rtpsirendepay
Extracts Siren audio from RTP packets
Hierarchy
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstElement ╰──GstRTPBaseDepayload ╰──rtpsirendepay
Factory details
Authors: – Philippe Kalaf
Classification: – Codec/Depayloader/Network/RTP
Rank – secondary
Plugin – gstrtp
Package – GStreamer Good Plug-ins
Pad Templates
sink
application/x-rtp:
media: audio
clock-rate: 16000
encoding-name: SIREN
Properties
max-reorder
“max-reorder” gint
Max seqnum reorder before assuming sender has restarted
Flags : Read / Write
Default value : 100
source-info
“source-info” gboolean
Add RTP source information as buffer meta
Flags : Read / Write
Default value : false
stats
“stats” GstStructure *
Various statistics
Flags : Read
Default value :
application/x-rtp-depayload-stats, clock_rate=(uint)0, npt-start=(guint64)0, npt-stop=(guint64)18446744073709551615, play-speed=(double)1, play-scale=(double)1, running-time-dts=(guint64)18446744073709551615, running-time-pts=(guint64)18446744073709551615, seqnum=(uint)0, timestamp=(uint)0;
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