Hi, I'm currently performing some tests with the sync property of alsasink set to false. It's not really perfect since instead of having audio blanks, I now have some instants of audio being dropped (as there were missing words in a sentence). I'm not really satisfied with this, so I keep on investigating. Cheers. <div class="gmail_quote">2011/12/6 Christophe Dongieux <<a href="mailto:christophe.dongieux@gmail.com">christophe.dongieux@gmail.com</a>> <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi, I'm running gstreamer on a device that is not a PC, and after some tests, it appears that the audio clock on this device is slower than it should be. The test to reveal this was to take an audio file and start playing it on both the device and a PC: the file is about 1 hour long, and at the end of it, there was about 1min30sec difference between the time the file ended on the device and on the PC (yeah, big drift!). In the case of a local file playing, this is not a big deal to be a little slower. But in the case of a live playing (in my case MMS radio), there is a drift between the internet stream time and alsasink (local) time, so alsasink starts complaining after about an hour with audio cuts and this message: "Unexpected discontinuity in audio timestamps of more than half a second". There are 2 aproaches to deal with this problem: understand and fix the audio clock rate, or make some tricks in gstreamer/alsasink to avoid the drift. In the case I can't find out the audio clock rate fix, do you have any idea what would be the tricks to make internet radios play fine and long on my device? I saw in alsasink plenty of properties that are related to the clock, but it is not really clear for me what could help me (documentation does not help me so much). Thanks in advance, Christophe Dongieux. </blockquote></div>