tsdemux & ffec_h264: ffmpeg :0:: non-existing PPS referenced
ccrisler at mutualink.net
Wed Apr 23 08:10:46 PDT 2014
The problem is in the server pipeline. It looks like you are generating
straight MP2T but your client is decoding MP2T encapsulated in RTP, which
gives the error. Since you seem to want to multiplex (mux) the audio and
video, use straight MP2T. Your server pipeline might be ok. Your client
will be receiving something like application/x-MP2T (not sure of the exact
syntax), no payload, no ssrc, timestamp or seqno. Use the MP2T demuxer and
'T' for the audio and video. I use 0.10 so the element names have changed.
You will have to use gst-inspect | grep -i mp2t to find the proper element
name for the MP2T demuxer. There are examples of this but most are for
0.10, not the 1.x code.
On Tue, Apr 22, 2014 at 7:20 PM, kayoss12 <noureddinetariba at gmail.com>wrote:
> thank you for you answer , to so i try run pepepline client the first ,
> second i run the commande on RPI , but still not working , the Camera run
> on PI , but i can't receive the stream on my PC client
> there are any fault in the pepeline ??
> thanks in advance for your answer
> client :
> gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp, media=(string)video,
> clock-rate=(int)90000, encoding-name=(string)MP2T, payload=(int)33,
> ssrc=(uint)4251539348, \
> timestamp-offset=(uint)2033210929, seqnum-offset=(uint)34671" !
> rtpmp2tdepay \
> ! tsparse ! tsdemux ! "video/x-h264" ! avdec_h264 ! autovideosink
> RPI :
> gst-launch-1.0 rpicamsrc bitrate=1000000 !
> "video/x-h264,width=640,height=480" !\
> queue ! h264parse ! queue ! mux. alsasrc device=hw:1,0 ! audioconvert ! \
> lamemp3enc ! "audio/mpeg"! queue ! mux. mpegtsmux name=mux ! queue ! \
> udpsink host=192.168.1.9 port=5000 -v
> i got this warning
> Warnig client :
> Additional debug info:
> gstrtpbasedepayload.c(381): gst_rtp_base_depayload_chain ():
> Received invalid RTP payload, dropping
> WARNING: from element /GstPipeline:pipeline0/GstRtpMP2TDepay:rtpmp2tdepay0:
> d not decode stream.
> also i have warning in RPI
> WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't
> record audio fast enough
> Additional debug info:
> gstaudiobasesrc.c(848): gst_audio_base_src_create ():
> Dropped 8820 samples. This is most likely because downstream can't keep up
> and is consuming samples too slowly.
> View this message in context:
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the gstreamer-devel