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Thread: Digital that sounds like analog

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    Digital that sounds like analog

    A few days ago I started up a thread on DIYA to talk about some of the progress I'd made on digital systems design with a primary focus on sound quality (rather than measurements). Its here:

    http://www.diyaudio.com/forums/digit...t-numbers.html

    The thread got beseiged by trolls of objectivist persuasion (people who take measurements as primary guarantees of audio quality) and then when it was subequently cleaned up by a mod the overall balance had clearly shifted in favour of the trolls in that some trolling posts were left behind but my rebuttals to those were deleted. Hence I've decided to continue the saga in the more conducive environment over here at WBF.

    For those who aren't so interested in the earlier thread, the idea is to present some digital design principles, but also examples of designs which employ those principles. There will be schematics because DIY is the ultimate way to get a frugal system Nothing beats DIYing in terms of value for money.

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    Site Founder And Administrator Steve Williams's Avatar
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    Hi Richard

    I started to read the thread but bowed out when I saw posts by a former member (fas42) here who always made people's blood boil with his outrageous comments and postulates
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    Yes Frank is an interesting one - he's come to similar conclusions as I have, though in his own way. Over on DIYA he doesn't seem to get up people's noses as much as I do. He was annoying to some at first but seems to have mellowed.

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    OK let's kick off with a fairly simple schematic for a filter. I've already posted this one on the older thread but didn't talk much about it and nobody commented on it at all. Its what's called an 'elliptic' filter - this form of filter has the steepest transition band of any known filter design. A transition band is the area in the jargon between the passband (the signal frequencies you want to keep) and the stopband (the stuff you don't want). Its designed to go on the output of a NOS DAC and starts cutting off around 17kHz. By 22kHz the attenuation is around -50dB. The filter I am hoping is going to improve the rather uneven HF of my NOS DAC (to be presented later) by chopping out the ultrasonic image frequencies that are inherent in NOS designs. As of writing, I've only built one channel and not listened to it so far. I'll post photos some time after both channels are completed.

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    Quote Originally Posted by opus111 View Post
    OK let's kick off with a fairly simple schematic for a filter. I've already posted this one on the older thread but didn't talk much about it and nobody commented on it at all. Its what's called an 'elliptic' filter - this form of filter has the steepest transition band of any known filter design. A transition band is the area in the jargon between the passband (the signal frequencies you want to keep) and the stopband (the stuff you don't want). Its designed to go on the output of a NOS DAC and starts cutting off around 17kHz. By 22kHz the attenuation is around -50dB. The filter I am hoping is going to improve the rather uneven HF of my NOS DAC (to be presented later) by chopping out the ultrasonic image frequencies that are inherent in NOS designs. As of writing, I've only built one channel and not listened to it so far. I'll post photos some time after both channels are completed.

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    This thread for me is likely equivalent to the college student taking Algebra 1...and sitting in on Calculus. But what the hell...;0

    So this is an analogue filter if i am not mistaken...which some designs use in their NOS dacs, yes? And it has to do with rolling off ultrasonic frequencies as you say, i think i got that part. The question i have is...if humans are 'technically' able to hear 20khz and some argue beyond, then why the rollof at 17khz? (which i think is probably my threshhold for all practical purposes anyway based on some half-baked listening/tone tests) Why not start the cutoff at 25khz? Just curious to learn.

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    Don't mean to be too factious but... why bother spending time trying to mimic the analog sound with digital means when you can easily get the real thing?

    Be content with what you have, and you'll have more time for music. I know...it is easier said than done.

    Ki

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    Wow, nice to see you over here Lloyd With your questioning we should be able to make this more generally accessible to others who would like to savour the 'school-room' environment!

    Excellent question to begin with - its 17kHz for a few reasons. Firstly because I can't hear up to anywhere near 20kHz any more. Secondly to make the filter practically possible. Its a requirement to cut off everything above 22kHz and if the transition band were just 2kHz (rather than 5kHz as we have here) the filter would be more than twice as complex. Its already a bit of a struggle winding 7 coils, per channel

    Actually I'm not aware of any commercial DACs using such a filter. Your Zanden's filter is rather a different design because he didn't want the phase shift which goes together with this kind of filter. Me, I'm not so worried about phase - not until I've heard it that is. Have to build it to see for myself.

    We don't start the cutoff at 25kHz because that's too late - the NOS DAC's image frequencies begin at 22.05kHz, half the sample rate.

    Keep the curiosity flowing!

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    Quote Originally Posted by Ki Choi View Post
    Don't mean to be too factious but... why bother spending time trying to mimic the analog sound with digital means when you can easily get the real thing?
    Well I sold all my LPs when I emigrated to China nearly 8 years ago, so the 'real thing' is no longer an option for me. Also I am fairly sure that with digital we can get the real thing much cheaper. If I wasn't I'd not have embarked on this project

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    Site Founder And Administrator Steve Williams's Avatar
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    I too am like Lloyd on this one but simply put can you tell me what it is about your design that makes it sound analog
    Steve Williams
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    That's a really hard question to answer - one I've puzzled over for a fairly long time. I think with NOS DACs what makes them sound analog is they use the lowest possible sample rate and that means the lowest possible glitching. Its glitching from the DAC which 'greys out' the tonal colours and reduces the dynamics. Every time the DAC updates (at 44k1 for redbook) there's a glitch - running oversampled increases the rate of glitch production by the oversampling ratio. I can put what the digital sound lovers call 'detail' into one of my DACs just by changing the filter and I'll show how I do that too in a future post.

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