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GObject ╰── GInitiallyUnowned ╰── GstObject ╰── GstElement ╰── GstRTPBasePayload ╰── GstRTPBaseAudioPayload ╰── GstRtpL16Pay
Payload raw audio into RTP packets according to RFC 3551. For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
1 |
gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink |
plugin |
rtp |
author |
Wim Taymans <wim.taymans@gmail.com> |
class |
Codec/Payloader/Network/RTP |
name |
sink |
direction |
sink |
presence |
always |
details |
audio/x-raw, format=(string)S16BE, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ] |
name |
src |
direction |
source |
presence |
always |
details |
application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)[ 1, 2147483647 ], encoding-name=(string)L16, channels=(int)[ 1, 2147483647 ] |
application/x-rtp, media=(string)audio, encoding-name=(string)L16, payload=(int)10, clock-rate=(int)44100 | |
application/x-rtp, media=(string)audio, encoding-name=(string)L16, payload=(int)11, clock-rate=(int)44100 |