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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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333f636555
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
256 lines
9.1 KiB
C
256 lines
9.1 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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#ifndef GST_WEBRTC_API
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#define GST_WEBRTC_API GST_EXPORT
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#endif
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#include <gst/webrtc/webrtc-enumtypes.h>
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typedef struct _GstWebRTCDTLSTransport GstWebRTCDTLSTransport;
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typedef struct _GstWebRTCDTLSTransportClass GstWebRTCDTLSTransportClass;
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typedef struct _GstWebRTCICETransport GstWebRTCICETransport;
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typedef struct _GstWebRTCICETransportClass GstWebRTCICETransportClass;
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typedef struct _GstWebRTCRTPReceiver GstWebRTCRTPReceiver;
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typedef struct _GstWebRTCRTPReceiverClass GstWebRTCRTPReceiverClass;
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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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typedef struct _GstWebRTCRTPTransceiver GstWebRTCRTPTransceiver;
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typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
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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 <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink>
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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 <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink>
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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 <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink>
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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 <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink>
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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,
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* GST_WEBRTC_ICE_COMPONENT_RTCP,
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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 <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink>
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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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#endif /* __GST_WEBRTC_FWD_H__ */
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