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    <div class="moz-cite-prefix">On 11/4/20 1:29 am, Fabio Zingaro
      wrote:<br>
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cite="mid:CAKO1X7_z0KbdsFHjuu=JvzV89DL5oGBhriYed=43GzwHPpymyw@mail.gmail.com">
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              <div dir="ltr">Hi guys,
                <div><br>
                </div>
                <div>i need your help on 2 topics:</div>
                <div><br>
                </div>
                <div><b>1) Build problems on embedded yocto linux</b></div>
                <div>i'm trying to build this example (<a
href="https://github.com/centricular/gstwebrtc-demos/tree/master/sendonly"
                    moz-do-not-send="true">https://github.com/centricular/gstwebrtc-demos/tree/master/sendonly</a>)
                  on a Evaluation Board ARMv7l with gstreamer 1.14.4
                  installed (with all plugins, include files, etc).</div>
                <div><br>
                </div>
                <div>when i launch make i obtain this:</div>
                <div><br>
                </div>
                <div>```</div>
                <div>webrtc-unidirectional-h264.c: In function
                  'create_receiver_entry':</div>
                <div>webrtc-unidirectional-h264.c:177:3: error: unknown
                  type name 'GstWebRTCRTPTransceiver'</div>
                <div>   GstWebRTCRTPTransceiver *trans;</div>
                <div>   ^~~~~~~~~~~~~~~~~~~~~~~</div>
                <div>In file included from
                  /usr/include/glib-2.0/glib.h:31:0,</div>
                <div>                 from
                  webrtc-unidirectional-h264.c:2:</div>
                <div>webrtc-unidirectional-h264.c:213:40: error:
                  'GstWebRTCRTPTransceiver' undeclared (first use in
                  this function)</div>
                <div>   trans = g_array_index (transceivers,
                  GstWebRTCRTPTransceiver *, 0); </div>
                <div>```</div>
                <div><br>
                </div>
                <div>on a Ubuntu Linux x86_64 18.04 with gstreamer
                  1.14.5 it compiles and run without problems.</div>
                <div><br>
                </div>
                <div><u>Comparing *.so, *.pc and *.h files on embedded
                    board and linux machine i don't see differences.</u></div>
                <div><br>
                </div>
                <div>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.</div>
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    </blockquote>
    <p>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.<br>
    </p>
    <blockquote type="cite"
cite="mid:CAKO1X7_z0KbdsFHjuu=JvzV89DL5oGBhriYed=43GzwHPpymyw@mail.gmail.com">
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                <div><b>2) Bridging RTSP stream to webrtc web page</b></div>
                <div>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.</div>
                <div>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.</div>
                <div><br>
                </div>
                <div>this is the pipeline: </div>
                <div>```</div>
                <div>
                  <div>receiver_entry->pipeline = gst_parse_launch ("
                    rtspsrc location=//" RTSPSRC " "</div>
                  <div><span style="white-space:pre">     </span>"!
                    webrtcbin name=webrtcbin stun-server=stun://"
                    STUN_SERVER " "</div>
                  <div><span style="white-space:pre">     </span>"! queue !
                    rtph264depay ! rtph264pay !
                    application/x-rtp,media=video,encoding-name=H264,payload=96
                    ! webrtcbin. ", &error);</div>
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    </blockquote>
    <p>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:<br>
    </p>
    <div>```</div>
    <div>
      <div>receiver_entry->pipeline = gst_parse_launch (<span style="white-space:pre"></span>"webrtcbin
        name=webrtcbin stun-server=stun://" STUN_SERVER " <br>
      </div>
      <div><span style="white-space:pre"> </span>" rtspsrc location=//"
        RTSPSRC " ! queue ! rtph264depay ! rtph264pay !
        application/x-rtp,media=video,encoding-name=H264,payload=96 !
        webrtcbin. ", &error);</div>
    </div>
    <p>```</p>
    <p>Regards,</p>
    <p>Jan.<br>
    </p>
    <blockquote type="cite"
cite="mid:CAKO1X7_z0KbdsFHjuu=JvzV89DL5oGBhriYed=43GzwHPpymyw@mail.gmail.com">
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                <div>```</div>
                <div><br>
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                <div>Thanks.</div>
                <div><br>
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                <div>Fabio</div>
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      <pre class="moz-quote-pre" wrap="">_______________________________________________
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