gst-libs/gst/webrtc/webrtc_fwd.h

   1 /* GStreamer
   2  * Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
   3  *
   4  * This library is free software; you can redistribute it and/or
   5  * modify it under the terms of the GNU Library General Public
   6  * License as published by the Free Software Foundation; either
   7  * version 2 of the License, or (at your option) any later version.
   8  *
   9  * This library is distributed in the hope that it will be useful,
  10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  12  * Library General Public License for more details.
  13  *
  14  * You should have received a copy of the GNU Library General Public
  15  * License along with this library; if not, write to the
  16  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
  17  * Boston, MA 02110-1301, USA.
  18  */
  19
  20 #ifndef __GST_WEBRTC_FWD_H__
  21 #define __GST_WEBRTC_FWD_H__
  22
  23 #ifndef GST_USE_UNSTABLE_API
  24 #warning "The WebRTC library from gst-plugins-bad is unstable API and may change in future."
  25 #warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
  26 #endif
  27
  28 #include <gst/gst.h>
  29 #include <gst/webrtc/webrtc-enumtypes.h>
  30
  31 typedef struct _GstWebRTCDTLSTransport GstWebRTCDTLSTransport;
  32 typedef struct _GstWebRTCDTLSTransportClass GstWebRTCDTLSTransportClass;
  33
  34 typedef struct _GstWebRTCICETransport GstWebRTCICETransport;
  35 typedef struct _GstWebRTCICETransportClass GstWebRTCICETransportClass;
  36
  37 typedef struct _GstWebRTCRTPReceiver GstWebRTCRTPReceiver;
  38 typedef struct _GstWebRTCRTPReceiverClass GstWebRTCRTPReceiverClass;
  39
  40 typedef struct _GstWebRTCRTPSender GstWebRTCRTPSender;
  41 typedef struct _GstWebRTCRTPSenderClass GstWebRTCRTPSenderClass;
  42
  43 typedef struct _GstWebRTCSessionDescription GstWebRTCSessionDescription;
  44
  45 typedef struct _GstWebRTCRTPTransceiver GstWebRTCRTPTransceiver;
  46 typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
  47
  48 /**
  49  * GstWebRTCDTLSTransportState:
  50  * GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
  51  * GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
  52  * GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
  53  * GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
  54  * GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
  55  */
  56 typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
  57 {
  58   GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW,
  59   GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED,
  60   GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED,
  61   GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING,
  62   GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED,
  63 } GstWebRTCDTLSTransportState;
  64
  65 /**
  66  * GstWebRTCICEGatheringState:
  67  * GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
  68  * GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
  69  * GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
  70  *
  71  * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink>
  72  */
  73 typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
  74 {
  75   GST_WEBRTC_ICE_GATHERING_STATE_NEW,
  76   GST_WEBRTC_ICE_GATHERING_STATE_GATHERING,
  77   GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE,
  78 } GstWebRTCICEGatheringState; /*< underscore_name=gst_webrtc_ice_gathering_state >*/
  79
  80 /**
  81  * GstWebRTCICEConnectionState:
  82  * GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
  83  * GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
  84  * GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
  85  * GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
  86  * GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
  87  * GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
  88  * GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
  89  *
  90  * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink>
  91  */
  92 typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
  93 {
  94   GST_WEBRTC_ICE_CONNECTION_STATE_NEW,
  95   GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING,
  96   GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED,
  97   GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED,
  98   GST_WEBRTC_ICE_CONNECTION_STATE_FAILED,
  99   GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED,
 100   GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED,
 101 } GstWebRTCICEConnectionState;
 102
 103 /**
 104  * GstWebRTCSignalingState:
 105  * GST_WEBRTC_SIGNALING_STATE_STABLE: stable
 106  * GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
 107  * GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
 108  * GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
 109  * GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
 110  * GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
 111  *
 112  * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink>
 113  */
 114 typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
 115 {
 116   GST_WEBRTC_SIGNALING_STATE_STABLE,
 117   GST_WEBRTC_SIGNALING_STATE_CLOSED,
 118   GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER,
 119   GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER,
 120   GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER,
 121   GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER,
 122 } GstWebRTCSignalingState;
 123
 124 /**
 125  * GstWebRTCPeerConnectionState:
 126  * GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
 127  * GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
 128  * GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
 129  * GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
 130  * GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
 131  * GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
 132  *
 133  * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink>
 134  */
 135 typedef enum /*< underscore_name=gst_webrtc_peer_connection_state >*/
 136 {
 137   GST_WEBRTC_PEER_CONNECTION_STATE_NEW,
 138   GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING,
 139   GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED,
 140   GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED,
 141   GST_WEBRTC_PEER_CONNECTION_STATE_FAILED,
 142   GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED,
 143 } GstWebRTCPeerConnectionState;
 144
 145 /**
 146  * GstWebRTCIceRole:
 147  * GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
 148  * GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
 149  */
 150 typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
 151 {
 152   GST_WEBRTC_ICE_ROLE_CONTROLLED,
 153   GST_WEBRTC_ICE_ROLE_CONTROLLING,
 154 } GstWebRTCIceRole;
 155
 156 /**
 157  * GstWebRTCIceComponent:
 158  * GST_WEBRTC_ICE_COMPONENT_RTP,
 159  * GST_WEBRTC_ICE_COMPONENT_RTCP,
 160  */
 161 typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
 162 {
 163   GST_WEBRTC_ICE_COMPONENT_RTP,
 164   GST_WEBRTC_ICE_COMPONENT_RTCP,
 165 } GstWebRTCICEComponent;
 166
 167 /**
 168  * GstWebRTCSDPType:
 169  * GST_WEBRTC_SDP_TYPE_OFFER: offer
 170  * GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
 171  * GST_WEBRTC_SDP_TYPE_ANSWER: answer
 172  * GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
 173  *
 174  * See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink>
 175  */
 176 typedef enum /*< underscore_name=gst_webrtc_sdp_type >*/
 177 {
 178   GST_WEBRTC_SDP_TYPE_OFFER = 1,
 179   GST_WEBRTC_SDP_TYPE_PRANSWER,
 180   GST_WEBRTC_SDP_TYPE_ANSWER,
 181   GST_WEBRTC_SDP_TYPE_ROLLBACK,
 182 } GstWebRTCSDPType;
 183
 184 /**
 185  * GstWebRTCRtpTransceiverDirection:
 186  * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none
 187  * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive
 188  * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly
 189  * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly
 190  * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv
 191  */
 192 typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
 193 {
 194   GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
 195   GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE,
 196   GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY,
 197   GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY,
 198   GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV,
 199 } GstWebRTCRTPTransceiverDirection;
 200
 201 /**
 202  * GstWebRTCDTLSSetup:
 203  * GST_WEBRTC_DTLS_SETUP_NONE: none
 204  * GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
 205  * GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
 206  * GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
 207  */
 208 typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
 209 {
 210   GST_WEBRTC_DTLS_SETUP_NONE,
 211   GST_WEBRTC_DTLS_SETUP_ACTPASS,
 212   GST_WEBRTC_DTLS_SETUP_ACTIVE,
 213   GST_WEBRTC_DTLS_SETUP_PASSIVE,
 214 } GstWebRTCDTLSSetup;
 215
 216 /**
 217  * GstWebRTCStatsType:
 218  * GST_WEBRTC_STATS_CODEC: codec
 219  * GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
 220  * GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
 221  * GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
 222  * GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
 223  * GST_WEBRTC_STATS_CSRC: csrc
 224  * GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
 225  * GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
 226  * GST_WEBRTC_STATS_STREAM: stream
 227  * GST_WEBRTC_STATS_TRANSPORT: transport
 228  * GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
 229  * GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
 230  * GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
 231  * GST_WEBRTC_STATS_CERTIFICATE: certificate
 232  */
 233 typedef enum /*< underscore_name=gst_webrtc_stats_type >*/
 234 {
 235   GST_WEBRTC_STATS_CODEC = 1,
 236   GST_WEBRTC_STATS_INBOUND_RTP,
 237   GST_WEBRTC_STATS_OUTBOUND_RTP,
 238   GST_WEBRTC_STATS_REMOTE_INBOUND_RTP,
 239   GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP,
 240   GST_WEBRTC_STATS_CSRC,
 241   GST_WEBRTC_STATS_PEER_CONNECTION,
 242   GST_WEBRTC_STATS_DATA_CHANNEL,
 243   GST_WEBRTC_STATS_STREAM,
 244   GST_WEBRTC_STATS_TRANSPORT,
 245   GST_WEBRTC_STATS_CANDIDATE_PAIR,
 246   GST_WEBRTC_STATS_LOCAL_CANDIDATE,
 247   GST_WEBRTC_STATS_REMOTE_CANDIDATE,
 248   GST_WEBRTC_STATS_CERTIFICATE,
 249 } GstWebRTCStatsType;
 250
 251 #endif /* __GST_WEBRTC_FWD_H__ */