mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer-rs.git
synced 2024-09-29 07:12:25 +00:00
555 lines
23 KiB
Rust
555 lines
23 KiB
Rust
// This file was generated by gir (https://github.com/gtk-rs/gir @ d1e88f9)
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// from gir-files (https://github.com/gtk-rs/gir-files @ 67b410c)
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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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)]
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extern crate glib_sys as glib;
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extern crate gobject_sys as gobject;
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extern crate gstreamer_sdp_sys as gst_sdp;
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extern crate gstreamer_sys as gst;
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extern crate libc;
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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, time_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_NEW: GstWebRTCDataChannelState = 0;
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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 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 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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// Records
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCDTLSTransportClass {
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pub parent_class: gst::GstObjectClass,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCDTLSTransportClass {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!(
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"GstWebRTCDTLSTransportClass @ {:?}",
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self as *const _
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))
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.field("parent_class", &self.parent_class)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCICETransportClass {
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pub parent_class: gst::GstObjectClass,
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pub gather_candidates: Option<unsafe extern "C" fn(*mut GstWebRTCICETransport) -> gboolean>,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCICETransportClass {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!(
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"GstWebRTCICETransportClass @ {:?}",
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self as *const _
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))
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.field("parent_class", &self.parent_class)
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.field("gather_candidates", &self.gather_candidates)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCRTPReceiverClass {
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pub parent_class: gst::GstObjectClass,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCRTPReceiverClass {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!(
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"GstWebRTCRTPReceiverClass @ {:?}",
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self as *const _
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))
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.field("parent_class", &self.parent_class)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCRTPSenderClass {
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pub parent_class: gst::GstObjectClass,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCRTPSenderClass {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPSenderClass @ {:?}", self as *const _))
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.field("parent_class", &self.parent_class)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCRTPTransceiverClass {
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pub parent_class: gst::GstObjectClass,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCRTPTransceiverClass {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!(
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"GstWebRTCRTPTransceiverClass @ {:?}",
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self as *const _
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))
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.field("parent_class", &self.parent_class)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCSessionDescription {
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pub type_: GstWebRTCSDPType,
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pub sdp: *mut gst_sdp::GstSDPMessage,
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}
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impl ::std::fmt::Debug for GstWebRTCSessionDescription {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!(
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"GstWebRTCSessionDescription @ {:?}",
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self as *const _
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))
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.field("type_", &self.type_)
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.field("sdp", &self.sdp)
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.finish()
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}
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}
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// Classes
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCDTLSTransport {
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pub parent: gst::GstObject,
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pub transport: *mut GstWebRTCICETransport,
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pub state: GstWebRTCDTLSTransportState,
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pub is_rtcp: gboolean,
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pub client: gboolean,
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pub session_id: c_uint,
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pub dtlssrtpenc: *mut gst::GstElement,
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pub dtlssrtpdec: *mut gst::GstElement,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCDTLSTransport {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCDTLSTransport @ {:?}", self as *const _))
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.field("parent", &self.parent)
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.field("transport", &self.transport)
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.field("state", &self.state)
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.field("is_rtcp", &self.is_rtcp)
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.field("client", &self.client)
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.field("session_id", &self.session_id)
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.field("dtlssrtpenc", &self.dtlssrtpenc)
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.field("dtlssrtpdec", &self.dtlssrtpdec)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCICETransport {
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pub parent: gst::GstObject,
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pub role: GstWebRTCICERole,
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pub component: GstWebRTCICEComponent,
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pub state: GstWebRTCICEConnectionState,
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pub gathering_state: GstWebRTCICEGatheringState,
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pub src: *mut gst::GstElement,
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pub sink: *mut gst::GstElement,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCICETransport {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCICETransport @ {:?}", self as *const _))
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.field("parent", &self.parent)
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.field("role", &self.role)
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.field("component", &self.component)
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.field("state", &self.state)
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.field("gathering_state", &self.gathering_state)
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.field("src", &self.src)
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.field("sink", &self.sink)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCRTPReceiver {
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pub parent: gst::GstObject,
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pub transport: *mut GstWebRTCDTLSTransport,
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pub rtcp_transport: *mut GstWebRTCDTLSTransport,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCRTPReceiver {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPReceiver @ {:?}", self as *const _))
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.field("parent", &self.parent)
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.field("transport", &self.transport)
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.field("rtcp_transport", &self.rtcp_transport)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCRTPSender {
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pub parent: gst::GstObject,
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pub transport: *mut GstWebRTCDTLSTransport,
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pub rtcp_transport: *mut GstWebRTCDTLSTransport,
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pub send_encodings: *mut glib::GArray,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCRTPSender {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPSender @ {:?}", self as *const _))
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.field("parent", &self.parent)
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.field("transport", &self.transport)
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.field("rtcp_transport", &self.rtcp_transport)
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.field("send_encodings", &self.send_encodings)
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.field("_padding", &self._padding)
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.finish()
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}
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}
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#[repr(C)]
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#[derive(Copy, Clone)]
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pub struct GstWebRTCRTPTransceiver {
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pub parent: gst::GstObject,
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pub mline: c_uint,
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pub mid: *mut c_char,
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pub stopped: gboolean,
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pub sender: *mut GstWebRTCRTPSender,
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pub receiver: *mut GstWebRTCRTPReceiver,
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pub direction: GstWebRTCRTPTransceiverDirection,
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pub current_direction: GstWebRTCRTPTransceiverDirection,
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pub codec_preferences: *mut gst::GstCaps,
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pub _padding: [gpointer; 4],
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}
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impl ::std::fmt::Debug for GstWebRTCRTPTransceiver {
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fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPTransceiver @ {:?}", self as *const _))
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.field("parent", &self.parent)
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.field("mline", &self.mline)
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.field("mid", &self.mid)
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.field("stopped", &self.stopped)
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.field("sender", &self.sender)
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.field("receiver", &self.receiver)
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.field("direction", &self.direction)
|
|
.field("current_direction", &self.current_direction)
|
|
.field("codec_preferences", &self.codec_preferences)
|
|
.field("_padding", &self._padding)
|
|
.finish()
|
|
}
|
|
}
|
|
|
|
extern "C" {
|
|
|
|
//=========================================================================
|
|
// GstWebRTCBundlePolicy
|
|
//=========================================================================
|
|
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
|
|
//=========================================================================
|
|
pub fn gst_webrtc_data_channel_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCFECType
|
|
//=========================================================================
|
|
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
|
|
//=========================================================================
|
|
pub fn gst_webrtc_ice_transport_policy_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCPeerConnectionState
|
|
//=========================================================================
|
|
pub fn gst_webrtc_peer_connection_state_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCPriorityType
|
|
//=========================================================================
|
|
pub fn gst_webrtc_priority_type_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCRTPTransceiverDirection
|
|
//=========================================================================
|
|
pub fn gst_webrtc_rtp_transceiver_direction_get_type() -> GType;
|
|
|
|
//=========================================================================
|
|
// GstWebRTCSCTPTransportState
|
|
//=========================================================================
|
|
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;
|
|
|
|
//=========================================================================
|
|
// 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;
|
|
pub fn gst_webrtc_dtls_transport_new(
|
|
session_id: c_uint,
|
|
rtcp: gboolean,
|
|
) -> *mut GstWebRTCDTLSTransport;
|
|
pub fn gst_webrtc_dtls_transport_set_transport(
|
|
transport: *mut GstWebRTCDTLSTransport,
|
|
ice: *mut GstWebRTCICETransport,
|
|
);
|
|
|
|
//=========================================================================
|
|
// 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: *mut c_char,
|
|
);
|
|
pub fn gst_webrtc_ice_transport_selected_pair_change(ice: *mut GstWebRTCICETransport);
|
|
|
|
//=========================================================================
|
|
// GstWebRTCRTPReceiver
|
|
//=========================================================================
|
|
pub fn gst_webrtc_rtp_receiver_get_type() -> GType;
|
|
pub fn gst_webrtc_rtp_receiver_new() -> *mut GstWebRTCRTPReceiver;
|
|
pub fn gst_webrtc_rtp_receiver_set_rtcp_transport(
|
|
receiver: *mut GstWebRTCRTPReceiver,
|
|
transport: *mut GstWebRTCDTLSTransport,
|
|
);
|
|
pub fn gst_webrtc_rtp_receiver_set_transport(
|
|
receiver: *mut GstWebRTCRTPReceiver,
|
|
transport: *mut GstWebRTCDTLSTransport,
|
|
);
|
|
|
|
//=========================================================================
|
|
// GstWebRTCRTPSender
|
|
//=========================================================================
|
|
pub fn gst_webrtc_rtp_sender_get_type() -> GType;
|
|
pub fn gst_webrtc_rtp_sender_new() -> *mut GstWebRTCRTPSender;
|
|
pub fn gst_webrtc_rtp_sender_set_rtcp_transport(
|
|
sender: *mut GstWebRTCRTPSender,
|
|
transport: *mut GstWebRTCDTLSTransport,
|
|
);
|
|
pub fn gst_webrtc_rtp_sender_set_transport(
|
|
sender: *mut GstWebRTCRTPSender,
|
|
transport: *mut GstWebRTCDTLSTransport,
|
|
);
|
|
|
|
//=========================================================================
|
|
// GstWebRTCRTPTransceiver
|
|
//=========================================================================
|
|
pub fn gst_webrtc_rtp_transceiver_get_type() -> GType;
|
|
|
|
}
|