mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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607ef6db60
GstWebRTCSCTPTransport is now made into into an abstract base class that only contains property specifications matching the RTCSctpTransport interface of the W3C WebRTC specification, see https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This class is put into the WebRTC library to expose it for applications and to allow for generation of bindings for non-dynamic languages using GObject introspection. The actual implementation is moved to the subclass WebRTCSCTPTransport located in the WebRTC plugin. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
426 lines
13 KiB
C
426 lines
13 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_WEBRTC_FWD_H__
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#define __GST_WEBRTC_FWD_H__
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#ifndef GST_USE_UNSTABLE_API
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#warning "The WebRTC library from gst-plugins-bad is unstable API and may change in future."
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#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
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#endif
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#include <gst/gst.h>
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/**
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* SECTION:webrtc_fwd.h
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* @title: GstWebRTC Enumerations
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*/
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#ifndef GST_WEBRTC_API
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# ifdef BUILDING_GST_WEBRTC
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# define GST_WEBRTC_API GST_API_EXPORT /* from config.h */
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# else
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# define GST_WEBRTC_API GST_API_IMPORT
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# endif
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#endif
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#include <gst/webrtc/webrtc-enumtypes.h>
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/**
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* GstWebRTCDTLSTransport:
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*/
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typedef struct _GstWebRTCDTLSTransport GstWebRTCDTLSTransport;
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typedef struct _GstWebRTCDTLSTransportClass GstWebRTCDTLSTransportClass;
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/**
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* GstWebRTCICETransport:
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*/
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typedef struct _GstWebRTCICETransport GstWebRTCICETransport;
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typedef struct _GstWebRTCICETransportClass GstWebRTCICETransportClass;
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/**
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* GstWebRTCRTPReceiver:
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*
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* An object to track the receiving aspect of the stream
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*
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* Mostly matches the WebRTC RTCRtpReceiver interface.
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*/
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typedef struct _GstWebRTCRTPReceiver GstWebRTCRTPReceiver;
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typedef struct _GstWebRTCRTPReceiverClass GstWebRTCRTPReceiverClass;
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/**
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* GstWebRTCRTPSender:
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*
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* An object to track the sending aspect of the stream
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*
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* Mostly matches the WebRTC RTCRtpSender interface.
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*/
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typedef struct _GstWebRTCRTPSender GstWebRTCRTPSender;
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typedef struct _GstWebRTCRTPSenderClass GstWebRTCRTPSenderClass;
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typedef struct _GstWebRTCSessionDescription GstWebRTCSessionDescription;
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/**
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* GstWebRTCRTPTransceiver:
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*
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* Mostly matches the WebRTC RTCRtpTransceiver interface.
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*/
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typedef struct _GstWebRTCRTPTransceiver GstWebRTCRTPTransceiver;
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typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
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/**
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* GstWebRTCDataChannel:
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*
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* Since: 1.18
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*/
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typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
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typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
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typedef struct _GstWebRTCSCTPTransport GstWebRTCSCTPTransport;
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typedef struct _GstWebRTCSCTPTransportClass GstWebRTCSCTPTransportClass;
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/**
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* GstWebRTCDTLSTransportState:
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* @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
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* @GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
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* @GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
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* @GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
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* @GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
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*/
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typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
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{
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GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW,
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED,
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GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED,
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING,
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED,
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} GstWebRTCDTLSTransportState;
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/**
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* GstWebRTCICEGatheringState:
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* @GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
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* @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
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* @GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
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*
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* See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate>
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*/
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typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
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{
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GST_WEBRTC_ICE_GATHERING_STATE_NEW,
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GST_WEBRTC_ICE_GATHERING_STATE_GATHERING,
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GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE,
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} GstWebRTCICEGatheringState; /*< underscore_name=gst_webrtc_ice_gathering_state >*/
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/**
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* GstWebRTCICEConnectionState:
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* @GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
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* @GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
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* @GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
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* @GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
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* @GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
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* @GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
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* @GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
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*
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* See <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate>
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*/
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typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
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{
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GST_WEBRTC_ICE_CONNECTION_STATE_NEW,
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GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING,
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GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED,
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GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED,
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GST_WEBRTC_ICE_CONNECTION_STATE_FAILED,
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GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED,
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GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED,
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} GstWebRTCICEConnectionState;
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/**
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* GstWebRTCSignalingState:
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* @GST_WEBRTC_SIGNALING_STATE_STABLE: stable
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* @GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
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* @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
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* @GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
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* @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
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* @GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
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*
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* See <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate>
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*/
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typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
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{
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GST_WEBRTC_SIGNALING_STATE_STABLE,
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GST_WEBRTC_SIGNALING_STATE_CLOSED,
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GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER,
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GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER,
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GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER,
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GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER,
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} GstWebRTCSignalingState;
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/**
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* GstWebRTCPeerConnectionState:
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* @GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
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* @GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
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* @GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
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* @GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
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* @GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
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* @GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
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*
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* See <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate>
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*/
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typedef enum /*< underscore_name=gst_webrtc_peer_connection_state >*/
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{
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GST_WEBRTC_PEER_CONNECTION_STATE_NEW,
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GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING,
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GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED,
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GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED,
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GST_WEBRTC_PEER_CONNECTION_STATE_FAILED,
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GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED,
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} GstWebRTCPeerConnectionState;
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/**
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* GstWebRTCICERole:
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* @GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
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* @GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
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*/
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typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
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{
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GST_WEBRTC_ICE_ROLE_CONTROLLED,
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GST_WEBRTC_ICE_ROLE_CONTROLLING,
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} GstWebRTCICERole;
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/**
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* GstWebRTCICEComponent:
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* @GST_WEBRTC_ICE_COMPONENT_RTP: RTP component
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* @GST_WEBRTC_ICE_COMPONENT_RTCP: RTCP component
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*/
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typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
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{
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GST_WEBRTC_ICE_COMPONENT_RTP,
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GST_WEBRTC_ICE_COMPONENT_RTCP,
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} GstWebRTCICEComponent;
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/**
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* GstWebRTCSDPType:
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* @GST_WEBRTC_SDP_TYPE_OFFER: offer
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* @GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
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* @GST_WEBRTC_SDP_TYPE_ANSWER: answer
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* @GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
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*
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* See <http://w3c.github.io/webrtc-pc/#rtcsdptype>
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*/
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typedef enum /*< underscore_name=gst_webrtc_sdp_type >*/
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{
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GST_WEBRTC_SDP_TYPE_OFFER = 1,
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GST_WEBRTC_SDP_TYPE_PRANSWER,
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GST_WEBRTC_SDP_TYPE_ANSWER,
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GST_WEBRTC_SDP_TYPE_ROLLBACK,
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} GstWebRTCSDPType;
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/**
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* GstWebRTCRTPTransceiverDirection:
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* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none
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* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive
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* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly
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* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly
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* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv
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*/
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typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
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{
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE,
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY,
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY,
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV,
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} GstWebRTCRTPTransceiverDirection;
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/**
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* GstWebRTCDTLSSetup:
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* @GST_WEBRTC_DTLS_SETUP_NONE: none
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* @GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
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* @GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
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* @GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
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*/
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typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
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{
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GST_WEBRTC_DTLS_SETUP_NONE,
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GST_WEBRTC_DTLS_SETUP_ACTPASS,
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GST_WEBRTC_DTLS_SETUP_ACTIVE,
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GST_WEBRTC_DTLS_SETUP_PASSIVE,
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} GstWebRTCDTLSSetup;
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/**
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* GstWebRTCStatsType:
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* @GST_WEBRTC_STATS_CODEC: codec
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* @GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
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* @GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
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* @GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
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* @GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
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* @GST_WEBRTC_STATS_CSRC: csrc
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* @GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
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* @GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
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* @GST_WEBRTC_STATS_STREAM: stream
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* @GST_WEBRTC_STATS_TRANSPORT: transport
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* @GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
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* @GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
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* @GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
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* @GST_WEBRTC_STATS_CERTIFICATE: certificate
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*/
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typedef enum /*< underscore_name=gst_webrtc_stats_type >*/
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{
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GST_WEBRTC_STATS_CODEC = 1,
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GST_WEBRTC_STATS_INBOUND_RTP,
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GST_WEBRTC_STATS_OUTBOUND_RTP,
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GST_WEBRTC_STATS_REMOTE_INBOUND_RTP,
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GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP,
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GST_WEBRTC_STATS_CSRC,
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GST_WEBRTC_STATS_PEER_CONNECTION,
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GST_WEBRTC_STATS_DATA_CHANNEL,
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GST_WEBRTC_STATS_STREAM,
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GST_WEBRTC_STATS_TRANSPORT,
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GST_WEBRTC_STATS_CANDIDATE_PAIR,
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GST_WEBRTC_STATS_LOCAL_CANDIDATE,
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GST_WEBRTC_STATS_REMOTE_CANDIDATE,
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GST_WEBRTC_STATS_CERTIFICATE,
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} GstWebRTCStatsType;
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/**
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* GstWebRTCFECType:
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* @GST_WEBRTC_FEC_TYPE_NONE: none
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* @GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red
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*
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* Since: 1.14.1
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*/
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typedef enum /*< underscore_name=gst_webrtc_fec_type >*/
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{
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GST_WEBRTC_FEC_TYPE_NONE,
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GST_WEBRTC_FEC_TYPE_ULP_RED,
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} GstWebRTCFECType;
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/**
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* GstWebRTCSCTPTransportState:
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* @GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new
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* @GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting
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* @GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
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* @GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
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*
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* See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate>
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*
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* Since: 1.16
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*/
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typedef enum /*< underscore_name=gst_webrtc_sctp_transport_state >*/
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{
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GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
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GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING,
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GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED,
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GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED,
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} GstWebRTCSCTPTransportState;
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/**
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* GstWebRTCPriorityType:
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* @GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low
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* @GST_WEBRTC_PRIORITY_TYPE_LOW: low
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* @GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
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* @GST_WEBRTC_PRIORITY_TYPE_HIGH: high
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*
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* See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
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*
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* Since: 1.16
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*/
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typedef enum /*< underscore_name=gst_webrtc_priority_type >*/
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{
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GST_WEBRTC_PRIORITY_TYPE_VERY_LOW = 1,
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GST_WEBRTC_PRIORITY_TYPE_LOW,
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GST_WEBRTC_PRIORITY_TYPE_MEDIUM,
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GST_WEBRTC_PRIORITY_TYPE_HIGH,
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} GstWebRTCPriorityType;
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/**
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* GstWebRTCDataChannelState:
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* @GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
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* @GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
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* @GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
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* @GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
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* @GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
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*
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* See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate>
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*
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* Since: 1.16
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*/
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typedef enum /*< underscore_name=gst_webrtc_data_channel_state >*/
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{
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GST_WEBRTC_DATA_CHANNEL_STATE_NEW,
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GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING,
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GST_WEBRTC_DATA_CHANNEL_STATE_OPEN,
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GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING,
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GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED,
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} GstWebRTCDataChannelState;
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/**
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* GstWebRTCBundlePolicy:
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* @GST_WEBRTC_BUNDLE_POLICY_NONE: none
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* @GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
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* @GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
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* @GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
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*
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* See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
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* for more information.
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*
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* Since: 1.16
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*/
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typedef enum /*<underscore_name=gst_webrtc_bundle_policy>*/
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{
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GST_WEBRTC_BUNDLE_POLICY_NONE,
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GST_WEBRTC_BUNDLE_POLICY_BALANCED,
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GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT,
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GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE,
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} GstWebRTCBundlePolicy;
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|
|
|
/**
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|
* GstWebRTCICETransportPolicy:
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* @GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
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|
* @GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
|
|
*
|
|
* See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
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* for more information.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
typedef enum /*<underscore_name=gst_webrtc_ice_transport_policy>*/
|
|
{
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|
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL,
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|
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY,
|
|
} GstWebRTCICETransportPolicy;
|
|
|
|
/**
|
|
* GstWebRTCKind:
|
|
* @GST_WEBRTC_KIND_UNKNOWN: Kind has not yet been set
|
|
* @GST_WEBRTC_KIND_AUDIO: Kind is audio
|
|
* @GST_WEBRTC_KIND_VIDEO: Kind is audio
|
|
*
|
|
* https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
typedef enum /*<underscore_name=gst_webrtc_kind>*/
|
|
{
|
|
GST_WEBRTC_KIND_UNKNOWN,
|
|
GST_WEBRTC_KIND_AUDIO,
|
|
GST_WEBRTC_KIND_VIDEO,
|
|
} GstWebRTCKind;
|
|
|
|
#endif /* __GST_WEBRTC_FWD_H__ */
|