/* GStreamer * Copyright (C) 2017 Matthew Waters * * 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 /** * SECTION:webrtc_fwd.h * @title: GstWebRTC Enumerations */ G_BEGIN_DECLS #ifndef GST_WEBRTC_API # ifdef BUILDING_GST_WEBRTC # define GST_WEBRTC_API GST_API_EXPORT /* from config.h */ # else # define GST_WEBRTC_API GST_API_IMPORT # endif #endif /** * GST_WEBRTC_DEPRECATED: (attributes doc.skip=true) */ /** * GST_WEBRTC_DEPRECATED_FOR: (attributes doc.skip=true) */ #ifndef GST_DISABLE_DEPRECATED #define GST_WEBRTC_DEPRECATED GST_WEBRTC_API #define GST_WEBRTC_DEPRECATED_FOR(f) GST_WEBRTC_API #else #define GST_WEBRTC_DEPRECATED G_DEPRECATED GST_WEBRTC_API #define GST_WEBRTC_DEPRECATED_FOR(f) G_DEPRECATED_FOR(f) GST_WEBRTC_API #endif /** * GstWebRTCDTLSTransport: */ typedef struct _GstWebRTCDTLSTransport GstWebRTCDTLSTransport; typedef struct _GstWebRTCDTLSTransportClass GstWebRTCDTLSTransportClass; /** * GstWebRTCICE: * * Since: 1.22 */ typedef struct _GstWebRTCICE GstWebRTCICE; typedef struct _GstWebRTCICEClass GstWebRTCICEClass; typedef struct _GstWebRTCICECandidate GstWebRTCICECandidate; /** * GstWebRTCICECandidatePair: * * Since: 1.28 */ typedef struct _GstWebRTCICECandidatePair GstWebRTCICECandidatePair; typedef struct _GstWebRTCICECandidateStats GstWebRTCICECandidateStats; /** * GstWebRTCICETcpCandidateType: * @GST_WEBRTC_ICE_TCP_CANDIDATE_TYPE_ACTIVE: An "active" TCP candidate is one for which the transport * will attempt to open an outbound connection but will not * receive incoming connection requests. * @GST_WEBRTC_ICE_TCP_CANDIDATE_TYPE_PASSIVE: A "passive" TCP candidate is one for which the transport * will receive incoming connection attempts but not attempt * a connection. * @GST_WEBRTC_ICE_TCP_CANDIDATE_TYPE_SO: An "so" candidate is one for which the transport will attempt * to open a connection simultaneously with its peer. * @GST_WEBRTC_ICE_TCP_CANDIDATE_TYPE_NONE: Value used for non-TCP candidate type. * * Since: 1.28 */ typedef enum /*< underscore_name=gst_webrtc_ice_tcp_candidate_type >*/ { GST_WEBRTC_ICE_TCP_CANDIDATE_TYPE_ACTIVE, GST_WEBRTC_ICE_TCP_CANDIDATE_TYPE_PASSIVE, GST_WEBRTC_ICE_TCP_CANDIDATE_TYPE_SO, GST_WEBRTC_ICE_TCP_CANDIDATE_TYPE_NONE, } GstWebRTCICETcpCandidateType; /** * GstWebRTCICECandidateProtocolType: * @GST_WEBRTC_ICE_CANDIDATE_PROTOCOL_TYPE_TCP: * @GST_WEBRTC_ICE_CANDIDATE_PROTOCOL_TYPE_UDP: * * Since: 1.28 */ typedef enum /*< underscore_name=gst_webrtc_ice_candidate_protocol_type >*/ { GST_WEBRTC_ICE_CANDIDATE_PROTOCOL_TYPE_TCP, GST_WEBRTC_ICE_CANDIDATE_PROTOCOL_TYPE_UDP, } GstWebRTCICECandidateProtocolType; /** * GstWebRTCICECandidateType: * @GST_WEBRTC_ICE_CANDIDATE_TYPE_HOST: The candidate is a host candidate, whose * IP address as specified in the RTCIceCandidate.address property is in fact the * true address of the remote peer. * @GST_WEBRTC_ICE_CANDIDATE_TYPE_SERVER_REFLEXIVE: The candidate is a server * reflexive candidate; the ip and port are a binding allocated by a NAT for an * agent when it sent a packet through the NAT to a server. They can be learned by * the STUN server and TURN server to represent the candidate's peer anonymously. * @GST_WEBRTC_ICE_CANDIDATE_TYPE_PEER_REFLEXIVE: The candidate is a peer * reflexive candidate; the ip and port are a binding allocated by a NAT when it * sent a STUN request to represent the candidate's peer anonymously. * @GST_WEBRTC_ICE_CANDIDATE_TYPE_RELAYED: The candidate is a relay candidate, * obtained from a TURN server. The relay candidate's IP address is an address the * TURN server uses to forward the media between the two peers. * * Since: 1.28 */ typedef enum /*< underscore_name=gst_webrtc_ice_candidate_type >*/ { GST_WEBRTC_ICE_CANDIDATE_TYPE_HOST, GST_WEBRTC_ICE_CANDIDATE_TYPE_SERVER_REFLEXIVE, GST_WEBRTC_ICE_CANDIDATE_TYPE_PEER_REFLEXIVE, GST_WEBRTC_ICE_CANDIDATE_TYPE_RELAYED, } GstWebRTCICECandidateType; /** * GstWebRTCICEStream: * * Since: 1.22 */ typedef struct _GstWebRTCICEStream GstWebRTCICEStream; typedef struct _GstWebRTCICEStreamClass GstWebRTCICEStreamClass; /** * GstWebRTCICETransport: */ typedef struct _GstWebRTCICETransport GstWebRTCICETransport; typedef struct _GstWebRTCICETransportClass GstWebRTCICETransportClass; /** * GstWebRTCRTPReceiver: * * An object to track the receiving aspect of the stream * * Mostly matches the WebRTC RTCRtpReceiver interface. */ typedef struct _GstWebRTCRTPReceiver GstWebRTCRTPReceiver; typedef struct _GstWebRTCRTPReceiverClass GstWebRTCRTPReceiverClass; /** * GstWebRTCRTPSender: * * An object to track the sending aspect of the stream * * Mostly matches the WebRTC RTCRtpSender interface. */ typedef struct _GstWebRTCRTPSender GstWebRTCRTPSender; typedef struct _GstWebRTCRTPSenderClass GstWebRTCRTPSenderClass; typedef struct _GstWebRTCSessionDescription GstWebRTCSessionDescription; /** * GstWebRTCRTPTransceiver: * * Mostly matches the WebRTC RTCRtpTransceiver interface. */ typedef struct _GstWebRTCRTPTransceiver GstWebRTCRTPTransceiver; typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass; /** * GstWebRTCDataChannel: * * Since: 1.18 */ typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel; typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass; typedef struct _GstWebRTCSCTPTransport GstWebRTCSCTPTransport; typedef struct _GstWebRTCSCTPTransportClass GstWebRTCSCTPTransportClass; /** * 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; /** * GstWebRTCDTLSRole: * @GST_WEBRTC_DTLS_ROLE_CLIENT: client * @GST_WEBRTC_DTLS_ROLE_SERVER: server * @GST_WEBRTC_DTLS_ROLE_UNKNOWN: unknown * * Since: 1.28 */ typedef enum /*< underscore_name=gst_webrtc_dtls_role >*/ { GST_WEBRTC_DTLS_ROLE_CLIENT, GST_WEBRTC_DTLS_ROLE_SERVER, GST_WEBRTC_DTLS_ROLE_UNKNOWN } GstWebRTCDTLSRole; /** * GstWebRTCICEGatheringState: * @GST_WEBRTC_ICE_GATHERING_STATE_NEW: new * @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering * @GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete * * See */ 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 */ 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 */ 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 */ 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: RTP component * @GST_WEBRTC_ICE_COMPONENT_RTCP: RTCP component */ 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 */ 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-connection * @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 * * See */ 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 * * Since: 1.14.1 */ typedef enum /*< underscore_name=gst_webrtc_fec_type >*/ { GST_WEBRTC_FEC_TYPE_NONE, GST_WEBRTC_FEC_TYPE_ULP_RED, } GstWebRTCFECType; /** * GstWebRTCSCTPTransportState: * @GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new * @GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting * @GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected * @GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed * * See * * Since: 1.16 */ typedef enum /*< underscore_name=gst_webrtc_sctp_transport_state >*/ { GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW, GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING, GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED, GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED, } GstWebRTCSCTPTransportState; /** * GstWebRTCPriorityType: * @GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low * @GST_WEBRTC_PRIORITY_TYPE_LOW: low * @GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium * @GST_WEBRTC_PRIORITY_TYPE_HIGH: high * * See * * Since: 1.16 */ typedef enum /*< underscore_name=gst_webrtc_priority_type >*/ { GST_WEBRTC_PRIORITY_TYPE_VERY_LOW = 1, GST_WEBRTC_PRIORITY_TYPE_LOW, GST_WEBRTC_PRIORITY_TYPE_MEDIUM, GST_WEBRTC_PRIORITY_TYPE_HIGH, } GstWebRTCPriorityType; /** * GstWebRTCDataChannelState: * @GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connecting * @GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open * @GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing * @GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed * * See * * Since: 1.16 */ typedef enum /*< underscore_name=gst_webrtc_data_channel_state >*/ { GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING = 1, GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING, GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED, } GstWebRTCDataChannelState; /** * GstWebRTCBundlePolicy: * @GST_WEBRTC_BUNDLE_POLICY_NONE: none * @GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced * @GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat * @GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle * * See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 * for more information. * * Since: 1.16 */ typedef enum /**/ { GST_WEBRTC_BUNDLE_POLICY_NONE, GST_WEBRTC_BUNDLE_POLICY_BALANCED, GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT, GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE, } GstWebRTCBundlePolicy; /** * GstWebRTCICETransportPolicy: * @GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all * @GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay * * See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 * for more information. * * Since: 1.16 */ typedef enum /**/ { GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL, 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 video * * https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind * * Since: 1.20 */ typedef enum /**/ { GST_WEBRTC_KIND_UNKNOWN, GST_WEBRTC_KIND_AUDIO, GST_WEBRTC_KIND_VIDEO, } GstWebRTCKind; GST_WEBRTC_API GQuark gst_webrtc_error_quark (void); /** * GST_WEBRTC_ERROR: * * Since: 1.20 */ #define GST_WEBRTC_ERROR gst_webrtc_error_quark () /** * GstWebRTCError: * @GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE: data-channel-failure * @GST_WEBRTC_ERROR_DTLS_FAILURE: dtls-failure * @GST_WEBRTC_ERROR_FINGERPRINT_FAILURE: fingerprint-failure * @GST_WEBRTC_ERROR_SCTP_FAILURE: sctp-failure * @GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR: sdp-syntax-error * @GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE: hardware-encoder-not-available * @GST_WEBRTC_ERROR_ENCODER_ERROR: encoder-error * @GST_WEBRTC_ERROR_INVALID_STATE: invalid-state (part of WebIDL specification) * @GST_WEBRTC_ERROR_INTERNAL_FAILURE: GStreamer-specific failure, not matching any other value from the specification * * See for more information. * * Since: 1.20 */ /** * GST_WEBRTC_ERROR_INVALID_MODIFICATION: * * invalid-modification (part of WebIDL specification) * * Since: 1.22 */ /** * GST_WEBRTC_ERROR_TYPE_ERROR: * * type-error (maps to JavaScript TypeError) * * Since: 1.22 */ typedef enum /**/ { GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, GST_WEBRTC_ERROR_DTLS_FAILURE, GST_WEBRTC_ERROR_FINGERPRINT_FAILURE, GST_WEBRTC_ERROR_SCTP_FAILURE, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE, GST_WEBRTC_ERROR_ENCODER_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, GST_WEBRTC_ERROR_INTERNAL_FAILURE, GST_WEBRTC_ERROR_INVALID_MODIFICATION, GST_WEBRTC_ERROR_TYPE_ERROR, } GstWebRTCError; G_END_DECLS #endif /* __GST_WEBRTC_FWD_H__ */