Hello dear devs! I am new to gstreamer and it seems to be an awesome tool, I downloaded gstreamer-1.0 SDK for windows and everything works fine, but I have one problem. I want to stream a wav file to another client, which in fact is a VoiP client. Everything works fine, by using decodebin and rtpL16pay into a rtpbin. However it is only sending with payload type 98, but my client needs to get pt 11 rtp pakets. I tried with g_object_set, but it doesn't change a thing. g_object_set(GST_OBJECT(rtpL16pay), "pt", 11, NULL); Unfortunatly I cant post the complete code now since I am on vacation. If anybody could give me hint here, how to change the payload type I would be very thankful. Kind regards JB _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On Sa, 2015-04-11 at 16:51 +0200, Johannes Bauer wrote:
> Hello dear devs! > > I am new to gstreamer and it seems to be an awesome tool, I downloaded gstreamer-1.0 SDK for windows and everything works fine, but I have one problem. > > I want to stream a wav file to another client, which in fact is a VoiP client. > Everything works fine, by using decodebin and rtpL16pay into a rtpbin. > However it is only sending with payload type 98, but my client needs to get pt 11 rtp pakets. > I tried with g_object_set, but it doesn't change a thing. > > g_object_set(GST_OBJECT(rtpL16pay), "pt", 11, NULL); > > Unfortunatly I cant post the complete code now since I am on vacation. > If anybody could give me hint here, how to change the payload type I would be very thankful. capsfilter and providing the static pt caps, e.g. gst-launch-1.0 audiotestsrc ! audioconvert ! rtpL16pay ! application/x-rtp,payload=11 ! fakesink The property only seems to work for the dynamic pt range. -- Sebastian Dröge, Centricular Ltd · http://www.centricular.com _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel signature.asc (968 bytes) Download Attachment |
In reply to this post by Johannes Bauer
Hi,
I am facing the same problem. Are you able to solve your this problem. Thank You -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Hi amsts!
You can solve this using a capsfilter. This one is for a MPA pipeline (pt=14): rtpaudio = gst_element_factory_make("rtpmpapay", "rtpaudio"); caps_rtp = gst_caps_new_simple("application/x-rtp", "media", G_TYPE_STRING, "audio", "clock-rate", G_TYPE_INT, 90000, "encoding-name", G_TYPE_STRING, "MPA", "channels", G_TYPE_INT, 1, "payload", G_TYPE_INT, 14, NULL); rtpbin = gst_element_factory_make("rtpbin", "rtpbin"); then u have to link the payloader and the rtpbin with this function: ret = gst_element_link_pads_filtered(rtpaudio, "src", rtpbin, "send_rtp_sink_0", caps_rtp); if (ret != TRUE) { Log(LOG_TYPE_ERROR, "RTP", "Streaming Elements rtpaudio and rtpbin could not be linked!"); return -1; } Hope this helps Kind regards JB _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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