( 
+        [&]{ result = ((WebrtcRosServer*) _this)->handle_new_signaling_channel(channel);}
+    );
+    return result;
 }
 
 WebrtcRosServer::WebrtcRosServer(rclcpp::Node::SharedPtr nh)