<html dir="ltr"> <head> <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> <style id="owaParaStyle" type="text/css">  </style> </head> <body> <div style="direction:ltr; font-family:Tahoma; color:#000000; font-size:10pt">I'm using gstreamer-0.10.36 (we have some target-specific elements that don't support 1.X). I am receiving a live feed via mpeg-TS encapsulated with RTP. Since this is a live feed I'm getting real timestamps, so they don't start from 0. If I set sync=true on the alsasink the pipeline goes to the PLAYING state and then just stops. Placing an identity element in front of the alsasink shows that a single audio buffer is sent to the alsasink with a timestamp of something like 55:35.0000 (55 minutes in). It appears the alsasink is patiently waiting this amount of time before playing any audio. It plays immediately with sync=false, but then I lose lip sync.<br> <br> Is there any way to tell alsasink that a large discontinuity like this does not denote an incredibly long delay but rather that it should consider this the current time and play the buffers from this timestamp on?<br> </div> <br> <hr> <font face="Arial" color="Gray" size="1"><br> <br> This email and any files transmitted with it are confidential & proprietary to Systems and Software Enterprises, LLC. This information is intended solely for the use of the individual or entity to which it is addressed. Access or transmittal of the information contained in this e-mail, in full or in part, to any other organization or persons is not authorized.<br> </font> </body> </html>