Hi, Using: gstreamer-0.10.32 gst-plugins-base-0.10.32 gst-plugins-good-0.10.26 gst-plugins-bad-0.10.20 gst-plugins-ugly-0.10.16 Running on an openembedded linux build on a TI OMAPL138 I cannot get playback of AIFF to be perfect. My pipeline has had to be modified to include audioconvert with capabilities set. It is as follows: gst-launch -v souphttpsrc location=<a href="http://some">http://some</a> aif file.aif ! queue2 ! decodebin2 ! audioconvert ! audio/x-raw-int,channels=2,endianness=1234 ! alsasink Now this produces audio which plays ok for the most part, but if I pause it will swap channels, if I pause again the channels remain un-swapped, if I let the complete track play it will swap the channels at random times during playback. I have installed sox and used wget to pipe the track into sox, it plays fine when I do this. If I install mplayer and use it to playback the track, I get channel swapping as before. So this tells me that my kernel and ALSA layers are fine, the issue is somewhere within gstreamer. I output the decoded audio to a file and then took the file onto my Linux PC, it played back slowly. I then used the following aplay command to get it to playback correctly: aplay -c 2 -f S16_LE -r 88200 aifFile.raw The file playback completes without any channel swapping. For the above pipeline I get the following output from gstreamer: decodebin20.GstDecodePad:src0.GstProxyPad:proxypad2: caps = audio/x-raw-int, width=(int)16, depth=(int)16, channels=(int)2, endianness=(int)4321, rate=(int)44100, signed=(boolean)true /GstPipeline:pipeline0/ GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0 /GstCapsFilter:capsfilter0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock Decodebin correctly reports 16 bit audio at 44.1khz, but even without the audioconvert and caps, gstreamer will convert this to 32 bit. If I try the pipeline without the caps and play 24 bit aiff at 88.2 or 96kHz it plays fine. Why does aiff at 44.1kHz/16bit not playback correctly through gstreamer? Thanks, Stuart