Playing and streaming, is it possible?
Paulo Paiva
paivalhao at gmail.com
Thu Jun 2 07:50:47 PDT 2011
Hello all!
I'm struggling here trying to play the video acquired from the web-cam
and stream it simultaneously but it isn't working. Here goes my pipeline
(inspired in the examples given):
#!/bin/sh
# Destination of the stream
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=0
AOFFSET=0
# Video setup
VELEM="v4l2src "
VCAPS="video/x-raw-yuv,width=352,height=288,framerate=20/1"
VSOURCE="$VELEM ! $VCAPS ! queue ! ffmpegcolorspace"
VENC="x264enc tune=zerolatency byte-stream=true bitrate=550 threads=0
speed-preset=3"
# Video transmission setup
VRTPSINK="udpsink port=5000 host=$DEST ts-offset=$VOFFSET name=vrtpsink"
VRTCPSINK="udpsink port=5001 host=$DEST sync=false async=false
name=vrtcpsink"
VRTCPSRC="udpsrc port=5005 name=vrtpsrc"
# Audio setup
AELEM="pulsesrc"
ASOURCE="$AELEM ! queue ! audioconvert"
AENC=" faac! rtpmp4apay"
# Audio transmission setup
ARTPSINK="udpsink port=5002 host=$DEST ts-offset=$AOFFSET name=artpsink"
ARTCPSINK="udpsink port=5003 host=$DEST sync=false async=false
name=artcpsink"
ARTCPSRC="udpsrc port=5007 name=artpsrc"
# Pipeline construction
gst-launch -v gstrtpbin name=rtpbin \
$VSOURCE ! tee name=v ! autovideosink v. ! $VENC ! rtph264pay !
rtpbin.send_rtp_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 ! $ARTPSINK \
rtpbin.send_rtcp_src_1 ! $ARTCPSINK \
$ARTCPSRC ! rtpbin.recv_rtcp_sink_1
Has all can see I got a tee after the acquisition, after that
autovideosink in order to displays and I pass on the video flux for
streaming.
The output of execution is:
$ sh v4l2server.sh
Setting pipeline to PAUSED ...
/GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: actual-buffer-time = 47551927
/GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: actual-latency-time = 9977
/GstPipeline:pipeline0/GstPulseSrc:pulsesrc0.GstPad:src: caps =
audio/x-raw-int, endianness=(int)1234, signed=(boolean)true,
width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)1,
channel-positions=(GstAudioChannelPosition)<
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO >
ERROR: Pipeline doesn't want to pause.
ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could
not negotiate format
Additional debug info:
gstbasesrc.c(2811): gst_base_src_start ():
/GstPipeline:pipeline0/GstV4l2Src:v4l2src0:
Check your filtered caps, if any
Setting pipeline to NULL ...
/GstPipeline:pipeline0/GstPulseSrc:pulsesrc0.GstPad:src: caps = NULL
Freeing pipeline ...
The last message makes suggestion to look at the caps, but removing the
tee element and the autovideosink it works.
Any ideas/suggestions would be great!!
Thanks all for help!!
Regards,
Paulo Paiva
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20110602/647f1690/attachment.htm>
More information about the gstreamer-devel
mailing list