rtpopusdepay2

Extracts an Opus audio stream from RTP packets as per RFC 7587 or libwebrtc's multiopus extension.

Example pipeline

 gst-launch-1.0 udpsrc caps='application/x-rtp, media=audio, clock-rate=48000, encoding-name=OPUS, encoding-params=(string)2, sprop-stereo=(string)1, payload=96' ! rtpjitterbuffer latency=50 ! rtpopusdepay2 ! opusdec ! audioconvert ! audioresample ! autoaudiosink

This will depayload an incoming RTP Opus audio stream. You can use the opusenc and rtpopuspay2 elements to create such an RTP stream.

Hierarchy

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstRtpBaseDepay2
                    ╰──rtpopusdepay2

Factory details

Authors: – Tim-Philipp Müller

Classification: – Codec/Depayloader/Network/RTP

Rank – marginal

Plugin – rsrtp

Package – gst-plugin-rtp

Pad Templates

`sink`

application/x-rtp:
          media: audio
  encoding-name: { (string)OPUS, (string)MULTIOPUS }
     clock-rate: 48000

Presence – always

Direction – sink

Object type – GstPad

`src`

audio/x-opus:
channel-mapping-family: [ 0, 1 ]

Presence – always

Direction – src

Object type – GstPad

The results of the search are