This bug is evident running openembedded Linux on an arm core.

The file was tested, on the same system using wget and piping it into sox where it was decoded and playback was perfect.

My system has the following gstreamer revisions:

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

Using the following pipeline:

{{{
gst-launch -v souphttpsrc location=http://192.168.0.144/testAudio/Travis/The%20Man%20Who/01%20Writing%20To%20Reach%20You.aif ! queue2 max-size-buffers=0 max-size-time=0 max-size-bytes=500000 ! decodebin2 ! audioconvert ! audio/x-raw-int,endianness=1234,channels=2 ! audioresample ! alsasink sync=false async=false
}}}

As playback begins a channel swap always seems to occur approximately 10 seconds into the track, then this is repeated at random times during track playback.

Looking at debug from gstreamer, I tried setting this to various levels including 5 shows nothing that looks relevant or an error.

The pipeline produces the following:
{{{
Setting pipeline to PAUSED ...
davinci-pcm: davinci_pcm_open: sram buffer size = 0x2000
GConf Error: Failed to contact configuration server; some possible causes are that you need to enable TCP/IP networking for ORBit, or you have stale NFS locks due to a system crash. See http://projects.gnome.org/gconf/ for information. (Details -  1: Not running within active session)
GConf Error: Failed to contact configuration server; some possible causes are that you need to enable TCP/IP networking for ORBit, or you have stale NFS locks due to a system crash. See http://projects.gnome.org/gconf/ for information. (Details -  1: Not running within active session)
GConf Error: Failed to contact configuration server; some possible causes are that you need to enable TCP/IP networking for ORBit, or you have stale NFS locks due to a system crash. See http://projects.gnome.org/gconf/ for information. (Details -  1: Not running within active session)
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstDecodeBin2:decodebin20/GstTypeFindElement:typefind.GstPad:src: caps = audio/x-aiff
Pipeline is PREROLLED ...
/GstPipeline:pipeline0/GstDecodeBin2:decodebin20/GstAiffParse:aiffparse0.GstPad:sink: caps = audio/x-aiff
/GstPipeline:pipeline0/GstDecodeBin2:decodebin20.GstDecodePad:src0: 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/GstDecodeBin2:decodebin20/GstAiffParse:aiffparse0.GstPad:src: caps = audio/x-raw-int, width=(int)16, depth=(int)16, channels=(int)2, endianness=(int)4321, rate=(int)44100, signed=(boolean)true
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.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/GstAudioConvert:audioconvert0.GstPad:sink: 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/GstDecodeBin2: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/GstAudioResample:resample0.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/GstAudioResample:audioresample0.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
}}}

This happens with any aiff file that can be produced using iTunes and setting the import settings to aiff.

If the caps are altered to include width=16, depth=16 and rate=44100 then there is less channel swapping, but it still happens.

Directing the output of the audioresample to a file and then copying this onto my Linux PC, I can playback the audio, but only when I set aplay to use a rate of 88.2kHz:

aplay -c 2 -f S16_LE -r 88200 aifFile.raw

If I use 44100 it plays back slow.

When I test using the same file on a Linux running Ubuntu 10.04, the issue does not appear, it is only on the embedded system.

I must stress that you can use any aiff file.