gstreamer/tests/examples/rtp/client-H264-PCMA.sh
Wim Taymans bd8c40c014 gst/avi/gstavidemux.c: Fix typo in comments.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows):
Fix typo in comments.
* tests/examples/rtp/client-H263p-PCMA.sdp:
* tests/examples/rtp/client-H263p-PCMA.sh:
* tests/examples/rtp/client-H264-PCMA.sdp:
* tests/examples/rtp/client-H264-PCMA.sh:
* tests/examples/rtp/client-H264.sdp:
* tests/examples/rtp/client-H264.sh:
* tests/examples/rtp/client-PCMA.sdp:
* tests/examples/rtp/client-PCMA.sh:
* tests/examples/rtp/server-alsasrc-PCMA.sh:
* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
Add some more docs and fix examples.
2008-04-25 07:56:12 +00:00

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#!/bin/sh
#
# A simple RTP receiver
#
# receives H264 encoded RTP video on port 5000, RTCP is received on port 5001.
# the receiver RTCP reports are sent to port 5006
# receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003.
# the receiver RTCP reports are sent to port 5007
#
# .-------. .----------. .---------. .-------. .-----------.
# RTP |udpsrc | | rtpbin | |h264depay| |h264dec| |xvimagesink|
# port=5000 | src->recv_rtp recv_rtp->sink src->sink src->sink |
# '-------' | | '---------' '-------' '-----------'
# | |
# | | .-------.
# | | |udpsink| RTCP
# | send_rtcp->sink | port=5005
# .-------. | | '-------' sync=false
# RTCP |udpsrc | | | async=false
# port=5001 | src->recv_rtcp |
# '-------' | |
# | |
# .-------. | | .---------. .-------. .--------.
# RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |alsasink|
# port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink |
# '-------' | | '---------' '-------' '--------'
# | |
# | | .-------.
# | | |udpsink| RTCP
# | send_rtcp->sink | port=5007
# .-------. | | '-------' sync=false
# RTCP |udpsrc | | | async=false
# port=5003 | src->recv_rtcp |
# '-------' '----------'
# the destination machine to send RTCP to. This is the address of the sender and
# is used to send back the RTCP reports of this receiver. If the data is sent
# from another machine, change this address.
DEST=127.0.0.1
# this adjusts the latency in the receiver
LATENCY=200
# the caps of the sender RTP stream. This is usually negotiated out of band with
# SDP or RTSP. normally these caps will also include SPS and PPS but we don't
# have a mechanism to get this from the sender with a -launch line.
VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"
AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
gst-launch -v gstrtpbin name=rtpbin latency=$LATENCY \
udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=$DEST sync=false async=false \
udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \
rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=$DEST sync=false async=false