diff options
Diffstat (limited to 'include/gst/webrtc/webrtc_fwd.h')
| -rw-r--r-- | include/gst/webrtc/webrtc_fwd.h | 467 |
1 files changed, 467 insertions, 0 deletions
diff --git a/include/gst/webrtc/webrtc_fwd.h b/include/gst/webrtc/webrtc_fwd.h new file mode 100644 index 0000000000..d181873884 --- /dev/null +++ b/include/gst/webrtc/webrtc_fwd.h @@ -0,0 +1,467 @@ +/* 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> + +/** + * SECTION:webrtc_fwd.h + * @title: GstWebRTC Enumerations + */ + +#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 + +#include <gst/webrtc/webrtc-enumtypes.h> + +/** + * GstWebRTCDTLSTransport: + */ +typedef struct _GstWebRTCDTLSTransport GstWebRTCDTLSTransport; +typedef struct _GstWebRTCDTLSTransportClass GstWebRTCDTLSTransportClass; + +/** + * 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; + +/** + * GstWebRTCICEGatheringState: + * @GST_WEBRTC_ICE_GATHERING_STATE_NEW: new + * @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering + * @GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete + * + * See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate> + */ +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 <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate> + */ +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 <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate> + */ +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 <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate> + */ +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 <http://w3c.github.io/webrtc-pc/#rtcsdptype> + */ +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 + * + * 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 <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate> + * + * 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 <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype> + * + * 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_NEW: new + * @GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection + * @GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open + * @GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing + * @GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed + * + * See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate> + * + * Since: 1.16 + */ +typedef enum /*< underscore_name=gst_webrtc_data_channel_state >*/ +{ + GST_WEBRTC_DATA_CHANNEL_STATE_NEW, + GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING, + 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 /*<underscore_name=gst_webrtc_bundle_policy>*/ +{ + 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 /*<underscore_name=gst_webrtc_ice_transport_policy>*/ +{ + 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 audio + * + * https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind + * + * Since: 1.20 + */ +typedef enum /*<underscore_name=gst_webrtc_kind>*/ +{ + 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 <https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype> for more information. + * + * Since: 1.20 + */ +typedef enum /*<underscore_name=gst_webrtc_error>*/ +{ + 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 +} GstWebRTCError; + + +#endif /* __GST_WEBRTC_FWD_H__ */ |