Pad negotiation at udpsrc element
debruyn
debruynels1 at gmail.com
Mon Nov 14 06:26:10 UTC 2016
Good day guys,
So the current system is set up to push a stream from an ARm board to the
webserver. On the board it is a simple rtspsrc from the camera into a
udpsink. Then on the server a udpsrc receives the stream and does work. This
works a hundred persent with video only streams. Now i am trying to add
audio to this system.
On the camera you are able to send a audio(from a mic) and video in the same
rtsp stream. I know this works as i am able to record locally on the board a
audio and video file. now when sending this over udp to the server i am
running into an issue. When setting up a remote recording and creating the
following pipe:
* | queue | rtph264depay | h264parse | queue \
udpsrc | tee <
| mp4mux | filesink
| queue | rtpmpadepay | mpegaudioparse | queue /*
I am getting a not negotiated error when going from paused to playing:
*Debugging information: gstbasesrc.c(2950): gst_base_src_loop ():
/GstPipeline:main-pipeline/GstUDPSrc:source:
streaming stopped, reason not-negotiated (-4)*
When removing just the audio or video part of that pipeline i do not receive
that error and it seems to be working fine.
I did some checking and think it might be a capability problem as the pads
can't be negotiated between the elements. Is this true? And if it is what
can be a fix? currently i am setting up my udpsrc as follows :
*g_object_set(gstSource,"port",moProfileInstance->getDestPort(),"close-socket",FALSE,"caps",gst_caps_new_simple
("application/x-rtp",NULL),NULL);*
If the caps is not the problem what can it be then? As i said when not
sending the stream over udp and doing the recording on the board, that pipe
works a hundred persent fine as reference here is the pipeline on the board:
*rtspsrc | queue | rtph264depay | h264parse | queue \
| mp4mux | filesink
rtspsrc | queue | rtpmpadepay | mpegaudioparse | queue /*
Any help or insight will be musc appreciated
Regards
DB
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