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Old 22nd October 2002, 11:10 AM   #1
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Default Non OS opinions

A new thread, I tried the glove and it fits.

I hope people can give their experiences with non os DAC's, listening experiences with known brand non os DAC's ( Audio Note etc. ), which chips to choose etc. and share them with others.
Also people that don't like non os are welcome to give their opinions and technical explanations.

I like both techniques to be clear.

Please let the thread be respectful to adepts of both techniques.

Jean-Paul
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Old 22nd October 2002, 12:01 PM   #2
Electrons are yellow and more is better!
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My Denon DCD-1000 from 1985 with Burr-Brown PCM56 sounds crap...if you compare to a modern player of rather cheap quality. My DCD-1500 from 1986 with 2x oversampling sounds also crap but with slightly better than DCD-1000. DCD-1000 has a LC filter with 7th order and DCD-1500 has a 9th order LC filter. My point of view is clear. NON-OS are history! Nothing can change my mind. Tubes has a certian character but "old" digital sound is just cold.
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Old 22nd October 2002, 12:05 PM   #3
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I don't know the DCD 1000 but does it have one or two PCM56's ?
I personally never liked the PCM56 and remember replacing them with AD1860 gave a big difference in quality of signal.

Newer non os DAC's don't have the higher order filters ( if they have filtering at all ). Could the way they are filtered not be the reason for the cold sound you describe...

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Old 22nd October 2002, 12:14 PM   #4
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The first CD has only one DAC with sample and hold in order to not create a time differnce. The second one has two DAC's.

Filters are central in digital reproduction. You don't have them because it's fun, it's neccesary. And yes, LC-filters are poor, that's why they came up with oversampling in order to get rid of them!. Nothing beats digital filters, can be made as good as you want. You forget one thing: In Europe it's forebidden to have equipment which generates much RF emission. This is a side effect but not unimportant.
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Old 22nd October 2002, 12:22 PM   #5
Werner is offline Werner  Europe
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How about this then...

Ideally we would record with non-oversampling Flash or SAR ADCs running at 192kHz or more, and we would replay with non-oversampling R2R DACs running at the same rate.

Please think this thoroughly through before attempting to refute this.


About the filters. They are necessary only if you subscribe to the notion that reconstruction filtering is required. Well, it is required if you want to recover the original waveform. But since we are looking at 44.1k sampling (at least), and since we accept that the ear cuts off sharply at 20k or lower, the necessity for reconstruction seems a lot less strict.

Digital filters are not a cure at all. The one function we really need here, Sinc, is a function they can not implement with finite means. The approximation gets better with more oversampling, but please then show me a 20 bit R2R DAC that can keep up with, say, 1024 times oversampling.

See where this leads at?

Oversampling DACs are wrong for one specific reason (no filter approximates Sinc like we want to).

Non-oversampling DACs are wrong for one reason (HF injection into the system), and may be wrong for a second reason (reconstruction required or not).
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Old 22nd October 2002, 12:38 PM   #6
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First of all, the analog output stage is (a major) part of the equation and should be noted. Ears are analog.

TDA1541 + SAA7220 + opamp: dry sound.

TDA1541 + SAA7220 + resistor IV + passive filter + tube output driver: more spatial, lacking bass.

TDA1541 + SAA7220 + tube IV + passive filter + tube output: bass better than before, more linear, more dynamic

TDA1541 + SAA7220 + bjt IV + passive filter + tube output:
best dynamics with a 1541

TDA1541 + SAA7220 + bjt IV + passive filter + emitter follower output:
only 1% less than previous.

TDA 1541 non-os + passive output: slow, muddy, dark.

TDA 1541 non-os + tube output: not so dark anymore, but harsh.

PCM63 non-os + tube output: less harsh than 1541, but not relaxed.

PCM63 + SM5842 + opamp output: details are good, but sounds restless and nervous. Not good for longterm listening.

PCM63 + SM5842 + tube output: more air, better dynamics and detail. Still not so smooth.

PCM63 + PMD100 + tube output: best multibit system. Not as smooth as 1-bit, very up front stage. Tried with PCM56, PCM58, PCM61, PCM 1701, 1702 and 1704 with little differences.

Pioneer LegatoLink + tube output: space, air, smooth and easy to the ear. Details aren't en par with pcm63.

PCM1732 (sigma/delta) + tube output: BB sound (reminds of PCM6x series) with 1 bit smoothness and relaxedness.

TDA1305 (sigma/delta) + tube output: muddy bass, but very spatial.

SAA7325, SAA7350, TDA1547 + tube output: what a mess.

AKM EK4357 + opamp output: Highs a bit harsh, especially in PCM mode. Long warm-up time.

AKM EK4357 + tube output: very balanced, especially nice in the bass with DSD. No harshness anymore.

AKM EK4357 + transistor output: harshness is less, warm-up time remains.

BB DSD1700 + tube output: little lift in the highs, lacking bass. Not too spatial.

I tried a variety of other dacs, but most weren't in my system for very long, so a good judgement can't be given.

Remco
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Old 22nd October 2002, 12:45 PM   #7
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Quote:
Ideally we would record with non-oversampling Flash or SAR ADCs running at 192kHz or more, and we would replay with non-oversampling R2R DACs running at the same rate.
Unrefuted. I'd even go for 2.8 Mhz.
Quote:
since we accept that the ear cuts off sharply at 20k or lower, the necessity for reconstruction seems a lot less strict.
If only it were as simple as that. However, the output signal with all its HF passes through a series of other circuits, often including opamps with 120 dB of gain and 100 dB of feedback, and a (generally observed) higly non-linear element: the loudspeaker.
In all of these circuits intermodulation occurs and even the intermodulation products may intermodulate again in a subsequent circuit.

This accumulation of IMD is like poison in a food chain: the last one gets the most. In this case, the loudspeaker. And the listener.

Remco
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Old 22nd October 2002, 01:03 PM   #8
alvaius is offline alvaius  Canada
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You can not look at the issue on Non-OS DACS with little or no reconstruction filter without looking at everything that is connected to the output of your DAC, i.e. your amp, speakers, and ears!. While your ears can not hears any sounds over 20KHz, if anything in the reproduction chain has enough IM distortion to create artifacts under 20KHz, then you will hear it. That could be your amp, your speakers, and yes, even your ears, though I doubt they would be very effective at creating IM distortion....

Most oversampling DAC implementations use off the shelf oversampling filters. As one other poster has indicated, most off the shelf oversamplers are based on symetric linear phase FIR filters. These exhibit ringing that many consider not to sound good. There is no reason why you could not implement a digital filter that does not ring. You can implement a bessel filter in the digital domain [or you could use a minimum phase FIR filter].

Often when people create digital representations of analog filters, they use a zero order hold approximation [as Werner alludes to]. The zero order hold essentially assumes the signal is the same from the time it was sampled until the time the next sample is taken..... i.e. the typical flat-top digital representation of a signal. A way to improve this approximation, as Werner says, is to increase the sampling frequency. In my old days of designing control systems (one of many hats), we used to say that the zero order hold was only a good approximation if the sample frequency was 10 times the frequencies of interest....

However, higher sampling frequency is not the only way to improve the approximation. You can use higher order holds to create digital approximations of analog filters.

.... sorry if the theory is going over your head [I don't know your background], but this argument of non-os vs. os DACS I find is frought with a lack of understanding of the issues at hand or based on reviews of products that are not ideally implemented.
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Old 22nd October 2002, 01:06 PM   #9
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Quote:
In my old days of designing control systems (one of many hats), we used to say that the zero order hold was only a good approximation if the sample frequency was 10 times the frequencies of interest....
That is the rule of thumb given in my control engineering textbooks ;-)

Quote:
However, higher sampling frequency is not the only way to improve the approximation. You can use higher order holds to create digital approximations of analog filters.
Amen.

Remco
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Old 22nd October 2002, 01:22 PM   #10
Werner is offline Werner  Europe
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For the sake of simplicity I was keeping the IM issue under my hat, trying to keep everything, initially, at a more abstract level. That's also why I do not mention perceived sound quality of these technologies, as implementations differ marketedly. The AN DACs have quite steep filtering beyond 20kHz, the 47Labs machines have nothing at all, or perhaps a first-order.


Zero-order hold: was referring to the output stage of your typical R2R DAC chip (1541, 63, ...). To my knowledge no commercial audio DAC chips exist that implement higher order hold. However, there used to be a slew of Harman/Kardon CD-players that used two DACs per channel to join the dots with generated ramps.


As for the ringing. Read Nyquist. The ringing is required. If you make a digital reconstruction filter that does not ring, it is wrong.

Again, I am talking from a strictly mathematical point of view, striving to recover the original band-limited, then sampled, waveform.
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