public Timeline.audio_video ();
public bool commit ();
public bool commit_sync ();
+ [Version (since = "1.20")]
+ public void freeze_commit ();
[CCode (has_construct_function = false)]
public Timeline.from_uri (string uri) throws GLib.Error;
public bool get_auto_transition ();
public bool save_to_uri (string uri, GES.Asset? formatter_asset, bool overwrite) throws GLib.Error;
public void set_auto_transition (bool auto_transition);
public void set_snapping_distance (Gst.ClockTime snapping_distance);
+ [Version (since = "1.20")]
+ public void thaw_commit ();
public bool auto_transition { get; set; }
public uint64 duration { get; }
public uint64 snapping_distance { get; set; }
public bool wait_on_eos { get; set; }
public virtual signal Gst.FlowReturn new_preroll ();
public virtual signal Gst.FlowReturn new_sample ();
+ [Version (since = "1.20")]
+ public signal bool new_serialized_event ();
[HasEmitter]
public virtual signal Gst.Sample pull_preroll ();
[HasEmitter]
public virtual signal Gst.Sample pull_sample ();
+ [Version (since = "1.20")]
+ public virtual signal Gst.MiniObject try_pull_object (uint64 timeout);
[HasEmitter]
[Version (since = "1.10")]
public virtual signal Gst.Sample try_pull_preroll (uint64 timeout);
//buffer_*_rtp_source_meta parent="Gst.RTP.Buffer" name="buffer_(.+)"
RTPBuffer
.map unowned=false
-RTPHeaderExtension
- .read.data array array_length_idx=2
- .write.data array array_length_idx=4
RTPSourceMeta
.append_csrc.csrc array array_length_idx=1