gstreamer-rs/gstreamer-webrtc/sys/src/lib.rs

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// Generated by gir (https://github.com/gtk-rs/gir @ e53b4d7)
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// from gir-files (https://github.com/gtk-rs/gir-files @ 7d95377)
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// from gst-gir-files (https://gitlab.freedesktop.org/gstreamer/gir-files-rs.git @ 85bd06b)
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// DO NOT EDIT
#![allow(non_camel_case_types, non_upper_case_globals, non_snake_case)]
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#![allow(
clippy::approx_constant,
clippy::type_complexity,
clippy::unreadable_literal,
clippy::upper_case_acronyms
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)]
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#![cfg_attr(feature = "dox", feature(doc_cfg))]
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use glib_sys as glib;
use gobject_sys as gobject;
use gstreamer_sdp_sys as gst_sdp;
use gstreamer_sys as gst;
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#[allow(unused_imports)]
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use libc::{
c_char, c_double, c_float, c_int, c_long, c_short, c_uchar, c_uint, c_ulong, c_ushort, c_void,
intptr_t, size_t, ssize_t, time_t, uintptr_t, FILE,
};
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#[allow(unused_imports)]
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use glib::{gboolean, gconstpointer, gpointer, GType};
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// Enums
pub type GstWebRTCBundlePolicy = c_int;
pub const GST_WEBRTC_BUNDLE_POLICY_NONE: GstWebRTCBundlePolicy = 0;
pub const GST_WEBRTC_BUNDLE_POLICY_BALANCED: GstWebRTCBundlePolicy = 1;
pub const GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: GstWebRTCBundlePolicy = 2;
pub const GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: GstWebRTCBundlePolicy = 3;
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pub type GstWebRTCDTLSSetup = c_int;
pub const GST_WEBRTC_DTLS_SETUP_NONE: GstWebRTCDTLSSetup = 0;
pub const GST_WEBRTC_DTLS_SETUP_ACTPASS: GstWebRTCDTLSSetup = 1;
pub const GST_WEBRTC_DTLS_SETUP_ACTIVE: GstWebRTCDTLSSetup = 2;
pub const GST_WEBRTC_DTLS_SETUP_PASSIVE: GstWebRTCDTLSSetup = 3;
pub type GstWebRTCDTLSTransportState = c_int;
pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: GstWebRTCDTLSTransportState = 0;
pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: GstWebRTCDTLSTransportState = 1;
pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: GstWebRTCDTLSTransportState = 2;
pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: GstWebRTCDTLSTransportState = 3;
pub const GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: GstWebRTCDTLSTransportState = 4;
pub type GstWebRTCDataChannelState = c_int;
pub const GST_WEBRTC_DATA_CHANNEL_STATE_NEW: GstWebRTCDataChannelState = 0;
pub const GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: GstWebRTCDataChannelState = 1;
pub const GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: GstWebRTCDataChannelState = 2;
pub const GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: GstWebRTCDataChannelState = 3;
pub const GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: GstWebRTCDataChannelState = 4;
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pub type GstWebRTCFECType = c_int;
pub const GST_WEBRTC_FEC_TYPE_NONE: GstWebRTCFECType = 0;
pub const GST_WEBRTC_FEC_TYPE_ULP_RED: GstWebRTCFECType = 1;
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pub type GstWebRTCICEComponent = c_int;
pub const GST_WEBRTC_ICE_COMPONENT_RTP: GstWebRTCICEComponent = 0;
pub const GST_WEBRTC_ICE_COMPONENT_RTCP: GstWebRTCICEComponent = 1;
pub type GstWebRTCICEConnectionState = c_int;
pub const GST_WEBRTC_ICE_CONNECTION_STATE_NEW: GstWebRTCICEConnectionState = 0;
pub const GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: GstWebRTCICEConnectionState = 1;
pub const GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: GstWebRTCICEConnectionState = 2;
pub const GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: GstWebRTCICEConnectionState = 3;
pub const GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: GstWebRTCICEConnectionState = 4;
pub const GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: GstWebRTCICEConnectionState = 5;
pub const GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: GstWebRTCICEConnectionState = 6;
pub type GstWebRTCICEGatheringState = c_int;
pub const GST_WEBRTC_ICE_GATHERING_STATE_NEW: GstWebRTCICEGatheringState = 0;
pub const GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: GstWebRTCICEGatheringState = 1;
pub const GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: GstWebRTCICEGatheringState = 2;
pub type GstWebRTCICERole = c_int;
pub const GST_WEBRTC_ICE_ROLE_CONTROLLED: GstWebRTCICERole = 0;
pub const GST_WEBRTC_ICE_ROLE_CONTROLLING: GstWebRTCICERole = 1;
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pub type GstWebRTCICETransportPolicy = c_int;
pub const GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: GstWebRTCICETransportPolicy = 0;
pub const GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: GstWebRTCICETransportPolicy = 1;
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pub type GstWebRTCKind = c_int;
pub const GST_WEBRTC_KIND_UNKNOWN: GstWebRTCKind = 0;
pub const GST_WEBRTC_KIND_AUDIO: GstWebRTCKind = 1;
pub const GST_WEBRTC_KIND_VIDEO: GstWebRTCKind = 2;
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pub type GstWebRTCPeerConnectionState = c_int;
pub const GST_WEBRTC_PEER_CONNECTION_STATE_NEW: GstWebRTCPeerConnectionState = 0;
pub const GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: GstWebRTCPeerConnectionState = 1;
pub const GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: GstWebRTCPeerConnectionState = 2;
pub const GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: GstWebRTCPeerConnectionState = 3;
pub const GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: GstWebRTCPeerConnectionState = 4;
pub const GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: GstWebRTCPeerConnectionState = 5;
pub type GstWebRTCPriorityType = c_int;
pub const GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: GstWebRTCPriorityType = 1;
pub const GST_WEBRTC_PRIORITY_TYPE_LOW: GstWebRTCPriorityType = 2;
pub const GST_WEBRTC_PRIORITY_TYPE_MEDIUM: GstWebRTCPriorityType = 3;
pub const GST_WEBRTC_PRIORITY_TYPE_HIGH: GstWebRTCPriorityType = 4;
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pub type GstWebRTCRTPTransceiverDirection = c_int;
pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: GstWebRTCRTPTransceiverDirection = 0;
pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: GstWebRTCRTPTransceiverDirection = 1;
pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: GstWebRTCRTPTransceiverDirection = 2;
pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: GstWebRTCRTPTransceiverDirection = 3;
pub const GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: GstWebRTCRTPTransceiverDirection = 4;
pub type GstWebRTCSCTPTransportState = c_int;
pub const GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: GstWebRTCSCTPTransportState = 0;
pub const GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: GstWebRTCSCTPTransportState = 1;
pub const GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: GstWebRTCSCTPTransportState = 2;
pub const GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: GstWebRTCSCTPTransportState = 3;
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pub type GstWebRTCSDPType = c_int;
pub const GST_WEBRTC_SDP_TYPE_OFFER: GstWebRTCSDPType = 1;
pub const GST_WEBRTC_SDP_TYPE_PRANSWER: GstWebRTCSDPType = 2;
pub const GST_WEBRTC_SDP_TYPE_ANSWER: GstWebRTCSDPType = 3;
pub const GST_WEBRTC_SDP_TYPE_ROLLBACK: GstWebRTCSDPType = 4;
pub type GstWebRTCSignalingState = c_int;
pub const GST_WEBRTC_SIGNALING_STATE_STABLE: GstWebRTCSignalingState = 0;
pub const GST_WEBRTC_SIGNALING_STATE_CLOSED: GstWebRTCSignalingState = 1;
pub const GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: GstWebRTCSignalingState = 2;
pub const GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: GstWebRTCSignalingState = 3;
pub const GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: GstWebRTCSignalingState = 4;
pub const GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: GstWebRTCSignalingState = 5;
pub type GstWebRTCStatsType = c_int;
pub const GST_WEBRTC_STATS_CODEC: GstWebRTCStatsType = 1;
pub const GST_WEBRTC_STATS_INBOUND_RTP: GstWebRTCStatsType = 2;
pub const GST_WEBRTC_STATS_OUTBOUND_RTP: GstWebRTCStatsType = 3;
pub const GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: GstWebRTCStatsType = 4;
pub const GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: GstWebRTCStatsType = 5;
pub const GST_WEBRTC_STATS_CSRC: GstWebRTCStatsType = 6;
pub const GST_WEBRTC_STATS_PEER_CONNECTION: GstWebRTCStatsType = 7;
pub const GST_WEBRTC_STATS_DATA_CHANNEL: GstWebRTCStatsType = 8;
pub const GST_WEBRTC_STATS_STREAM: GstWebRTCStatsType = 9;
pub const GST_WEBRTC_STATS_TRANSPORT: GstWebRTCStatsType = 10;
pub const GST_WEBRTC_STATS_CANDIDATE_PAIR: GstWebRTCStatsType = 11;
pub const GST_WEBRTC_STATS_LOCAL_CANDIDATE: GstWebRTCStatsType = 12;
pub const GST_WEBRTC_STATS_REMOTE_CANDIDATE: GstWebRTCStatsType = 13;
pub const GST_WEBRTC_STATS_CERTIFICATE: GstWebRTCStatsType = 14;
// Records
#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCDTLSTransportClass {
pub parent_class: gst::GstObjectClass,
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pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCDTLSTransportClass {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCDTLSTransportClass @ {:p}", self))
.field("parent_class", &self.parent_class)
.field("_padding", &self._padding)
.finish()
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}
}
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#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCDataChannelClass {
pub parent_class: gobject::GObjectClass,
pub send_data: Option<unsafe extern "C" fn(*mut GstWebRTCDataChannel, *mut glib::GBytes)>,
pub send_string: Option<unsafe extern "C" fn(*mut GstWebRTCDataChannel, *const c_char)>,
pub close: Option<unsafe extern "C" fn(*mut GstWebRTCDataChannel)>,
pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCDataChannelClass {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCDataChannelClass @ {:p}", self))
.field("parent_class", &self.parent_class)
.field("send_data", &self.send_data)
.field("send_string", &self.send_string)
.field("close", &self.close)
.field("_padding", &self._padding)
.finish()
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}
}
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#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCICETransportClass {
pub parent_class: gst::GstObjectClass,
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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 {
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f.debug_struct(&format!("GstWebRTCICETransportClass @ {:p}", self))
.field("parent_class", &self.parent_class)
.field("gather_candidates", &self.gather_candidates)
.field("_padding", &self._padding)
.finish()
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}
}
#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCRTPReceiverClass {
pub parent_class: gst::GstObjectClass,
pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCRTPReceiverClass {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPReceiverClass @ {:p}", self))
.field("parent_class", &self.parent_class)
.field("_padding", &self._padding)
.finish()
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}
}
#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCRTPSenderClass {
pub parent_class: gst::GstObjectClass,
pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCRTPSenderClass {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPSenderClass @ {:p}", self))
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.field("parent_class", &self.parent_class)
.field("_padding", &self._padding)
.finish()
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}
}
#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCRTPTransceiverClass {
pub parent_class: gst::GstObjectClass,
pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCRTPTransceiverClass {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPTransceiverClass @ {:p}", self))
.field("parent_class", &self.parent_class)
.field("_padding", &self._padding)
.finish()
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}
}
#[repr(C)]
#[derive(Copy, Clone)]
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 {
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f.debug_struct(&format!("GstWebRTCSessionDescription @ {:p}", self))
.field("type_", &self.type_)
.field("sdp", &self.sdp)
.finish()
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}
}
// Classes
#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCDTLSTransport {
pub parent: gst::GstObject,
pub transport: *mut GstWebRTCICETransport,
pub state: GstWebRTCDTLSTransportState,
pub client: gboolean,
pub session_id: c_uint,
pub dtlssrtpenc: *mut gst::GstElement,
pub dtlssrtpdec: *mut gst::GstElement,
pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCDTLSTransport {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCDTLSTransport @ {:p}", self))
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.field("parent", &self.parent)
.field("transport", &self.transport)
.field("state", &self.state)
.field("client", &self.client)
.field("session_id", &self.session_id)
.field("dtlssrtpenc", &self.dtlssrtpenc)
.field("dtlssrtpdec", &self.dtlssrtpdec)
.field("_padding", &self._padding)
.finish()
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}
}
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#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCDataChannel {
pub parent: gobject::GObject,
pub lock: glib::GMutex,
pub label: *mut c_char,
pub ordered: gboolean,
pub max_packet_lifetime: c_uint,
pub max_retransmits: c_uint,
pub protocol: *mut c_char,
pub negotiated: gboolean,
pub id: c_int,
pub priority: GstWebRTCPriorityType,
pub ready_state: GstWebRTCDataChannelState,
pub buffered_amount: u64,
pub buffered_amount_low_threshold: u64,
pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCDataChannel {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCDataChannel @ {:p}", self))
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.field("parent", &self.parent)
.field("lock", &self.lock)
.field("label", &self.label)
.field("ordered", &self.ordered)
.field("max_packet_lifetime", &self.max_packet_lifetime)
.field("max_retransmits", &self.max_retransmits)
.field("protocol", &self.protocol)
.field("negotiated", &self.negotiated)
.field("id", &self.id)
.field("priority", &self.priority)
.field("ready_state", &self.ready_state)
.field("buffered_amount", &self.buffered_amount)
.field(
"buffered_amount_low_threshold",
&self.buffered_amount_low_threshold,
)
.field("_padding", &self._padding)
.finish()
}
}
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#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCICETransport {
pub parent: gst::GstObject,
pub role: GstWebRTCICERole,
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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 {
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f.debug_struct(&format!("GstWebRTCICETransport @ {:p}", self))
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.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()
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}
}
#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCRTPReceiver {
pub parent: gst::GstObject,
pub transport: *mut GstWebRTCDTLSTransport,
pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCRTPReceiver {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPReceiver @ {:p}", self))
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.field("parent", &self.parent)
.field("transport", &self.transport)
.field("_padding", &self._padding)
.finish()
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}
}
#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCRTPSender {
pub parent: gst::GstObject,
pub transport: *mut GstWebRTCDTLSTransport,
pub send_encodings: *mut glib::GArray,
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pub priority: GstWebRTCPriorityType,
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pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCRTPSender {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPSender @ {:p}", self))
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.field("parent", &self.parent)
.field("transport", &self.transport)
.field("send_encodings", &self.send_encodings)
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.field("priority", &self.priority)
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.field("_padding", &self._padding)
.finish()
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}
}
#[repr(C)]
#[derive(Copy, Clone)]
pub struct GstWebRTCRTPTransceiver {
pub parent: gst::GstObject,
pub mline: c_uint,
pub mid: *mut c_char,
pub stopped: gboolean,
pub sender: *mut GstWebRTCRTPSender,
pub receiver: *mut GstWebRTCRTPReceiver,
pub direction: GstWebRTCRTPTransceiverDirection,
pub current_direction: GstWebRTCRTPTransceiverDirection,
pub codec_preferences: *mut gst::GstCaps,
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pub kind: GstWebRTCKind,
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pub _padding: [gpointer; 4],
}
impl ::std::fmt::Debug for GstWebRTCRTPTransceiver {
fn fmt(&self, f: &mut ::std::fmt::Formatter) -> ::std::fmt::Result {
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f.debug_struct(&format!("GstWebRTCRTPTransceiver @ {:p}", self))
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.field("parent", &self.parent)
.field("mline", &self.mline)
.field("mid", &self.mid)
.field("stopped", &self.stopped)
.field("sender", &self.sender)
.field("receiver", &self.receiver)
.field("direction", &self.direction)
.field("current_direction", &self.current_direction)
.field("codec_preferences", &self.codec_preferences)
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.field("kind", &self.kind)
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.field("_padding", &self._padding)
.finish()
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}
}
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#[link(name = "gstwebrtc-1.0")]
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extern "C" {
//=========================================================================
// GstWebRTCBundlePolicy
//=========================================================================
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
pub fn gst_webrtc_bundle_policy_get_type() -> GType;
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//=========================================================================
// GstWebRTCDTLSSetup
//=========================================================================
pub fn gst_webrtc_dtls_setup_get_type() -> GType;
//=========================================================================
// GstWebRTCDTLSTransportState
//=========================================================================
pub fn gst_webrtc_dtls_transport_state_get_type() -> GType;
//=========================================================================
// GstWebRTCDataChannelState
//=========================================================================
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
pub fn gst_webrtc_data_channel_state_get_type() -> GType;
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//=========================================================================
// GstWebRTCFECType
//=========================================================================
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#[cfg(any(feature = "v1_14_1", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_14_1")))]
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pub fn gst_webrtc_fec_type_get_type() -> GType;
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//=========================================================================
// 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
//=========================================================================
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
pub fn gst_webrtc_ice_transport_policy_get_type() -> GType;
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//=========================================================================
// GstWebRTCKind
//=========================================================================
#[cfg(any(feature = "v1_20", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_20")))]
pub fn gst_webrtc_kind_get_type() -> GType;
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//=========================================================================
// GstWebRTCPeerConnectionState
//=========================================================================
pub fn gst_webrtc_peer_connection_state_get_type() -> GType;
//=========================================================================
// GstWebRTCPriorityType
//=========================================================================
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
pub fn gst_webrtc_priority_type_get_type() -> GType;
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//=========================================================================
// GstWebRTCRTPTransceiverDirection
//=========================================================================
pub fn gst_webrtc_rtp_transceiver_direction_get_type() -> GType;
//=========================================================================
// GstWebRTCSCTPTransportState
//=========================================================================
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
pub fn gst_webrtc_sctp_transport_state_get_type() -> GType;
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//=========================================================================
// 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;
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//=========================================================================
// GstWebRTCSessionDescription
//=========================================================================
pub fn gst_webrtc_session_description_get_type() -> GType;
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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;
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pub fn gst_webrtc_session_description_free(desc: *mut GstWebRTCSessionDescription);
//=========================================================================
// GstWebRTCDTLSTransport
//=========================================================================
pub fn gst_webrtc_dtls_transport_get_type() -> GType;
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pub fn gst_webrtc_dtls_transport_new(session_id: c_uint) -> *mut GstWebRTCDTLSTransport;
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pub fn gst_webrtc_dtls_transport_set_transport(
transport: *mut GstWebRTCDTLSTransport,
ice: *mut GstWebRTCICETransport,
);
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//=========================================================================
// GstWebRTCDataChannel
//=========================================================================
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_get_type() -> GType;
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_close(channel: *mut GstWebRTCDataChannel);
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_on_buffered_amount_low(channel: *mut GstWebRTCDataChannel);
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_on_close(channel: *mut GstWebRTCDataChannel);
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_on_error(
channel: *mut GstWebRTCDataChannel,
error: *mut glib::GError,
);
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_on_message_data(
channel: *mut GstWebRTCDataChannel,
data: *mut glib::GBytes,
);
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_on_message_string(
channel: *mut GstWebRTCDataChannel,
str: *const c_char,
);
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_on_open(channel: *mut GstWebRTCDataChannel);
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_send_data(
channel: *mut GstWebRTCDataChannel,
data: *mut glib::GBytes,
);
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#[cfg(any(feature = "v1_18", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_18")))]
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pub fn gst_webrtc_data_channel_send_string(
channel: *mut GstWebRTCDataChannel,
str: *const c_char,
);
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//=========================================================================
// GstWebRTCICETransport
//=========================================================================
pub fn gst_webrtc_ice_transport_get_type() -> GType;
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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,
);
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pub fn gst_webrtc_ice_transport_selected_pair_change(ice: *mut GstWebRTCICETransport);
//=========================================================================
// GstWebRTCRTPReceiver
//=========================================================================
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
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pub fn gst_webrtc_rtp_receiver_get_type() -> GType;
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
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pub fn gst_webrtc_rtp_receiver_new() -> *mut GstWebRTCRTPReceiver;
//=========================================================================
// GstWebRTCRTPSender
//=========================================================================
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
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pub fn gst_webrtc_rtp_sender_get_type() -> GType;
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
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pub fn gst_webrtc_rtp_sender_new() -> *mut GstWebRTCRTPSender;
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#[cfg(any(feature = "v1_20", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_20")))]
pub fn gst_webrtc_rtp_sender_set_priority(
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sender: *mut GstWebRTCRTPSender,
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priority: GstWebRTCPriorityType,
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);
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//=========================================================================
// GstWebRTCRTPTransceiver
//=========================================================================
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#[cfg(any(feature = "v1_16", feature = "dox"))]
#[cfg_attr(feature = "dox", doc(cfg(feature = "v1_16")))]
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pub fn gst_webrtc_rtp_transceiver_get_type() -> GType;
}