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Argument for expensive digital cables, sounds good but mostly BS?

watchnerd

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The USB 2.0 spec doesn't have a strict limit, but it recommends that implementations achieve a bit error rate of no more than 10^-12. USB3 makes this a requirement. In terms of audio that, works out to roughly one bad bit per week at CD quality or every 30 hours for stereo 192/24. Not a problem.

That's a compelling argument to avoid high resolution.

On a music/hour basis, you're actually getting more frequent errors for no audible benefit.
 

beefkabob

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Me at Guitar Center, with their cheapest XLR M-F cables: "What's the impedance on these?"
Employees: "No clue."
Me: "Well, if they don't work, I'll return them."

They work quite nicely.
 

o2so

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By reclocking to a fixed (stable) clock that is generated inside the DAC box/circuit.

This can be done in several ways using buffering.
This basically is storing incoming data temporarily and then clocking it out with a stable clock.
Tthe analog output of the DAC is not 'real time'. There are many clock cycles in between) or different ways of synchronising.

Only the very first external DACs had poor jitter reduction.
Today most DAC chips or circuits in DACs have very effective de-jitter circuits in there.
There may still be quite a few DACs out there that have poor jitter reduction though.
Thank you. Does this work regardless of the input (USB, coax, toslink)?
 

solderdude

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Depends a bit on how and where the de-jitter takes place in the DAC-device.
It may not be the same/equally effective for different inputs for some DACs.
 
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Eirikur

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Thank you. Does this work regardless of the input (USB, coax, toslink)?
On (asynchronous) USB the clocks are independent; the DAC simply requests to send more or less samples when they drift apart.
For coax and Toslink the base clock is determined by the source and may have jitter. Some of that jitter can still be 'fixed' by the DAC by buffering a few hundred samples and averaging the clock speed.

A few hundred samples is already a lot of information: e.g. a 16bit source produces more than 48bits per stereo sample (SPDIF always uses 24bit/channel + some overhead), so even one hundred samples already give between 4800 and 9600 clock 'ticks' depending on how many 1s it has.
d51ebe30125.gif

(picture used from National Instruments)
 
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Julf

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Some of that jitter can still be 'fixed' by the DAC by buffering a few hundred samples and averaging the clock speed.

One solution is the use of an asynchronous sample rate converter (as incorporated in many modern DACs). This is what solved the jitter and clocking issues in the broadcast industry.
 
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