mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-06-27 10:20:42 +00:00
6538ebbaf3
Fix a typo of peer-connectiion -> peer-connection Add a link to the w3c RTCStats type for a description of what each statistics type is. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>
520 lines
16 KiB
C
520 lines
16 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
|
|
|
|
/**
|
|
* 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
|
|
|
|
#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-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 <https://w3c.github.io/webrtc-stats/#dom-rtcstatstype>
|
|
*/
|
|
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
|
|
*/
|
|
/**
|
|
* GST_WEBRTC_ERROR_TYPE_ERROR:
|
|
*
|
|
* type-error (maps to JavaScript TypeError)
|
|
*
|
|
* 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,
|
|
GST_WEBRTC_ERROR_TYPE_ERROR,
|
|
} GstWebRTCError;
|
|
|
|
#endif /* __GST_WEBRTC_FWD_H__ */
|