rtpbin does not add "somtimes" pads

Benjamin Trent ben.w.trent at gmail.com
Tue Sep 23 08:03:15 PDT 2014


It seems that no request pads with that are source pads. Getting the same
error here:
(gstrtpmediastream:7266): GStreamer-CRITICAL **: Padname send_rtcp_src_0 is
not unique in element rtpbin, not adding


On Tue, Sep 23, 2014 at 8:55 AM, Benjamin Trent <ben.w.trent at gmail.com>
wrote:

> I am trying to create a dynamic pipeline through adding pads to a rtpbin
> object dynamically.
>
> Have a callback handling the pad-added signal but it is never called for
> the "send_rtp_src_u%" pad that is to be created after creating the
> "send_rtp_sink_%u" pad. The Sending sink is added just fine but the rtpbin,
> when trying to add the sending source fails.
>
> The error is "(gstrtpmediastream:4574): GStreamer-CRITICAL **: Padname
> send_rtp_src_0 is not unique in element rtpbin, not adding".
>
> I have verified that the pad does not already exist(iterating through the
> list).
>
> GstElement* rtp_bin = gst_bin_get_by_name(GST_BIN(pipeline),"rtpbin");
> GstPadTemplate* rtp_sink_template =
> gst_element_class_get_pad_template(GST_ELEMENT_GET_CLASS(rtp_bin),
> send_rtp_sink_%u");
> GstCaps* cap = GST_STATIC_CAPS("application/x-rtp, media=video");
> GstPad* rtpsink = gst_element_request_pad(rtp_bin, rtp_sink_template,
> NULL, cap);
>
>
> This is just some of my code. More can be made available if need be.
>
> The pipeline is NOT playing and was made from a parse launch
>
> Pipeline = "rtpbin name=rtpbin appsrc name=videoappsrc
> caps=\"application/x-rtp, media=video, clock-rate=90000,
> encoding-name=VP8-DRAFT-IETF-01, payload=96\" is-live=true ! tee
> name=videotee "
> "appsrc name=audioappsrc caps=\"application/x-rtp, media=audio,
> clock-rate=8000, encoding-name=PCMA, payload=8\" is-live=true ! tee
> name=audiotee"
>
>
>
> The overall goal is to be able to add an arbitrary number of tee splits
> for the rtpbin for each stream request so that the party getting the stream
> can provide control packets for each session created uniquely. This is so
> time-outs can be handle, audio/video sync for each stream requested, and
> all the other good things that rtcp brings to the table through the rtpbin.
>
> Thanks!
>
> P.S. I am using the newest release of gstreamer-1.0 on a xubuntu
> installation.
>
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