public uint64 discont_wait { get; set; }
[NoAccessorMethod]
public uint64 output_buffer_duration { get; set; }
+ [NoAccessorMethod]
+ public Gst.Fraction output_buffer_duration_fraction { owned get; set; }
}
[CCode (cheader_filename = "gst/audio/audio.h", type_id = "gst_audio_aggregator_convert_pad_get_type ()")]
[GIR (name = "AudioAggregatorConvertPad")]
[Version (since = "1.16")]
public Gst.ClockTime get_processing_deadline ();
public Gst.ClockTime get_render_delay ();
+ [Version (since = "1.18")]
public Gst.Structure get_stats ();
public bool get_sync ();
public uint64 get_throttle_time ();
[NoAccessorMethod]
public bool qos { get; set; }
public uint64 render_delay { get; set; }
+ [Version (since = "1.18")]
+ public Gst.Structure stats { owned get; }
public bool sync { get; set; }
public uint64 throttle_time { get; set; }
public int64 ts_offset { get; set; }
public void free ();
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_BUNDLE_POLICY_", type_id = "gst_webrtc_bundle_policy_get_type ()")]
+ [Version (since = "1.16")]
public enum WebRTCBundlePolicy {
NONE,
BALANCED,
CONNECTED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DATA_CHANNEL_STATE_", type_id = "gst_webrtc_data_channel_state_get_type ()")]
+ [Version (since = "1.16")]
public enum WebRTCDataChannelState {
NEW,
CONNECTING,
CLOSED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_FEC_TYPE_", type_id = "gst_webrtc_fec_type_get_type ()")]
+ [Version (since = "1.14.1")]
public enum WebRTCFECType {
NONE,
ULP_RED
CONTROLLING
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_TRANSPORT_POLICY_", type_id = "gst_webrtc_ice_transport_policy_get_type ()")]
+ [Version (since = "1.16")]
public enum WebRTCICETransportPolicy {
ALL,
RELAY
CLOSED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_PRIORITY_TYPE_", type_id = "gst_webrtc_priority_type_get_type ()")]
+ [Version (since = "1.16")]
public enum WebRTCPriorityType {
VERY_LOW,
LOW,
SENDRECV
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SCTP_TRANSPORT_STATE_", type_id = "gst_webrtc_sctp_transport_state_get_type ()")]
+ [Version (since = "1.16")]
public enum WebRTCSCTPTransportState {
NEW,
CONNECTING,