Hi, gstreamer-devel:<br><br> Also the gstrtpbin will be used by uridecodebin because gstrtpbin will be used in rtspsrc which is a source element handles RTSP medias and inherits from GstBin.<br><br>Eric Zhang<br><br><div class="gmail_quote">
2008/12/19 Eric Zhang <span dir="ltr"><<a href="mailto:nicolas.m.zhang@gmail.com">nicolas.m.zhang@gmail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi, gstreamer-devel:<br><br> 1. Seems you should write some codes to achieve your goal because you mentioned you have other non-gstreamer codes to handle the audio data. So, you can read the description and algorithms on RTP lip-synchronization and do it yourself. "gst_rtp_bin_associate" is a good reference as well.<br>
<br> 2. uridecodebin will select rtspsrc and autoplug other stuffs when you play a RTSP movie. I skimmed the source codes of uridecodebin and noticed there is function named "gen_source_element" and it also calls "gst_element_make_from_uri" as well.<br>
<br>Eric Zhang<br><br><div class="gmail_quote">2008/12/18 Jon Burgess <span dir="ltr"><<a href="mailto:jkburges@gmail.com" target="_blank">jkburges@gmail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div><div></div><div class="Wj3C7c">
<div><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div> </div>
<div> "gst_rtp_bin_associate" in gstrtpbin.c do what you want. But I think it needs improvements. :)</div>
<div> </div></blockquote></div><br></div>Thanks Olivier and Eric...<br><br>From what I can tell, this bit of code handles synchronization between RTP streams within the same session, or in other words, all streams handled by one RTP bin will be synced.<br>
<br>But my case is slightly different because of the fact that one stream (video) will be received by a gstreamer pipeline, while the other (audio) will be received by legacy code.<br><br>So my thought of how to sync them (either when playing live, or playing back recorded data) would be to have gstreamer timestamp the buffers with something like a unix (wallclock) timestamp, have the legacy code do the same, and then things should stay synced.<br>
<br>Would this work, or would timestamping like this upset gstreamer?<br><br>Also, I'm using a uridecodebin - would that use a gstrtpbin internally for an RTSP source?<br><br>Jono<br>
<br></div></div><div class="Ih2E3d">------------------------------------------------------------------------------<br>
SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada.<br>
The future of the web can't happen without you. Join us at MIX09 to help<br>
pave the way to the Next Web now. Learn more and register at<br>
<a href="http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/" target="_blank">http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/</a><br>_______________________________________________<br>
gstreamer-devel mailing list<br>
<a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">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>
<br></div></blockquote></div><br>
</blockquote></div><br>