webrtc demos

[Jan Schmidt]

On 11/4/20 1:29 am, Fabio Zingaro wrote:
> Hi guys,
>
> i need your help on 2 topics:
>
> *1) Build problems on embedded yocto linux*
> i'm trying to build this example
> (https://github.com/centricular/gstwebrtc-demos/tree/master/sendonly)
> on a Evaluation Board ARMv7l with gstreamer 1.14.4 installed (with all
> plugins, include files, etc).
>
> when i launch make i obtain this:
>
> ```
> webrtc-unidirectional-h264.c: In function 'create_receiver_entry':
> webrtc-unidirectional-h264.c:177:3: error: unknown type name
> 'GstWebRTCRTPTransceiver'
>    GstWebRTCRTPTransceiver *trans;
>    ^~~~~~~~~~~~~~~~~~~~~~~
> In file included from /usr/include/glib-2.0/glib.h:31:0,
>                  from webrtc-unidirectional-h264.c:2:
> webrtc-unidirectional-h264.c:213:40: error: 'GstWebRTCRTPTransceiver'
> undeclared (first use in this function)
>    trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0); 
> ```
>
> on a Ubuntu Linux x86_64 18.04 with gstreamer 1.14.5 it compiles and
> run without problems.
>
> _Comparing *.so, *.pc and *.h files on embedded board and linux
> machine i don't see differences._
>
> Searching on google i've found some similar problems related to GNU
> Make version. I've 4.2.1 on the board and 4.1 on the Linux PC.

I can't think of any good reason for that error. I don't know how GNU
Make could affect things, unless the compilation command-line is
different between the versions - but I'd expect you'd get a lot more
errors than just the one if so.

>
> *2) Bridging RTSP stream to webrtc web page*
> continuing my experiments on ubuntu x86_64 i've tried to change the
> pipeline removing v4l2 source and inserting a videotestsrc with no
> problems, it worked.
> Now i'm trying to read a RTSP stream from an IP camera but i doesn't
> work and i've no errors on console.
>
> this is the pipeline: 
> ```
> receiver_entry->pipeline = gst_parse_launch (" rtspsrc location=//"
> RTSPSRC " "
> "! webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " "
> "! queue ! rtph264depay ! rtph264pay !
> application/x-rtp,media=video,encoding-name=H264,payload=96 !
> webrtcbin. ", &error);

Your pipeline description is out of order here. You'll end up connecting
the rtspsrc to webrtcbin first, and then creating a 2nd chain with queue
! .../ ! webrtcbin. that doesn't produce any data and stops the whole
pipeline from pre-rolling. Try this:

```
receiver_entry->pipeline = gst_parse_launch ("webrtcbin name=webrtcbin
stun-server=stun://" STUN_SERVER "
" rtspsrc location=//" RTSPSRC " ! queue ! rtph264depay ! rtph264pay !
application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin.
", &error);

```

Regards,

Jan.

> ```
>
> Thanks.
>
> Fabio
>
>
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