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Old 20th August 2006, 07:13 PM   #1
diyAudio Member
 
Join Date: Oct 2004
Question my dac project...?

hi, finally i have an idea for building a DAC:

CS8414 --> AD1852 --> AD797

now my question is, do i need a sample converter before AD1852?because i've seen very often ic's like ad1896 used that way, but still don't understand why are they used and why just sometimes...if someone can explain and maybe give me an opinion...
many thanx!!
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Old 20th August 2006, 07:59 PM   #2
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Location: Brazil
Ciao,

The ASRC is mainly used to *separate clock domains*. It is not a requirement for your dac. When you hear someone talking about *upsampling* they usually mean ASRC.

The ASRC would allow you place a fixed master clock at the dac, and to receive spdif with any sample rate (asynchronously converting it to the sample rate of your choice).

If using the AD1852 + ASRC, you could for example choose 96KHz as your sample rate. Youīd use a master clock of 24.576 MHz (256*96KHz) directly on the clock input of your DAC and also on the mck input of the ASRC. Your converter would always operate at 96KHz regardless of source sample rate.

I cant comment on how it sounds since I havenīt built any dac with ASRC yet.

Regards
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Old 20th August 2006, 08:39 PM   #3
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thanx a lot alexandre, quick and excellent answer...
so it just something that forces the system to work at a determined frequency, right?but if i use a normal audio cd it's unuseful to sample digital data at double frequency, or there is difference?it's like sampling twice the same data, am i wrong?
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Old 20th August 2006, 09:36 PM   #4
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Join Date: Jan 2004
Location: Brazil
Quote:
Originally posted by raikkonen
so it just something that forces the system to work at a determined frequency, right?
You can think like this: it takes an input of arbitrary sample rate and removes or interpolate samples, to match to the output sample rate.

In reality its more complicated than that, because all samples are actually changed. The ASRC will create a lot of in-between values on a 44.1 KHz input, and then will downsample to 96KHz (tipically).

This paper explains it better: http://www.iet.ntnu.no/courses/fe811...0conversion%22

As I said before, I canīt comment on the possible benefits/drawbacks, I havenīt tried it yet.

Regards,
Alexandre
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Old 24th August 2006, 10:20 PM   #5
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Default and the idea changes...

spdif-->pcm2902-->usb-->pcm2707-->i2s-->ad1955-->i/v

that's it!!without the cirrus!!!ha ha!!
but...guys, help, please!!i need a tqfp adapter, but cheap!!!
please if you have an idea, a pcb, anything!!
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Old 25th August 2006, 03:48 PM   #6
Schaef is offline Schaef  United States
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Location: Cuyahoga Falls, OH
So, you're putting a PC between the two PCM chips? You do realize that USB requires a master and a device or slave, don't you? Just hooking the USB connections together will get you nothing, without a master between them, meaning a computer. Unless I'm missing something about the PCM2902, I believe its a USB device, and I know the PCM2707 is also a USB device, neither are a USB master.

Also, it seems like a rather convoluted way of getting there, why did you decide to drop just using a standard SPDIF reciever, namely the Cirrus chip? Your current design will cause ten times the headaches.
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Old 25th August 2006, 05:15 PM   #7
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ops... i didn't know it...even i supposed it...so if i connect usb out and usb in ot the chips i won't have the connection...this is bad...
yes, probably you're right, but doing everything with free samples is a real fun!!and keeps your mind awake!!
let's go buying a cs8414...
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Old 28th August 2006, 02:48 PM   #8
Schaef is offline Schaef  United States
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Glad I spoke up then, I saved you a massive headache. Good luck with the rest of the design! One day soon I hope to be doing one myself.
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