| /* GStreamer |
| * Copyright (C) 2017 Matthew Waters <matthew@centricular.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| #ifndef __GST_WEBRTC_FWD_H__ |
| #define __GST_WEBRTC_FWD_H__ |
| |
| #ifndef GST_USE_UNSTABLE_API |
| #warning "The WebRTC library from gst-plugins-bad is unstable API and may change in future." |
| #warning "You can define GST_USE_UNSTABLE_API to avoid this warning." |
| #endif |
| |
| #include <gst/gst.h> |
| |
| #ifndef GST_WEBRTC_API |
| #define GST_WEBRTC_API GST_EXPORT |
| #endif |
| |
| #include <gst/webrtc/webrtc-enumtypes.h> |
| |
| typedef struct _GstWebRTCDTLSTransport GstWebRTCDTLSTransport; |
| typedef struct _GstWebRTCDTLSTransportClass GstWebRTCDTLSTransportClass; |
| |
| typedef struct _GstWebRTCICETransport GstWebRTCICETransport; |
| typedef struct _GstWebRTCICETransportClass GstWebRTCICETransportClass; |
| |
| typedef struct _GstWebRTCRTPReceiver GstWebRTCRTPReceiver; |
| typedef struct _GstWebRTCRTPReceiverClass GstWebRTCRTPReceiverClass; |
| |
| typedef struct _GstWebRTCRTPSender GstWebRTCRTPSender; |
| typedef struct _GstWebRTCRTPSenderClass GstWebRTCRTPSenderClass; |
| |
| typedef struct _GstWebRTCSessionDescription GstWebRTCSessionDescription; |
| |
| typedef struct _GstWebRTCRTPTransceiver GstWebRTCRTPTransceiver; |
| typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass; |
| |
| /** |
| * GstWebRTCDTLSTransportState: |
| * GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new |
| * GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed |
| * GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed |
| * GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting |
| * GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/ |
| { |
| GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW, |
| GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED, |
| GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED, |
| GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING, |
| GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED, |
| } GstWebRTCDTLSTransportState; |
| |
| /** |
| * GstWebRTCICEGatheringState: |
| * GST_WEBRTC_ICE_GATHERING_STATE_NEW: new |
| * GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering |
| * GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete |
| * |
| * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink> |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/ |
| { |
| GST_WEBRTC_ICE_GATHERING_STATE_NEW, |
| GST_WEBRTC_ICE_GATHERING_STATE_GATHERING, |
| GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE, |
| } GstWebRTCICEGatheringState; /*< underscore_name=gst_webrtc_ice_gathering_state >*/ |
| |
| /** |
| * GstWebRTCICEConnectionState: |
| * GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new |
| * GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking |
| * GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected |
| * GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed |
| * GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed |
| * GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected |
| * GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed |
| * |
| * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink> |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/ |
| { |
| GST_WEBRTC_ICE_CONNECTION_STATE_NEW, |
| GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING, |
| GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED, |
| GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED, |
| GST_WEBRTC_ICE_CONNECTION_STATE_FAILED, |
| GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED, |
| GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED, |
| } GstWebRTCICEConnectionState; |
| |
| /** |
| * GstWebRTCSignalingState: |
| * GST_WEBRTC_SIGNALING_STATE_STABLE: stable |
| * GST_WEBRTC_SIGNALING_STATE_CLOSED: closed |
| * GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer |
| * GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer |
| * GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer |
| * GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer |
| * |
| * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink> |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/ |
| { |
| GST_WEBRTC_SIGNALING_STATE_STABLE, |
| GST_WEBRTC_SIGNALING_STATE_CLOSED, |
| GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER, |
| GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER, |
| GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER, |
| GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER, |
| } GstWebRTCSignalingState; |
| |
| /** |
| * GstWebRTCPeerConnectionState: |
| * GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new |
| * GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting |
| * GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected |
| * GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected |
| * GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed |
| * GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed |
| * |
| * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink> |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_peer_connection_state >*/ |
| { |
| GST_WEBRTC_PEER_CONNECTION_STATE_NEW, |
| GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING, |
| GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED, |
| GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED, |
| GST_WEBRTC_PEER_CONNECTION_STATE_FAILED, |
| GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED, |
| } GstWebRTCPeerConnectionState; |
| |
| /** |
| * GstWebRTCICERole: |
| * GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled |
| * GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_ice_role >*/ |
| { |
| GST_WEBRTC_ICE_ROLE_CONTROLLED, |
| GST_WEBRTC_ICE_ROLE_CONTROLLING, |
| } GstWebRTCICERole; |
| |
| /** |
| * GstWebRTCICEComponent: |
| * GST_WEBRTC_ICE_COMPONENT_RTP, |
| * GST_WEBRTC_ICE_COMPONENT_RTCP, |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_ice_component >*/ |
| { |
| GST_WEBRTC_ICE_COMPONENT_RTP, |
| GST_WEBRTC_ICE_COMPONENT_RTCP, |
| } GstWebRTCICEComponent; |
| |
| /** |
| * GstWebRTCSDPType: |
| * GST_WEBRTC_SDP_TYPE_OFFER: offer |
| * GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer |
| * GST_WEBRTC_SDP_TYPE_ANSWER: answer |
| * GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback |
| * |
| * See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink> |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_sdp_type >*/ |
| { |
| GST_WEBRTC_SDP_TYPE_OFFER = 1, |
| GST_WEBRTC_SDP_TYPE_PRANSWER, |
| GST_WEBRTC_SDP_TYPE_ANSWER, |
| GST_WEBRTC_SDP_TYPE_ROLLBACK, |
| } GstWebRTCSDPType; |
| |
| /** |
| * GstWebRTCRtpTransceiverDirection: |
| * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none |
| * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive |
| * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly |
| * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly |
| * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/ |
| { |
| GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE, |
| GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE, |
| GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, |
| GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, |
| GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, |
| } GstWebRTCRTPTransceiverDirection; |
| |
| /** |
| * GstWebRTCDTLSSetup: |
| * GST_WEBRTC_DTLS_SETUP_NONE: none |
| * GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass |
| * GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly |
| * GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/ |
| { |
| GST_WEBRTC_DTLS_SETUP_NONE, |
| GST_WEBRTC_DTLS_SETUP_ACTPASS, |
| GST_WEBRTC_DTLS_SETUP_ACTIVE, |
| GST_WEBRTC_DTLS_SETUP_PASSIVE, |
| } GstWebRTCDTLSSetup; |
| |
| /** |
| * GstWebRTCStatsType: |
| * GST_WEBRTC_STATS_CODEC: codec |
| * GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp |
| * GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp |
| * GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp |
| * GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp |
| * GST_WEBRTC_STATS_CSRC: csrc |
| * GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion |
| * GST_WEBRTC_STATS_DATA_CHANNEL: data-channel |
| * GST_WEBRTC_STATS_STREAM: stream |
| * GST_WEBRTC_STATS_TRANSPORT: transport |
| * GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair |
| * GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate |
| * GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate |
| * GST_WEBRTC_STATS_CERTIFICATE: certificate |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_stats_type >*/ |
| { |
| GST_WEBRTC_STATS_CODEC = 1, |
| GST_WEBRTC_STATS_INBOUND_RTP, |
| GST_WEBRTC_STATS_OUTBOUND_RTP, |
| GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, |
| GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP, |
| GST_WEBRTC_STATS_CSRC, |
| GST_WEBRTC_STATS_PEER_CONNECTION, |
| GST_WEBRTC_STATS_DATA_CHANNEL, |
| GST_WEBRTC_STATS_STREAM, |
| GST_WEBRTC_STATS_TRANSPORT, |
| GST_WEBRTC_STATS_CANDIDATE_PAIR, |
| GST_WEBRTC_STATS_LOCAL_CANDIDATE, |
| GST_WEBRTC_STATS_REMOTE_CANDIDATE, |
| GST_WEBRTC_STATS_CERTIFICATE, |
| } GstWebRTCStatsType; |
| |
| /** |
| * GstWebRTCFECType: |
| * GST_WEBRTC_FEC_TYPE_NONE: none |
| * GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red |
| */ |
| typedef enum /*< underscore_name=gst_webrtc_fec_type >*/ |
| { |
| GST_WEBRTC_FEC_TYPE_NONE, |
| GST_WEBRTC_FEC_TYPE_ULP_RED, |
| } GstWebRTCFECType; |
| |
| #endif /* __GST_WEBRTC_FWD_H__ */ |
