GstRemoteParticipant.hxx

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#if !defined(GstRemoteParticipant_hxx)
#define GstRemoteParticipant_hxx
 
#include <functional>
#include <map>
 
#include <media/gstreamer/GstRtpManager.hxx>
#include <media/gstreamer/GstRtpSession.hxx>
 
#include "ConversationManager.hxx"
#include "Participant.hxx"
#include "RemoteParticipant.hxx"
#include "GstRemoteParticipantDialogSet.hxx"
#include "GstParticipant.hxx"
 
#include <resip/dum/AppDialogSet.hxx>
#include <resip/dum/AppDialog.hxx>
#include <resip/dum/InviteSessionHandler.hxx>
#include <resip/dum/DialogSetHandler.hxx>
#include <resip/dum/SubscriptionHandler.hxx>
 
#include <memory>
 
#include <gst/sdp/gstsdpmessage.h>
#include <glibmm/object.h>
 
#include <gst/webrtc/webrtc.h>
 
namespace resip
{
class DialogUsageManager;
class SipMessage;
}
 
// Disable warning 4250
// VS2019 give a 4250 warning:  
// GstRemoteParticipant.hxx(80,1): warning C4250: 'recon::GstRemoteParticipant': inherits 'recon::RemoteParticipant::recon::RemoteParticipant::addToConversation' via dominance
#if defined(WIN32) && !defined(__GNUC__)
#pragma warning( disable : 4250 )
#endif
 
namespace recon
{
class GstMediaStackAdapter;
 
class GstRemoteParticipant : public virtual RemoteParticipant, public virtual GstParticipant
{
public:
   GstRemoteParticipant(ParticipantHandle partHandle,   // UAC
                     ConversationManager& conversationManager,
                     GstMediaStackAdapter& gstreamerMediaStackAdapter,
                     resip::DialogUsageManager& dum,
                     RemoteParticipantDialogSet& remoteParticipantDialogSet);
 
   GstRemoteParticipant(ConversationManager& conversationManager,            // UAS or forked leg
                     GstMediaStackAdapter& gstreamerMediaStackAdapter,
                     resip::DialogUsageManager& dum,
                     RemoteParticipantDialogSet& remoteParticipantDialogSet);
 
   virtual ~GstRemoteParticipant();
 
   virtual void buildSdpOffer(bool holdSdp, CallbackSdpReady sdpReady, bool preferExistingSdp = false);
 
   virtual int getConnectionPortOnBridge();
   virtual bool hasInput() { return true; }
   virtual bool hasOutput() { return true; }
   virtual int getMediaConnectionId();
 
   virtual void applyBridgeMixWeights() override;
   virtual void applyBridgeMixWeights(Conversation* removedConversation) override;
 
   virtual void adjustRTPStreams(bool sendingOffer=false);
 
   void onIceGatheringDone() { mIceGatheringDone = true; };
 
   virtual void addToConversation(Conversation *conversation, unsigned int inputGain = 100, unsigned int outputGain = 100) override;
   virtual void removeFromConversation(Conversation *conversation) override;
 
   //virtual std::shared_ptr<gstreamer::BaseRtpEndpoint> getEndpoint() { return mEndpoint; }; // FIXME Gst
   //std::shared_ptr<gstreamer::MediaElement> mMultiqueue; // FIXME Gst
   bool mWaitingModeVideo = false;
   //std::shared_ptr<gstreamer::PlayerEndpoint> mPlayer; // FIXME Gst
   //std::shared_ptr<gstreamer::PassThroughElement> mPassThrough; // FIXME Gst
 
   virtual void waitingMode();
   //virtual std::shared_ptr<gstreamer::MediaElement> getWaitingModeElement();
 
   virtual bool onMediaControlEvent(resip::MediaControlContents::MediaControl& mediaControl);
   virtual bool onTrickleIce(resip::TrickleIceContents& trickleIce) override;
 
   virtual void requestKeyframe() override;
 
protected:
   virtual bool mediaStackPortAvailable();
 
   virtual GstRemoteParticipantDialogSet& getGstDialogSet() { return dynamic_cast<GstRemoteParticipantDialogSet&>(getDialogSet()); };
 
   virtual bool holdPreferExistingSdp() override { return true; };
 
private:
   void initRtpManager();
   void initRtpSession(bool isWebRTC);
   //gstreamer::BaseRtpEndpoint* newEndpoint();
   virtual void linkOutgoingPipeline(unsigned int streamId, Glib::RefPtr<Gst::Bin> bin, const Glib::RefPtr<Gst::Caps> caps);
   virtual void prepareStream(unsigned int streamId, bool loopbackMode);
   virtual bool initEndpointIfRequired(bool isWebRTC);
   //virtual void doIceGathering(gstreamer::ContinuationString sdpReady);
   //typedef Glib::ustring StreamKey;
   virtual bool isWebRTCSession() const;
   virtual void onMediaSourceAdded(const Glib::RefPtr<Gst::Pad>& pad);
   virtual Glib::RefPtr<Gst::Bin> createIncomingPipeline(Glib::RefPtr<Gst::Pad> pad);
   virtual void debugGraph();
   virtual void createAndConnectElements(bool isWebRTC, std::function<void()> cConnected, CallbackSdpReady sdpReady);
   virtual void createAndConnectElementsWebRTC(std::function<void()> cConnected, CallbackSdpReady sdpReady);
   virtual void createAndConnectElementsStandard(std::function<void()> cConnected, CallbackSdpReady sdpReady);
   virtual void setLocalDescription(GstWebRTCSessionDescription* gstwebrtcdesc);
   virtual void setRemoteDescription(GstWebRTCSessionDescription* gstwebrtcdesc);
   virtual void buildSdpAnswer(const resip::SdpContents& offer, CallbackSdpReady sdpReady) override;
 
   bool onGstBusMessage(const Glib::RefPtr<Gst::Bus>&, const Glib::RefPtr<Gst::Message>& message);
   virtual void onRemoteIceCandidate(const resip::Data& candidate, unsigned int lineIndex, const resip::Data& mid);
 
   //std::shared_ptr<gstreamer::BaseRtpEndpoint> mEndpoint;
   volatile bool mIceGatheringDone;  // FIXME Gst use a concurrency primitive, e.g. condition_variable
 
   std::chrono::time_point<std::chrono::steady_clock> mLastLocalKeyframeRequest = std::chrono::steady_clock::now();
 
   Glib::RefPtr<Gst::Caps> mCapsVideo;
   Glib::RefPtr<Gst::Caps> mCapsAudio;
 
public: // FIXME
   bool mRemoveExtraMediaDescriptors;
   bool mSipRtpEndpoint;
   bool mReuseSdpAnswer;
   bool mWSAcceptsKeyframeRequests;
   resip::SdpContents* mLastRemoteSdp;
   bool mWaitingAnswer;  // have sent an offer, waiting for peer to send answer
   bool mTrickleIcePermitted = false; // FIXME - complete
   bool mWebRTCOutgoing = false; // FIXME - use WebRTC for outgoing call
 
   //GstElement* mPipeline;
   //GstElement* mRtpTransportElement;
   Glib::RefPtr<Gst::Pipeline> mPipeline;
   Glib::RefPtr<Gst::Element> mMediaBin;
   Glib::RefPtr<Gst::Element> mRtpTransportElement;
 
   //std::map<MediaTypeName, Glib::RefPtr<Gst::Bin>> mDecodes;
   //std::map<StreamKey, Glib::RefPtr<Gst::Bin>> mEncodes;
   //std::map<MediaTypeName, Glib::RefPtr<Gst::Pad>> mEncodes;
 
   std::function<void(const Glib::RefPtr<Gst::Pad>& pad)> mPadAdded;
 
   /*class MyWebRTCElement : public Gst::Element
         {
            public:
               MyWebRTCElement() : Gst::Element(Glib::ConstructParams()) {};
         };*/
 
   std::shared_ptr<resipgst::GstRtpManager> mRtpManager = 0;  // FIXME - move to MediaStackAdapter
   std::shared_ptr<resipgst::GstRtpSession> mRtpSession = 0;
};
 
}
 
#endif
 
 
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 Copyright (c) 2022, Software Freedom Institute https://softwarefreedom.institute
 Copyright (c) 2022, Daniel Pocock https://danielpocock.com
 Copyright (c) 2021, SIP Spectrum, Inc. www.sipspectrum.com
 Copyright (c) 2007-2008, Plantronics, Inc.
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