<div>Thanks, Wim.</div><div><br></div>Sorry for problem 1. <div>My server can not control the flow because I have modified rtsp server by g_object_set (G_OBJECT (udpsink0), "sync", FALSE, NULL);</div><div><div><br> </div><div>When I enable the sync, it works well with mp3parse pipeline.</div><div><br></div><div>I will test the server with h264parse, aacparse and others.</div><div><br></div><div>For problem 2, 3:</div><div>Is there any plugin can auto detect acodec/vcodec of a file?<br> </div><div><br></div><div>BTW: Is there a roadmap for rtsp-server? </div><div><br></div><div>Thanks again.</div></div><br><div class="gmail_quote">2009/9/26 Wim Taymans <span dir="ltr"><<a href="mailto:wim.taymans@gmail.com">wim.taymans@gmail.com</a>></span><br> <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class="im">On Fri, 2009-09-25 at 18:06 +0800, dxssx wrote:<br> > Hi,<br> > I want to use gst-rtsp-server as my vod server, but I found it is too<br> > hard for some reasons:<br> ><br> ><br> > 1. rtsp-server has no flow control in sending packet.<br> > It sends packets as fast as possible which makes client can not decode<br> > in time.<br> <br> </div>for flow control if relies on a gstreamer element that produces correct<br> timestamps. If you are, for example, sending an mp3 file, you need to<br> insert mp3parse after the filesrc so that timestamps are properly set on<br> the data.<br> <div class="im">><br> ><br> > 2. rtsp-server has no dynamic url.<br> > Everytime I add a file into vod list, I need to add a url mapping for<br> > it.<br> <br> </div>You can make a subclass of the mediafactory to dynamically create<br> pipelines. The example base class only accept gst-launch type syntax.<br> <div class="im"><br> ><br> ><br> > 3. rtsp-server doesn't support dynamic pipeline.<br> > Everytime I add a file into vod list, I need to rewrite a new pipeline<br> > to parse and rtppay it.<br> <br> </div>Again this can be implemented using subclasses of the standard example<br> factories.<br> <br> Wim<br> <div class="im">><br> ><br> > I am new here, so I am afraid that I missed something useful in<br> > rtsp-server or gstreamer to overcome the problems above.<br> ><br> ><br> > Does anyone can help me?<br> ><br> ><br> > Thanks.<br> ><br> ><br> </div>> ------------------------------------------------------------------------------<br> > Come build with us! The BlackBerry&reg; Developer Conference in SF, CA<br> > is the only developer event you need to attend this year. Jumpstart your<br> > developing skills, take BlackBerry mobile applications to market and stay<br> > ahead of the curve. Join us from November 9&#45;12, 2009. Register now&#33;<br> > <a href="http://p.sf.net/sfu/devconf" target="_blank">http://p.sf.net/sfu/devconf</a><br> > _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><br> <br> <br> ------------------------------------------------------------------------------<br> Come build with us! The BlackBerry&reg; Developer Conference in SF, CA<br> is the only developer event you need to attend this year. Jumpstart your<br> developing skills, take BlackBerry mobile applications to market and stay<br> ahead of the curve. Join us from November 9&#45;12, 2009. Register now&#33;<br> <a href="http://p.sf.net/sfu/devconf" target="_blank">http://p.sf.net/sfu/devconf</a><br> _______________________________________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a><br> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><br> </blockquote></div><br>