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.
This commit is contained in:
Wim Taymans 2008-04-25 07:56:12 +00:00
parent 4a7cf24fcd
commit bd8c40c014
13 changed files with 273 additions and 52 deletions

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@ -1,3 +1,21 @@
2008-04-25 Wim Taymans <wim.taymans@collabora.co.uk>
* 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 Sebastian Dröge <slomo@circular-chaos.org>
* tests/check/elements/multifile.c:

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@ -3361,7 +3361,7 @@ gst_avi_demux_combine_flows (GstAviDemux * avi, avi_stream_context * stream,
/* if we get here, all other pads were unlinked and we return
* NOT_LINKED then */
done:
GST_LOG_OBJECT (avi, "cobined return %s", gst_flow_get_name (ret));
GST_LOG_OBJECT (avi, "combined return %s", gst_flow_get_name (ret));
return ret;
}

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@ -1,12 +1,12 @@
v=0
o=- 1188340656180883 1 IN IP4 192.168.1.110
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=video 5000 RTP/AVP 96
c=IN IP4 0.0.0.0
c=IN IP4 127.0.0.1
a=rtpmap:96 H263-1998/90000
m=audio 5002 RTP/AVP 8
c=IN IP4 0.0.0.0
c=IN IP4 127.0.0.1

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@ -6,12 +6,17 @@
VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998"
AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
gst-launch -v gstrtpbin name=rtpbin \
#DEST=192.168.1.126
DEST=localhost
LATENCY=100
gst-launch -v gstrtpbin name=rtpbin latency=$LATENCY \
udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
rtpbin.send_rtcp_src_0 ! udpsink host=$DEST port=5005 sync=false async=false \
udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \
rtpbin. ! rtppcmadepay ! alawdec ! alsasink \
rtpbin. ! rtppcmadepay ! alawdec ! alsasink buffer-time=10000 \
udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
rtpbin.send_rtcp_src_1 ! udpsink host=$DEST port=5007 sync=false async=false

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@ -1,12 +1,12 @@
v=0
o=- 1188340656180883 1 IN IP4 192.168.1.110
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=video 5000 RTP/AVP 96
c=IN IP4 0.0.0.0
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000
m=audio 5002 RTP/AVP 8
c=IN IP4 0.0.0.0
c=IN IP4 127.0.0.1

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@ -2,16 +2,57 @@
#
# 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=100 \
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 sync=false async=false \
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 ! alsasink slave-method=1 \
rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=$DEST sync=false async=false

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@ -0,0 +1,10 @@
v=0
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=video 5000 RTP/AVP 96
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000

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@ -0,0 +1,36 @@
#!/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
#
# .-------. .----------. .---------. .-------. .-----------.
# 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 |
# '-------' '----------'
# 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"
# 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
gst-launch -v gstrtpbin name=rtpbin \
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

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@ -0,0 +1,9 @@
v=0
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=audio 5002 RTP/AVP 8
c=IN IP4 127.0.0.1

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@ -2,11 +2,34 @@
#
# A simple RTP receiver
#
# 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 | |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 caps of the sender RTP stream. This is usually negotiated out of band with
# SDP or RTSP.
AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
# 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
gst-launch -v gstrtpbin name=rtpbin \
udpsrc caps=$AUDIO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! rtppcmadepay ! alawdec ! alsasink \
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! udpsink port=5003 sync=false async=false
udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false

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@ -1,10 +1,35 @@
#!/bin/sh
#
# A simple RTP server
# sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on
# port 5003. The destination is 127.0.0.1.
# the receiver RTCP reports are received on port 5007
#
# .-------. .-------. .-------. .----------. .-------.
# |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
# '-------' '-------' '-------' | | '-------'
# | |
# | | .-------.
# | | |udpsink| RTCP
# | send_rtcp->sink | port=5003
# .-------. | | '-------' sync=false
# RTCP |udpsrc | | | async=false
# port=5007 | src->recv_rtcp |
# '-------' '----------'
# change this to send the RTP data and RTCP to another host
DEST=127.0.0.1
#AELEM=alsasrc
AELEM=audiotestsrc
# PCMA encode from an the source
ASOURCE="$AELEM ! audioconvert"
AENC="alawenc ! rtppcmapay"
gst-launch -v gstrtpbin name=rtpbin \
alsasrc ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \
rtpbin.send_rtp_src_0 ! udpsink port=5000 \
rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0
$ASOURCE ! $AENC ! rtpbin.send_rtp_sink_0 \
rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \
rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0

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@ -11,12 +11,14 @@ VOFFSET=0
VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1"
DEST=192.168.1.126
gst-launch -v gstrtpbin name=rtpbin \
v4l2src ! $VCAPS ! videorate ! ffmpegcolorspace ! ffenc_h263p ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
rtpbin.send_rtp_src_0 ! queue ! udpsink port=5000 ts-offset=$AOFFSET \
rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
rtpbin.send_rtp_src_0 ! queue ! udpsink host=$HOST port=5000 ts-offset=$AOFFSET \
rtpbin.send_rtcp_src_0 ! udpsink host=$HOST port=5001 sync=false async=false \
udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
alsasrc ! audioconvert ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
rtpbin.send_rtp_src_1 ! queue ! udpsink port=5002 ts-offset=$VOFFSET \
rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
rtpbin.send_rtp_src_1 ! queue ! udpsink host=$HOST port=5002 ts-offset=$VOFFSET \
rtpbin.send_rtcp_src_1 ! udpsink host=$HOST port=5003 sync=false async=false \
udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

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@ -1,26 +1,78 @@
#!/bin/sh
#
# A simple RTP server
# sends the output of v4l2src as h264 encoded RTP on port 5000, RTCP is sent on
# port 5001. The destination is 127.0.0.1.
# the video receiver RTCP reports are received on port 5005
# sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on
# port 5003. The destination is 127.0.0.1.
# the receiver RTCP reports are received on port 5007
#
# .-------. .-------. .-------. .----------. .-------.
# |v4lssrc| |h264enc| |h264pay| | rtpbin | |udpsink| RTP
# | src->sink src->sink src->send_rtp send_rtp->sink | port=5000
# '-------' '-------' '-------' | | '-------'
# | |
# | | .-------.
# | | |udpsink| RTCP
# | send_rtcp->sink | port=5001
# .-------. | | '-------' sync=false
# RTCP |udpsrc | | | async=false
# port=5005 | src->recv_rtcp |
# '-------' | |
# | |
# .-------. .-------. .-------. | | .-------.
# |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
# '-------' '-------' '-------' | | '-------'
# | |
# | | .-------.
# | | |udpsink| RTCP
# | send_rtcp->sink | port=5003
# .-------. | | '-------' sync=false
# RTCP |udpsrc | | | async=false
# port=5007 | src->recv_rtcp |
# '-------' '----------'
#
# ideally we should transport the properties on the RTP udpsink pads to the
# receiver in order to transmit the SPS and PPS earlier.
# change this to send the RTP data and RTCP to another host
DEST=127.0.0.1
# tuning parameters to make the sender send the streams out of sync. Can be used
# ot test the client RTCP synchronisation.
#VOFFSET=500000000
VOFFSET=900000000
AOFFSET=0
VOFFSET=0
# H264 encode from a v4l2src
# H264 encode from the source
VELEM="v4l2src"
#VELEM="videotestsrc is-live=1"
VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1"
VSOURCE="v4l2src ! $VCAPS ! videorate ! ffmpegcolorspace"
VSOURCE="$VELEM ! $VCAPS ! queue ! videorate ! ffmpegcolorspace"
VENC="x264enc byte-stream=true bitrate=300 ! rtph264pay"
# PCMA encode from an alsasrc
ASOURCE="alsasrc ! audioconvert"
VRTPSINK="udpsink port=5000 host=$DEST ts-offset=$VOFFSET name=vrtpsink"
VRTCPSINK="udpsink port=5001 host=$DEST sync=false async=false name=vrtcpsink"
VRTCPSRC="udpsrc port=5005 name=vrtpsrc"
# PCMA encode from an the source
AELEM="alsasrc"
#AELEM="audiotestsrc is-live=1"
ASOURCE="$AELEM ! queue ! audioconvert"
AENC="alawenc ! rtppcmapay"
ARTPSINK="udpsink port=5002 host=$DEST ts-offset=$AOFFSET name=artpsink"
ARTCPSINK="udpsink port=5003 host=$DEST sync=false async=false name=artcpsink"
ARTCPSRC="udpsrc port=5007 name=artpsrc"
gst-launch -v gstrtpbin name=rtpbin \
$VSOURCE ! $VENC ! rtpbin.send_rtp_sink_0 \
rtpbin.send_rtp_src_0 ! queue ! udpsink port=5000 ts-offset=$AOFFSET \
rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtp_src_0 ! $VRTPSINK \
rtpbin.send_rtcp_src_0 ! $VRTCPSINK \
$VRTCPSRC ! rtpbin.recv_rtcp_sink_0 \
$ASOURCE ! $AENC ! rtpbin.send_rtp_sink_1 \
rtpbin.send_rtp_src_1 ! queue ! udpsink port=5002 ts-offset=$VOFFSET \
rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
rtpbin.send_rtp_src_1 ! $ARTPSINK \
rtpbin.send_rtcp_src_1 ! $ARTCPSINK \
$ARTCPSRC ! rtpbin.recv_rtcp_sink_1