<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body text="#000000" bgcolor="#ffffff"> On 06/02/2011 03:58 PM, Jose Antonio Santos Cadenas wrote: <blockquote cite="mid:BANLkTi=nBUxVr9agjYtSksGvz7JyhEgPEA@mail.gmail.com" type="cite"> <pre wrap="">Hi, 2011/6/2 Paulo Paiva <a class="moz-txt-link-rfc2396E" href="mailto:paivalhao@gmail.com"><paivalhao@gmail.com></a>: </pre> <blockquote type="cite"> <pre wrap="">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!! </pre> </blockquote> <pre wrap=""> Have you tried using a queue on each tee branch as is recomended on tee documentation page? <a class="moz-txt-link-freetext" href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-tee.html#gstreamer-plugins-tee.description">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-tee.html#gstreamer-plugins-tee.description</a> Regards </pre> </blockquote> Thanks for the quick reply Jose Cadenas! Well after doing a easier version of what I'm trying to do, just record and play, I came up with the solution (lines changed): VCAPS="video/x-raw-yuv,width=352,height=288,framerate=20/1 ! tee name=v" gst-launch -v gstrtpbin name=rtpbin             \     $VSOURCE ! $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   \           v. ! queue ! ffmpegcolorspace ! autovideosink    \     $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 Cheers all! Paulo Paiva </body> </html>