forked from mirrors/gstreamer-rs
926 lines
38 KiB
Rust
926 lines
38 KiB
Rust
// This file was generated by gir (https://github.com/gtk-rs/gir)
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// from gir-files (https://github.com/gtk-rs/gir-files)
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// from gst-gir-files (https://gitlab.freedesktop.org/gstreamer/gir-files-rs.git)
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// DO NOT EDIT
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#![allow(non_camel_case_types, non_upper_case_globals, non_snake_case)]
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#![allow(
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clippy::approx_constant,
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clippy::type_complexity,
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clippy::unreadable_literal,
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clippy::upper_case_acronyms
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)]
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#![cfg_attr(docsrs, feature(doc_cfg))]
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use glib_sys as glib;
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use gstreamer_sdp_sys as gst_sdp;
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#[allow(unused_imports)]
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use libc::{
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c_char, c_double, c_float, c_int, c_long, c_short, c_uchar, c_uint, c_ulong, c_ushort, c_void,
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intptr_t, size_t, ssize_t, uintptr_t, FILE,
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};
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#[allow(unused_imports)]
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use glib::{gboolean, gconstpointer, gpointer, GType};
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// Enums
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pub type GstWebRTCBundlePolicy = c_int;
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pub const GST_WEBRTC_BUNDLE_POLICY_NONE: GstWebRTCBundlePolicy = 0;
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pub const GST_WEBRTC_BUNDLE_POLICY_BALANCED: GstWebRTCBundlePolicy = 1;
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pub const GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: GstWebRTCBundlePolicy = 2;
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pub const GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: GstWebRTCBundlePolicy = 3;
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pub type GstWebRTCDTLSSetup = c_int;
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pub const GST_WEBRTC_DTLS_SETUP_NONE: GstWebRTCDTLSSetup = 0;
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pub const GST_WEBRTC_DTLS_SETUP_ACTPASS: GstWebRTCDTLSSetup = 1;
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pub const GST_WEBRTC_DTLS_SETUP_ACTIVE: GstWebRTCDTLSSetup = 2;
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pub const GST_WEBRTC_DTLS_SETUP_PASSIVE: GstWebRTCDTLSSetup = 3;
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pub type GstWebRTCDTLSTransportState = c_int;
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pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: GstWebRTCDTLSTransportState = 0;
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pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: GstWebRTCDTLSTransportState = 1;
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pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: GstWebRTCDTLSTransportState = 2;
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pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: GstWebRTCDTLSTransportState = 3;
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pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: GstWebRTCDTLSTransportState = 4;
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pub type GstWebRTCDataChannelState = c_int;
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pub const GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: GstWebRTCDataChannelState = 1;
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pub const GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: GstWebRTCDataChannelState = 2;
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pub const GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: GstWebRTCDataChannelState = 3;
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pub const GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: GstWebRTCDataChannelState = 4;
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pub type GstWebRTCError = c_int;
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pub const GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE: GstWebRTCError = 0;
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pub const GST_WEBRTC_ERROR_DTLS_FAILURE: GstWebRTCError = 1;
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pub const GST_WEBRTC_ERROR_FINGERPRINT_FAILURE: GstWebRTCError = 2;
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pub const GST_WEBRTC_ERROR_SCTP_FAILURE: GstWebRTCError = 3;
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pub const GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR: GstWebRTCError = 4;
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pub const GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE: GstWebRTCError = 5;
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pub const GST_WEBRTC_ERROR_ENCODER_ERROR: GstWebRTCError = 6;
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pub const GST_WEBRTC_ERROR_INVALID_STATE: GstWebRTCError = 7;
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pub const GST_WEBRTC_ERROR_INTERNAL_FAILURE: GstWebRTCError = 8;
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#[cfg(feature = "v1_22")]
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#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
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pub const GST_WEBRTC_ERROR_INVALID_MODIFICATION: GstWebRTCError = 9;
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#[cfg(feature = "v1_22")]
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#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
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pub const GST_WEBRTC_ERROR_TYPE_ERROR: GstWebRTCError = 10;
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pub type GstWebRTCFECType = c_int;
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pub const GST_WEBRTC_FEC_TYPE_NONE: GstWebRTCFECType = 0;
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pub const GST_WEBRTC_FEC_TYPE_ULP_RED: GstWebRTCFECType = 1;
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pub type GstWebRTCICEComponent = c_int;
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pub const GST_WEBRTC_ICE_COMPONENT_RTP: GstWebRTCICEComponent = 0;
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pub const GST_WEBRTC_ICE_COMPONENT_RTCP: GstWebRTCICEComponent = 1;
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pub type GstWebRTCICEConnectionState = c_int;
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pub const GST_WEBRTC_ICE_CONNECTION_STATE_NEW: GstWebRTCICEConnectionState = 0;
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pub const GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: GstWebRTCICEConnectionState = 1;
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pub const GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: GstWebRTCICEConnectionState = 2;
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pub const GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: GstWebRTCICEConnectionState = 3;
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pub const GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: GstWebRTCICEConnectionState = 4;
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pub const GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: GstWebRTCICEConnectionState = 5;
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pub const GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: GstWebRTCICEConnectionState = 6;
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pub type GstWebRTCICEGatheringState = c_int;
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pub const GST_WEBRTC_ICE_GATHERING_STATE_NEW: GstWebRTCICEGatheringState = 0;
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pub const GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: GstWebRTCICEGatheringState = 1;
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pub const GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: GstWebRTCICEGatheringState = 2;
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pub type GstWebRTCICERole = c_int;
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pub const GST_WEBRTC_ICE_ROLE_CONTROLLED: GstWebRTCICERole = 0;
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pub const GST_WEBRTC_ICE_ROLE_CONTROLLING: GstWebRTCICERole = 1;
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pub type GstWebRTCICETransportPolicy = c_int;
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pub const GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: GstWebRTCICETransportPolicy = 0;
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pub const GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: GstWebRTCICETransportPolicy = 1;
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pub type GstWebRTCKind = c_int;
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pub const GST_WEBRTC_KIND_UNKNOWN: GstWebRTCKind = 0;
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pub const GST_WEBRTC_KIND_AUDIO: GstWebRTCKind = 1;
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pub const GST_WEBRTC_KIND_VIDEO: GstWebRTCKind = 2;
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pub type GstWebRTCPeerConnectionState = c_int;
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pub const GST_WEBRTC_PEER_CONNECTION_STATE_NEW: GstWebRTCPeerConnectionState = 0;
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pub const GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: GstWebRTCPeerConnectionState = 1;
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pub const GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: GstWebRTCPeerConnectionState = 2;
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pub const GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: GstWebRTCPeerConnectionState = 3;
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pub const GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: GstWebRTCPeerConnectionState = 4;
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pub const GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: GstWebRTCPeerConnectionState = 5;
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pub type GstWebRTCPriorityType = c_int;
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pub const GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: GstWebRTCPriorityType = 1;
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pub const GST_WEBRTC_PRIORITY_TYPE_LOW: GstWebRTCPriorityType = 2;
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pub const GST_WEBRTC_PRIORITY_TYPE_MEDIUM: GstWebRTCPriorityType = 3;
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pub const GST_WEBRTC_PRIORITY_TYPE_HIGH: GstWebRTCPriorityType = 4;
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pub type GstWebRTCRTPTransceiverDirection = c_int;
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pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: GstWebRTCRTPTransceiverDirection = 0;
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pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: GstWebRTCRTPTransceiverDirection = 1;
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pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: GstWebRTCRTPTransceiverDirection = 2;
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pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: GstWebRTCRTPTransceiverDirection = 3;
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pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: GstWebRTCRTPTransceiverDirection = 4;
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pub type GstWebRTCSCTPTransportState = c_int;
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pub const GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: GstWebRTCSCTPTransportState = 0;
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pub const GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: GstWebRTCSCTPTransportState = 1;
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pub const GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: GstWebRTCSCTPTransportState = 2;
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pub const GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: GstWebRTCSCTPTransportState = 3;
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pub type GstWebRTCSDPType = c_int;
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pub const GST_WEBRTC_SDP_TYPE_OFFER: GstWebRTCSDPType = 1;
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pub const GST_WEBRTC_SDP_TYPE_PRANSWER: GstWebRTCSDPType = 2;
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pub const GST_WEBRTC_SDP_TYPE_ANSWER: GstWebRTCSDPType = 3;
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pub const GST_WEBRTC_SDP_TYPE_ROLLBACK: GstWebRTCSDPType = 4;
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pub type GstWebRTCSignalingState = c_int;
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pub const GST_WEBRTC_SIGNALING_STATE_STABLE: GstWebRTCSignalingState = 0;
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pub const GST_WEBRTC_SIGNALING_STATE_CLOSED: GstWebRTCSignalingState = 1;
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pub const GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: GstWebRTCSignalingState = 2;
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pub const GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: GstWebRTCSignalingState = 3;
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pub const GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: GstWebRTCSignalingState = 4;
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pub const GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: GstWebRTCSignalingState = 5;
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pub type GstWebRTCStatsType = c_int;
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pub const GST_WEBRTC_STATS_CODEC: GstWebRTCStatsType = 1;
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pub const GST_WEBRTC_STATS_INBOUND_RTP: GstWebRTCStatsType = 2;
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pub const GST_WEBRTC_STATS_OUTBOUND_RTP: GstWebRTCStatsType = 3;
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pub const GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: GstWebRTCStatsType = 4;
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pub const GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: GstWebRTCStatsType = 5;
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pub const GST_WEBRTC_STATS_CSRC: GstWebRTCStatsType = 6;
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pub const GST_WEBRTC_STATS_PEER_CONNECTION: GstWebRTCStatsType = 7;
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pub const GST_WEBRTC_STATS_DATA_CHANNEL: GstWebRTCStatsType = 8;
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pub const GST_WEBRTC_STATS_STREAM: GstWebRTCStatsType = 9;
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pub const GST_WEBRTC_STATS_TRANSPORT: GstWebRTCStatsType = 10;
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pub const GST_WEBRTC_STATS_CANDIDATE_PAIR: GstWebRTCStatsType = 11;
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pub const GST_WEBRTC_STATS_LOCAL_CANDIDATE: GstWebRTCStatsType = 12;
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pub const GST_WEBRTC_STATS_REMOTE_CANDIDATE: GstWebRTCStatsType = 13;
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pub const GST_WEBRTC_STATS_CERTIFICATE: GstWebRTCStatsType = 14;
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// Callbacks
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pub type GstWebRTCICEOnCandidateFunc =
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Option<unsafe extern "C" fn(*mut GstWebRTCICE, c_uint, *const c_char, gpointer)>;
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// Records
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#[repr(C)]
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pub struct _GstWebRTCDTLSTransportClass {
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_data: [u8; 0],
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_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
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}
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pub type GstWebRTCDTLSTransportClass = _GstWebRTCDTLSTransportClass;
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#[repr(C)]
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pub struct _GstWebRTCDataChannelClass {
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_data: [u8; 0],
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_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
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}
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pub type GstWebRTCDataChannelClass = _GstWebRTCDataChannelClass;
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#[derive(Copy, Clone)]
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#[repr(C)]
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pub struct GstWebRTCICECandidateStats {
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pub ipaddr: *mut c_char,
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pub port: c_uint,
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pub stream_id: c_uint,
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pub type_: *const c_char,
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pub proto: *const c_char,
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pub relay_proto: *const c_char,
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pub prio: c_uint,
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pub url: *mut c_char,
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pub _gst_reserved: [gpointer; 20],
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}
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impl ::std::fmt::Debug for GstWebRTCICECandidateStats {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCICECandidateStats @ {self:p}"))
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.field("ipaddr", &self.ipaddr)
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.field("port", &self.port)
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.field("stream_id", &self.stream_id)
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.field("type_", &self.type_)
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.field("proto", &self.proto)
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.field("relay_proto", &self.relay_proto)
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.field("prio", &self.prio)
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.field("url", &self.url)
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.field("_gst_reserved", &self._gst_reserved)
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.finish()
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}
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}
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#[derive(Copy, Clone)]
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#[repr(C)]
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pub struct GstWebRTCICEClass {
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pub parent_class: gst::GstObjectClass,
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pub add_stream:
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Option<unsafe extern "C" fn(*mut GstWebRTCICE, c_uint) -> *mut GstWebRTCICEStream>,
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pub find_transport: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICE,
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*mut GstWebRTCICEStream,
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GstWebRTCICEComponent,
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) -> *mut GstWebRTCICETransport,
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>,
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pub gather_candidates:
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Option<unsafe extern "C" fn(*mut GstWebRTCICE, *mut GstWebRTCICEStream) -> gboolean>,
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pub add_candidate: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICE,
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*mut GstWebRTCICEStream,
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*const c_char,
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*mut gst::GstPromise,
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),
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>,
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pub set_local_credentials: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICE,
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*mut GstWebRTCICEStream,
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*const c_char,
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*const c_char,
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) -> gboolean,
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>,
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pub set_remote_credentials: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICE,
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*mut GstWebRTCICEStream,
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*const c_char,
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*const c_char,
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) -> gboolean,
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>,
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pub add_turn_server: Option<unsafe extern "C" fn(*mut GstWebRTCICE, *const c_char) -> gboolean>,
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pub set_is_controller: Option<unsafe extern "C" fn(*mut GstWebRTCICE, gboolean)>,
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pub get_is_controller: Option<unsafe extern "C" fn(*mut GstWebRTCICE) -> gboolean>,
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pub set_force_relay: Option<unsafe extern "C" fn(*mut GstWebRTCICE, gboolean)>,
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pub set_stun_server: Option<unsafe extern "C" fn(*mut GstWebRTCICE, *const c_char)>,
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pub get_stun_server: Option<unsafe extern "C" fn(*mut GstWebRTCICE) -> *mut c_char>,
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pub set_turn_server: Option<unsafe extern "C" fn(*mut GstWebRTCICE, *const c_char)>,
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pub get_turn_server: Option<unsafe extern "C" fn(*mut GstWebRTCICE) -> *mut c_char>,
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pub set_http_proxy: Option<unsafe extern "C" fn(*mut GstWebRTCICE, *const c_char)>,
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pub get_http_proxy: Option<unsafe extern "C" fn(*mut GstWebRTCICE) -> *mut c_char>,
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pub set_tos: Option<unsafe extern "C" fn(*mut GstWebRTCICE, *mut GstWebRTCICEStream, c_uint)>,
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pub set_on_ice_candidate: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICE,
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GstWebRTCICEOnCandidateFunc,
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gpointer,
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glib::GDestroyNotify,
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),
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>,
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pub get_local_candidates: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICE,
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*mut GstWebRTCICEStream,
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) -> *mut *mut GstWebRTCICECandidateStats,
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>,
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pub get_remote_candidates: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICE,
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*mut GstWebRTCICEStream,
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) -> *mut *mut GstWebRTCICECandidateStats,
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>,
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pub get_selected_pair: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICE,
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*mut GstWebRTCICEStream,
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*mut *mut GstWebRTCICECandidateStats,
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*mut *mut GstWebRTCICECandidateStats,
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) -> gboolean,
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>,
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pub _gst_reserved: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCICEClass {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCICEClass @ {self:p}"))
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.field("parent_class", &self.parent_class)
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.field("add_stream", &self.add_stream)
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.field("find_transport", &self.find_transport)
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.field("gather_candidates", &self.gather_candidates)
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.field("add_candidate", &self.add_candidate)
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.field("set_local_credentials", &self.set_local_credentials)
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.field("set_remote_credentials", &self.set_remote_credentials)
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.field("add_turn_server", &self.add_turn_server)
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.field("set_is_controller", &self.set_is_controller)
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.field("get_is_controller", &self.get_is_controller)
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.field("set_force_relay", &self.set_force_relay)
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.field("set_stun_server", &self.set_stun_server)
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.field("get_stun_server", &self.get_stun_server)
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.field("set_turn_server", &self.set_turn_server)
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.field("get_turn_server", &self.get_turn_server)
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.field("set_http_proxy", &self.set_http_proxy)
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.field("get_http_proxy", &self.get_http_proxy)
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.field("set_tos", &self.set_tos)
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.field("set_on_ice_candidate", &self.set_on_ice_candidate)
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.field("get_local_candidates", &self.get_local_candidates)
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.field("get_remote_candidates", &self.get_remote_candidates)
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.field("get_selected_pair", &self.get_selected_pair)
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.field("_gst_reserved", &self._gst_reserved)
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.finish()
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}
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}
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#[derive(Copy, Clone)]
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#[repr(C)]
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pub struct GstWebRTCICEStreamClass {
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pub parent_class: gst::GstObjectClass,
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pub find_transport: Option<
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unsafe extern "C" fn(
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*mut GstWebRTCICEStream,
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GstWebRTCICEComponent,
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) -> *mut GstWebRTCICETransport,
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>,
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pub gather_candidates: Option<unsafe extern "C" fn(*mut GstWebRTCICEStream) -> gboolean>,
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}
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impl ::std::fmt::Debug for GstWebRTCICEStreamClass {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCICEStreamClass @ {self:p}"))
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.field("parent_class", &self.parent_class)
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.field("find_transport", &self.find_transport)
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.field("gather_candidates", &self.gather_candidates)
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.finish()
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}
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}
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#[derive(Copy, Clone)]
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#[repr(C)]
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pub struct GstWebRTCICETransportClass {
|
|
pub parent_class: gst::GstObjectClass,
|
|
pub gather_candidates: Option<unsafe extern "C" fn(*mut GstWebRTCICETransport) -> gboolean>,
|
|
pub _padding: [gpointer; 4],
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCICETransportClass {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCICETransportClass @ {self:p}"))
|
|
.field("parent_class", &self.parent_class)
|
|
.field("gather_candidates", &self.gather_candidates)
|
|
.field("_padding", &self._padding)
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[repr(C)]
|
|
pub struct _GstWebRTCRTPReceiverClass {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
pub type GstWebRTCRTPReceiverClass = _GstWebRTCRTPReceiverClass;
|
|
|
|
#[repr(C)]
|
|
pub struct _GstWebRTCRTPSenderClass {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
pub type GstWebRTCRTPSenderClass = _GstWebRTCRTPSenderClass;
|
|
|
|
#[repr(C)]
|
|
pub struct _GstWebRTCRTPTransceiverClass {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
pub type GstWebRTCRTPTransceiverClass = _GstWebRTCRTPTransceiverClass;
|
|
|
|
#[repr(C)]
|
|
pub struct _GstWebRTCSCTPTransportClass {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
pub type GstWebRTCSCTPTransportClass = _GstWebRTCSCTPTransportClass;
|
|
|
|
#[derive(Copy, Clone)]
|
|
#[repr(C)]
|
|
pub struct GstWebRTCSessionDescription {
|
|
pub type_: GstWebRTCSDPType,
|
|
pub sdp: *mut gst_sdp::GstSDPMessage,
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCSessionDescription {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCSessionDescription @ {self:p}"))
|
|
.field("type_", &self.type_)
|
|
.field("sdp", &self.sdp)
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
// Classes
|
|
#[repr(C)]
|
|
pub struct GstWebRTCDTLSTransport {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCDTLSTransport {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCDTLSTransport @ {self:p}"))
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[repr(C)]
|
|
pub struct GstWebRTCDataChannel {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCDataChannel {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCDataChannel @ {self:p}"))
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[derive(Copy, Clone)]
|
|
#[repr(C)]
|
|
pub struct GstWebRTCICE {
|
|
pub parent: gst::GstObject,
|
|
pub ice_gathering_state: GstWebRTCICEGatheringState,
|
|
pub ice_connection_state: GstWebRTCICEConnectionState,
|
|
pub min_rtp_port: c_uint,
|
|
pub max_rtp_port: c_uint,
|
|
pub _gst_reserved: [gpointer; 4],
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCICE {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCICE @ {self:p}"))
|
|
.field("parent", &self.parent)
|
|
.field("ice_gathering_state", &self.ice_gathering_state)
|
|
.field("ice_connection_state", &self.ice_connection_state)
|
|
.field("min_rtp_port", &self.min_rtp_port)
|
|
.field("max_rtp_port", &self.max_rtp_port)
|
|
.field("_gst_reserved", &self._gst_reserved)
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[derive(Copy, Clone)]
|
|
#[repr(C)]
|
|
pub struct GstWebRTCICEStream {
|
|
pub parent: gst::GstObject,
|
|
pub stream_id: c_uint,
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCICEStream {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCICEStream @ {self:p}"))
|
|
.field("parent", &self.parent)
|
|
.field("stream_id", &self.stream_id)
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[derive(Copy, Clone)]
|
|
#[repr(C)]
|
|
pub struct GstWebRTCICETransport {
|
|
pub parent: gst::GstObject,
|
|
pub role: GstWebRTCICERole,
|
|
pub component: GstWebRTCICEComponent,
|
|
pub state: GstWebRTCICEConnectionState,
|
|
pub gathering_state: GstWebRTCICEGatheringState,
|
|
pub src: *mut gst::GstElement,
|
|
pub sink: *mut gst::GstElement,
|
|
pub _padding: [gpointer; 4],
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCICETransport {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCICETransport @ {self:p}"))
|
|
.field("parent", &self.parent)
|
|
.field("role", &self.role)
|
|
.field("component", &self.component)
|
|
.field("state", &self.state)
|
|
.field("gathering_state", &self.gathering_state)
|
|
.field("src", &self.src)
|
|
.field("sink", &self.sink)
|
|
.field("_padding", &self._padding)
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[repr(C)]
|
|
pub struct GstWebRTCRTPReceiver {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCRTPReceiver {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCRTPReceiver @ {self:p}"))
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[repr(C)]
|
|
pub struct GstWebRTCRTPSender {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCRTPSender {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCRTPSender @ {self:p}"))
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[repr(C)]
|
|
pub struct GstWebRTCRTPTransceiver {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCRTPTransceiver {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCRTPTransceiver @ {self:p}"))
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[repr(C)]
|
|
pub struct GstWebRTCSCTPTransport {
|
|
_data: [u8; 0],
|
|
_marker: core::marker::PhantomData<(*mut u8, core::marker::PhantomPinned)>,
|
|
}
|
|
|
|
impl ::std::fmt::Debug for GstWebRTCSCTPTransport {
|
|
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
|
|
f.debug_struct(&format!("GstWebRTCSCTPTransport @ {self:p}"))
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
#[link(name = "gstwebrtc-1.0")]
|
|
extern "C" {
|
|
|
|
//=========================================================================
|
|
// GstWebRTCBundlePolicy
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_16")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))]
|
|
pub fn gst_webrtc_bundle_policy_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCDTLSSetup
|
|
//=========================================================================
|
|
pub fn gst_webrtc_dtls_setup_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCDTLSTransportState
|
|
//=========================================================================
|
|
pub fn gst_webrtc_dtls_transport_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCDataChannelState
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_16")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))]
|
|
pub fn gst_webrtc_data_channel_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCError
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_20")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))]
|
|
pub fn gst_webrtc_error_get_type() -> GType;
|
|
#[cfg(feature = "v1_20")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))]
|
|
pub fn gst_webrtc_error_quark() -> glib::GQuark;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCFECType
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_14_1")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_14_1")))]
|
|
pub fn gst_webrtc_fec_type_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICEComponent
|
|
//=========================================================================
|
|
pub fn gst_webrtc_ice_component_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICEConnectionState
|
|
//=========================================================================
|
|
pub fn gst_webrtc_ice_connection_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICEGatheringState
|
|
//=========================================================================
|
|
pub fn gst_webrtc_ice_gathering_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICERole
|
|
//=========================================================================
|
|
pub fn gst_webrtc_ice_role_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICETransportPolicy
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_16")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))]
|
|
pub fn gst_webrtc_ice_transport_policy_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCKind
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_20")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))]
|
|
pub fn gst_webrtc_kind_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCPeerConnectionState
|
|
//=========================================================================
|
|
pub fn gst_webrtc_peer_connection_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCPriorityType
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_16")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))]
|
|
pub fn gst_webrtc_priority_type_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCRTPTransceiverDirection
|
|
//=========================================================================
|
|
pub fn gst_webrtc_rtp_transceiver_direction_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCSCTPTransportState
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_16")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_16")))]
|
|
pub fn gst_webrtc_sctp_transport_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCSDPType
|
|
//=========================================================================
|
|
pub fn gst_webrtc_sdp_type_get_type() -> GType;
|
|
pub fn gst_webrtc_sdp_type_to_string(type_: GstWebRTCSDPType) -> *const c_char;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCSignalingState
|
|
//=========================================================================
|
|
pub fn gst_webrtc_signaling_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCStatsType
|
|
//=========================================================================
|
|
pub fn gst_webrtc_stats_type_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICECandidateStats
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_candidate_stats_get_type() -> GType;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_candidate_stats_copy(
|
|
stats: *mut GstWebRTCICECandidateStats,
|
|
) -> *mut GstWebRTCICECandidateStats;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_candidate_stats_free(stats: *mut GstWebRTCICECandidateStats);
|
|
|
|
//=========================================================================
|
|
// GstWebRTCSessionDescription
|
|
//=========================================================================
|
|
pub fn gst_webrtc_session_description_get_type() -> GType;
|
|
pub fn gst_webrtc_session_description_new(
|
|
type_: GstWebRTCSDPType,
|
|
sdp: *mut gst_sdp::GstSDPMessage,
|
|
) -> *mut GstWebRTCSessionDescription;
|
|
pub fn gst_webrtc_session_description_copy(
|
|
src: *const GstWebRTCSessionDescription,
|
|
) -> *mut GstWebRTCSessionDescription;
|
|
pub fn gst_webrtc_session_description_free(desc: *mut GstWebRTCSessionDescription);
|
|
|
|
//=========================================================================
|
|
// GstWebRTCDTLSTransport
|
|
//=========================================================================
|
|
pub fn gst_webrtc_dtls_transport_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCDataChannel
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_18")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_18")))]
|
|
pub fn gst_webrtc_data_channel_get_type() -> GType;
|
|
#[cfg(feature = "v1_18")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_18")))]
|
|
pub fn gst_webrtc_data_channel_close(channel: *mut GstWebRTCDataChannel);
|
|
#[cfg(feature = "v1_18")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_18")))]
|
|
pub fn gst_webrtc_data_channel_send_data(
|
|
channel: *mut GstWebRTCDataChannel,
|
|
data: *mut glib::GBytes,
|
|
);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_data_channel_send_data_full(
|
|
channel: *mut GstWebRTCDataChannel,
|
|
data: *mut glib::GBytes,
|
|
error: *mut *mut glib::GError,
|
|
) -> gboolean;
|
|
#[cfg(feature = "v1_18")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_18")))]
|
|
pub fn gst_webrtc_data_channel_send_string(
|
|
channel: *mut GstWebRTCDataChannel,
|
|
str: *const c_char,
|
|
);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_data_channel_send_string_full(
|
|
channel: *mut GstWebRTCDataChannel,
|
|
str: *const c_char,
|
|
error: *mut *mut glib::GError,
|
|
) -> gboolean;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICE
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_get_type() -> GType;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_add_candidate(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
candidate: *const c_char,
|
|
promise: *mut gst::GstPromise,
|
|
);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_add_stream(
|
|
ice: *mut GstWebRTCICE,
|
|
session_id: c_uint,
|
|
) -> *mut GstWebRTCICEStream;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_add_turn_server(ice: *mut GstWebRTCICE, uri: *const c_char) -> gboolean;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_find_transport(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
component: GstWebRTCICEComponent,
|
|
) -> *mut GstWebRTCICETransport;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_gather_candidates(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
) -> gboolean;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_get_http_proxy(ice: *mut GstWebRTCICE) -> *mut c_char;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_get_is_controller(ice: *mut GstWebRTCICE) -> gboolean;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_get_local_candidates(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
) -> *mut *mut GstWebRTCICECandidateStats;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_get_remote_candidates(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
) -> *mut *mut GstWebRTCICECandidateStats;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_get_selected_pair(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
local_stats: *mut *mut GstWebRTCICECandidateStats,
|
|
remote_stats: *mut *mut GstWebRTCICECandidateStats,
|
|
) -> gboolean;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_get_stun_server(ice: *mut GstWebRTCICE) -> *mut c_char;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_get_turn_server(ice: *mut GstWebRTCICE) -> *mut c_char;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_force_relay(ice: *mut GstWebRTCICE, force_relay: gboolean);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_http_proxy(ice: *mut GstWebRTCICE, uri: *const c_char);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_is_controller(ice: *mut GstWebRTCICE, controller: gboolean);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_local_credentials(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
ufrag: *const c_char,
|
|
pwd: *const c_char,
|
|
) -> gboolean;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_on_ice_candidate(
|
|
ice: *mut GstWebRTCICE,
|
|
func: GstWebRTCICEOnCandidateFunc,
|
|
user_data: gpointer,
|
|
notify: glib::GDestroyNotify,
|
|
);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_remote_credentials(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
ufrag: *const c_char,
|
|
pwd: *const c_char,
|
|
) -> gboolean;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_stun_server(ice: *mut GstWebRTCICE, uri: *const c_char);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_tos(
|
|
ice: *mut GstWebRTCICE,
|
|
stream: *mut GstWebRTCICEStream,
|
|
tos: c_uint,
|
|
);
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_set_turn_server(ice: *mut GstWebRTCICE, uri: *const c_char);
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICEStream
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_stream_get_type() -> GType;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_stream_find_transport(
|
|
stream: *mut GstWebRTCICEStream,
|
|
component: GstWebRTCICEComponent,
|
|
) -> *mut GstWebRTCICETransport;
|
|
#[cfg(feature = "v1_22")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_22")))]
|
|
pub fn gst_webrtc_ice_stream_gather_candidates(ice: *mut GstWebRTCICEStream) -> gboolean;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCICETransport
|
|
//=========================================================================
|
|
pub fn gst_webrtc_ice_transport_get_type() -> GType;
|
|
pub fn gst_webrtc_ice_transport_connection_state_change(
|
|
ice: *mut GstWebRTCICETransport,
|
|
new_state: GstWebRTCICEConnectionState,
|
|
);
|
|
pub fn gst_webrtc_ice_transport_gathering_state_change(
|
|
ice: *mut GstWebRTCICETransport,
|
|
new_state: GstWebRTCICEGatheringState,
|
|
);
|
|
pub fn gst_webrtc_ice_transport_new_candidate(
|
|
ice: *mut GstWebRTCICETransport,
|
|
stream_id: c_uint,
|
|
component: GstWebRTCICEComponent,
|
|
attr: *const c_char,
|
|
);
|
|
pub fn gst_webrtc_ice_transport_selected_pair_change(ice: *mut GstWebRTCICETransport);
|
|
|
|
//=========================================================================
|
|
// GstWebRTCRTPReceiver
|
|
//=========================================================================
|
|
pub fn gst_webrtc_rtp_receiver_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCRTPSender
|
|
//=========================================================================
|
|
pub fn gst_webrtc_rtp_sender_get_type() -> GType;
|
|
#[cfg(feature = "v1_20")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))]
|
|
pub fn gst_webrtc_rtp_sender_set_priority(
|
|
sender: *mut GstWebRTCRTPSender,
|
|
priority: GstWebRTCPriorityType,
|
|
);
|
|
|
|
//=========================================================================
|
|
// GstWebRTCRTPTransceiver
|
|
//=========================================================================
|
|
pub fn gst_webrtc_rtp_transceiver_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCSCTPTransport
|
|
//=========================================================================
|
|
#[cfg(feature = "v1_20")]
|
|
#[cfg_attr(docsrs, doc(cfg(feature = "v1_20")))]
|
|
pub fn gst_webrtc_sctp_transport_get_type() -> GType;
|
|
|
|
}
|