gstreamer/subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h
Thibault Saunier 050006e9f0 webrtc: doc: Cleanup Since markers
We hide some newly public symbols when moving some objects to the library
which was not really correct in the end. Properly add the Since markers
as needed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2891>
2022-08-18 08:34:36 +00:00

495 lines
15 KiB
C

/* 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;
/**
* GstWebRTCICE:
*
* Since: 1.22
*/
typedef struct _GstWebRTCICE GstWebRTCICE;
typedef struct _GstWebRTCICEClass GstWebRTCICEClass;
/**
* GstWebRTCICECandidateStats:
*
* Since: 1.22
*/
typedef struct _GstWebRTCICECandidateStats GstWebRTCICECandidateStats;
/**
* 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;
/**
* 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_CONNECTING: connecting
* @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_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 /*<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
*/
/**
* GST_WEBRTC_ERROR_INVALID_MODIFICATION:
*
* invalid-modification (part of WebIDL specification)
*
* Since: 1.22
*/
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,
GST_WEBRTC_ERROR_INVALID_MODIFICATION,
} GstWebRTCError;
#endif /* __GST_WEBRTC_FWD_H__ */