<div dir="ltr">How and when do you change the resolution to generate the error?<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Sat, Jul 6, 2013 at 5:04 PM, ftvfev <span dir="ltr"><<a href="mailto:ftv.dev@gmail.com" target="_blank">ftv.dev@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I am using gstreamer to stream a video. So I am using the 2 scripts:<br>
<br>
Client side :<br>
<br>
DEST=10.2.2.30<br>
<br>
# this adjusts the latency in the receiver<br>
LATENCY=0<br>
<br>
<br>
<br>
VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"<br>
<br>
AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"<br>
<br>
VIDEO_DEC="rtph264depay ! avdec_h264"<br>
<br>
VIDEO_SINK="videoconvert ! autovideosink"<br>
<br>
gst-launch-1.0 -v rtpbin name=rtpbin latency=$LATENCY<br>
\<br>
udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0<br>
\<br>
rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK<br>
\<br>
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0<br>
\<br>
rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=$DEST<br>
sync=false async=false<br>
<br>
Server side:<br>
<br>
DEST=10.2.2.42<br>
<br>
# tuning parameters to make the sender send the streams out of sync. Can<br>
be used<br>
# ot test the client RTCP synchronisation.<br>
#VOFFSET=900000000<br>
VOFFSET=0<br>
AOFFSET=0<br>
<br>
# H264 encode from the source<br>
# VELEM="v4l2src device=/dev/video0"<br>
VELEM="filesrc location=big_buck_bunny_1080p_stereo.ogg ! queue !<br>
decodebin"<br>
<br>
VCAPS="video/x-raw,width=1920,height=1080,framerate=60/1"<br>
VSOURCE="$VELEM ! queue ! videoscale ! videorate ! $VCAPS"<br>
VENC="x264enc subme=7 ref=3 bframes=4 me=hex tune=zerolatency<br>
byte-stream=true ! rtph264pay"<br>
<br>
VRTPSINK="udpsink port=5000 host=$DEST ts-offset=$VOFFSET name=vrtpsink"<br>
VRTCPSINK="udpsink port=5001 host=$DEST sync=false async=false<br>
name=vrtcpsink"<br>
VRTCPSRC="udpsrc port=5005 name=vrtpsrc"<br>
<br>
gst-launch-1.0 -v rtpbin name=rtpbin \<br>
$VSOURCE ! $VENC ! rtpbin.send_rtp_sink_0<br>
\<br>
rtpbin.send_rtp_src_0 ! $VRTPSINK<br>
\<br>
rtpbin.send_rtcp_src_0 ! $VRTCPSINK<br>
\<br>
$VRTCPSRC ! rtpbin.recv_rtcp_sink_0<br>
<br>
It works perfectly. However, when I change the resolution for example to<br>
this 640x480. I get the following error on the client side :<br>
<br>
/GstPipeline:pipeline0/avdec_h264:avdec_h264-0.GstPad:src: caps =<br>
video/x-raw, format=(string)I420, width=(int)640, height=(int)480,<br>
pixel-aspect-ratio=(fraction)4/3, interlace-mode=(string)progressive,<br>
colorimetry=(string)bt601, framerate=(fraction)0/1<br>
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal<br>
data flow error.<br>
Additional debug info:<br>
gstbasesrc.c(2812): gst_base_src_loop ():<br>
/GstPipeline:pipeline0/GstUDPSrc:udpsrc0:<br>
streaming task paused, reason not-negotiated (-4)<br>
Execution ended after 2217228820 ns.<br>
Setting pipeline to PAUSED ...<br>
Setting pipeline to READY ...<br>
Setting pipeline to NULL ...<br>
Freeing pipeline ...<br>
<br>
So what is the reason of this problem and how can I fix it ?<br>
<br>
<br>
<br>
--<br>
View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/H-264-Streaming-Error-when-changing-resolution-tp4660922.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/H-264-Streaming-Error-when-changing-resolution-tp4660922.html</a><br>
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</blockquote></div><br></div>