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GObject ╰── GInitiallyUnowned ╰── GstObject ╰── GstElement ╰── GstRTPBasePayload ╰── GstRtpAMRPay
Payload AMR audio into RTP packets according to RFC 3267. For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
1 |
gst-launch-1.0 -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink |
plugin |
rtp |
author |
Wim Taymans <wim.taymans@gmail.com> |
class |
Codec/Payloader/Network/RTP |
name |
sink |
direction |
sink |
presence |
always |
details |
audio/AMR, channels=(int)1, rate=(int)8000 |
audio/AMR-WB, channels=(int)1, rate=(int)16000 |
name |
src |
direction |
source |
presence |
always |
details |
application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, crc=(string)0, robust-sorting=(string)0, interleaving=(string)0, mode-set=(int)[ 0, 7 ], mode-change-period=(int)[ 1, 2147483647 ], mode-change-neighbor=(string){ 0, 1 }, maxptime=(int)[ 20, 2147483647 ], ptime=(int)[ 20, 2147483647 ] |
application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)16000, encoding-name=(string)AMR-WB, encoding-params=(string)1, octet-align=(string)1, crc=(string)0, robust-sorting=(string)0, interleaving=(string)0, mode-set=(int)[ 0, 7 ], mode-change-period=(int)[ 1, 2147483647 ], mode-change-neighbor=(string){ 0, 1 }, maxptime=(int)[ 20, 2147483647 ], ptime=(int)[ 20, 2147483647 ] |