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playing 24/96 files from mac music server on dell laptop

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Hello Computer Audiophile community. I just found this site today and it is fantastic. Thanks for all of the useful info. Now onto my questions . . .


Here's my set-up:

living room - dell latitude 620 laptop running xp pro sp3 docked in a port replicator/dock > spdif cable > denon receiver. denon takes care of everything after that.

home office - mac tower with 2TB of external storage running latest OSX version

Everything is loaded as apple lossless onto the office mac machine, I share my library via itunes, and then i play music on my living room stereo via the shared library on itunes on the dell laptop. I'm not using an Airport Express because i read that it does not support 24/96 audio in that it is converted to 16/44 for the wireless transfer. I control the laptop via logmein from another dell laptop I have.


I'd like some way to know whether the digital audio is being passed "as-is" from the office mac all the way to the digital input on the denon. If I have a 24/96 apple lossless file on the mac, is it getting to the dell at 24/96 via the shared library, and then is the dell sending it directly to the receiver without converting, resampling, etc.? The receiver can only display whether its PCM or 5.1 digital input, so is there another way to determine what's actually getting to the receiver? Also, are there specific drivers or xp configurations I should be using to make sure that everything is being passed through cleanly? In the advanced audio properties settings on xp I have the "use spdif" box checked and the default codecs are sigma tel.


I have run some tests using an airport express, playing the files locally on dell via itunes, playing them as flac files from a shared folder on the mac via VLC, playing them the way I described above, etc., and it all sounds very, very similar.


Thanks in advance for any advice!


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Well, in case anyone read this and had similar questions, I found some information which answers some of my questions. Based on various posts I found on this forum (http://www.computeraudiophile.com/node/574?page=1) and others, it seems that when just running iTunes on windows xp and checking the s/pdif box in the audio controls, you do not get "bit perfect" data sent directly out of the computer. Windows XP has something called the kernal mixer which gets in-between and does "bad" things. This can be remedied, it seems, with a few different pieces of software. Since I am tied to iTunes, the best solution for me was downgrade to iTunes 7.0, install multi-plugin 2.5.1 with foobar2000 direct passthrough, and install foobar2000 with the kmixer and apple lossless plugins. now when I play music through iTunes, it runs foobar2000 in the background and plays the music through that, which is bypassing the kernal mixer. Music sounded noticeably better when played this way instead of straight through iTunes.


That still leaves two questions, if anyone is willing to answer:

- Since I am using a shared iTunes library from another computer, when I play music through the share am I getting the full bit/sample rate or is it being changed to 16/44.1 on the fly like itunes does for the airport express?

- Is there software or a device I can use to monitor the s/pdif output from my laptop and "see" what is actually coming out of the computer?




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Hi bpalatt - Welcome to Computer Audiophile.



" ... when I play music through the share am I getting the full bit/sample rate or is it being changed to 16/44.1 on the fly like itunes does for the airport express?"


I'll have to test this as soon as I get a free moment. I'll let you know what I find.


" ... Is there software or a device I can use to monitor the s/pdif output from my laptop and "see" what is actually coming out of the computer?"


Certain DACs like the Bryston BDA-1 and Berkeley Audio Design Alpha DAC have visual displays of the sample rate.


Founder of Audiophile Style | My Audio Systems

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Hi Chris. I very much appreciate you looking into the iTunes thing, as our obviously very busy.


It'd be great to have one of those DACs, but I have a pretty budget system and I use the laptop to watch movies in 5.1 as well, so I don't think I'll be able to use those.


Thanks for the quick responses.



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I don't think the kernel streaming with itunes through foobar is working. Even though I had everything installed, the foobar passthrough turned on, and I was getting sound, when I tried to play just through foobar using KS: Sigmatel audio i got "KS output error: error opening device." I assume if it's not working though foobar by itself, its not working when coming through iTunes. I tried closing all programs and turning off audio codecs and drivers in the device manager, and I still get the same KS output error.


Also, I found a setting on my receiver which displays the the incoming sample rate on screen. As soon as I check the "use s/pdif" box in the advanced volume control the signal comes in at 48khz, and it never changes, regardless of what type of media I am playing and through which player. There seems to be no other way to get sound through s/pdif other than checking that box. My receiver reacts to other input devices with the correct sample rate, so I think this is further proof that not only am I not kernel streaming, but that something (maybe the kmixer) is resampling my audio data on the way out.


Any advice on how to get this working would be great. Thanks.


I did my installs like this:



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  • 1 year later...

I'm in the same situation. I'm not sure, but I think the Sigmatel chip cannot do more than 24/48. And I think kernel streaming isn't supported by the hardware but I'm not sure.


I tried to find the supported hardware x/y combo's that the chip supports but Sigmatel was bought by another semi company and their support site has very little info for any sigmatel branded hardware.


Anyway, for those of us with D620s and integrated Sigmatel audio it would be great if anyone knew what the supported modes were...


In the meantime it seems that the best combo is ASIO4all with DSP Sampling (if your software has that) settings set to use 24/48 if the source is above that value.


Oh - and to the above poster, you might try JRiver - it is great for quality audio.


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