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1894293d63
SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
251 lines
9.1 KiB
C
251 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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#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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