fix: datachannel deadlocks & dispose crashes (#87)

- Fix datachannel deadlock on dispose
- Fix audio device being disposed on the wrong thread
- Allow multiple RtcRuntime to be created (Useful for unit tests where we use multiple PeerConnectionFactory)

webrtc-sys/include/livekit/data_channel.h

@@ -40,6 +40,7 @@ class DataChannel {
   explicit DataChannel(
       std::shared_ptr<RtcRuntime> rtc_runtime,
       rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel);
+  ~DataChannel();
 
   void register_observer(rust::Box<DataChannelObserverWrapper> observer) const;
   void unregister_observer() const;
@@ -64,8 +65,6 @@ class NativeDataChannelObserver : public webrtc::DataChannelObserver {
   NativeDataChannelObserver(rust::Box<DataChannelObserverWrapper> observer,
                             const DataChannel* dc);
 
-  ~NativeDataChannelObserver();
-
   void OnStateChange() override;
   void OnMessage(const webrtc::DataBuffer& buffer) override;
   void OnBufferedAmountChange(uint64_t sent_data_size) override;

webrtc-sys/include/livekit/peer_connection.h

@@ -48,6 +48,8 @@ class PeerConnection {
       std::unique_ptr<NativePeerConnectionObserver> observer,
       rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection);
 
+  ~PeerConnection();
+
   void create_offer(
       RtcOfferAnswerOptions options,
       rust::Box<AsyncContext> ctx,

webrtc-sys/include/livekit/peer_connection_factory.h

@@ -17,6 +17,8 @@
 #pragma once
 
 #include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "livekit/audio_device.h"
 #include "media_stream.h"
 #include "peer_connection.h"
 #include "rtp_parameters.h"
@@ -57,6 +59,7 @@ class PeerConnectionFactory {
 
  private:
   std::shared_ptr<RtcRuntime> rtc_runtime_;
+  rtc::scoped_refptr<AudioDevice> audio_device_;
   rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> peer_factory_;
 };
 

webrtc-sys/include/livekit/webrtc.h

@@ -80,6 +80,8 @@ class RtcRuntime : public std::enable_shared_from_this<RtcRuntime> {
   // have one livekit::VideoTrack associated with it).
   // The only reason we to do that is to allow to add states inside our
   // wrappers (e.g: the sinks_ member inside AudioTrack)
+  // DataChannel and the PeerConnectionFactory don't need to do this (There's no
+  // way to retrieve them after creation)
   webrtc::Mutex mutex_;
   std::vector<std::weak_ptr<MediaStreamTrack>> media_stream_tracks_;
   // We don't have additonal state in RtpReceiver and RtpSender atm..
@@ -87,9 +89,9 @@ class RtcRuntime : public std::enable_shared_from_this<RtcRuntime> {
   // std::vector<std::weak_ptr<RtpSender>> rtp_senders_;
 
 #ifdef WEBRTC_WIN
-  rtc::WinsockInitializer winsock_;
-  rtc::PhysicalSocketServer ss_;
-  rtc::AutoSocketServerThread main_thread_{&ss_};
+  // rtc::WinsockInitializer winsock_;
+  // rtc::PhysicalSocketServer ss_;
+  // rtc::AutoSocketServerThread main_thread_{&ss_};
 #endif
 };