rtpamrpay2

Payloads an AMR audio stream into RTP packets as per RFC 3267.

Example pipeline

 gst-launch-1.0 audiotestsrc wave=ticks ! amrnbenc ! rtpamrpay2 ! udpsink host=127.0.0.1 port=5004

This will encode an audio test signal as AMR NB audio and payload it as RTP and send it out over UDP to localhost port 5004.

Hierarchy

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstRtpBasePay2
                    ╰──rtpamrpay2

Factory details

Authors: – Sebastian Dröge

Classification: – Codec/Payloader/Network/RTP

Rank – marginal

Plugin – rsrtp

Package – gst-plugin-rtp

Pad Templates

sink

audio/AMR:
       channels: 1
           rate: 8000
audio/AMR-WB:
       channels: 1
           rate: 16000

Presence – always

Direction – sink

Object type – GstPad

src

application/x-rtp:
          media: audio
  encoding-name: AMR
     clock-rate: 8000
encoding-params: 1
    octet-align: { (string)0, (string)1 }
            crc: 0
 robust-sorting: 0
   interleaving: 0
application/x-rtp:
          media: audio
  encoding-name: AMR-WB
     clock-rate: 16000
encoding-params: 1
    octet-align: { (string)0, (string)1 }
            crc: 0
 robust-sorting: 0
   interleaving: 0

Presence – always

Direction – src

Object type – GstPad

Properties

aggregate-mode 

“aggregate-mode” Rtp-amr-pay-aggregate-mode *

Whether to send out audio frames immediately or aggregate them until a packet is full.

Flags : Read / Write

Default value : auto (-1)

alignment-threshold 

“alignment-threshold” guint64

Timestamp alignment threshold in nanoseconds

Flags : Read / Write

Default value : 40000000

discont-wait 

“discont-wait” guint64

Window of time in nanoseconds to wait before creating a discontinuity

Flags : Read / Write

Default value : 1000000000

max-ptime 

“max-ptime” gint64

Maximum duration of the packet data in ns (-1 = unlimited up to MTU)

Flags : Read / Write

Default value : 18446744073709551615

Named constants

Rtp-amr-pay-aggregate-mode

Members

auto (-1) – Automatic: zero-latency if upstream is live, otherwise aggregate frames until packet is full.
zero-latency (0) – Zero Latency: always send out frames right away, do not wait for more frames to fill a packet.
aggregate (1) – Aggregate: collect audio frames until we have a full packet or the max-ptime limit is hit (if set).

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