I've listened to SACD once and it sure sounded good...

I've heard people say that SACD ought to fail in transient reproduction, (because it can only encode a rise and fall relative to the last time step) so I did some calculations:

'Sampling rate' of SACD = 2.8MHz = 64x 44.1kHz

For encoding tones up to 44.1kHz at full scale, it has to reach full-scale from 0 in 64 steps, that is, at any given time there can only be 64 possible positions for the waveform.

This gives it 6 bits resolution to CD's 16 bits??

OK, suppose you only need to encode up to 22.05kHz at full scale, the number of possible positions increases from 64 to 128--7 bits resolution, big improvement

I doubt this is how DSD actually works, but this article http://www.iar-80.com/page40.html (I linked to page 40, but it seems page 1-39 may be going on and on about the sonic flaws of DSD as well) seems to take this view seriously and goes on to talk about how you try to recover musical information from the 6 bit stream.

Ok, um, does DSD assume delta modulation or delta-sigma? It seems to me that the delta argument only applies for delta modulation only.

However, I can't see what's the conceptual difference between delta-sigma modulation and PCM if delta-sigma 'quantizes the signal directly' (http://www.cs.tut.fi/~rosti/1-bit/)

What does 'quantize the delta (difference) between the current signal and the sigma (sum) of the previous difference' mean anyway? (definition of delta-sigma?) It seems to me that on one interpretation of 'sigma', 'sigma' would yield the previous waveform position anyway, just as in delta modulation?

well, some people are apparently having problems with basic maths.
'Sampling rate' of SACD = 1bit * 2.8MHz = 64bits * 44.1kHz
meaning that you get 2^64 different values, compared to 2^16 different values for old cdaudio.

Problem is, the bits are not employed in the same way as CD?

My current understanding is that instead of using 64 bits to encode one time step, DSD uses 1 bit to encode each time step, which is 1/64th the length of a CD time step...

Each time step is simply encoded as higher or lower than the previous.

So as I was saying, if you want to traverse the full scale of the waveform in 1/22050s (corresponding to a maxiumum frequency of 22.05kHz), since you only have 128 time-steps in this time, you are limited to staying in one of 128 STEPS for the whole waveform if you don't want to get slope overload as illustrated here:



This is the kind of problem if you use too fine (i.e. too many steps) to represent the full scale. In the case of SACD DSD (if the bits indeed encode whether it's a step up or down from the last step), if you use more than 128 steps to represent your waveform, you would not be able to keep up with the slope of a full-scale 22.05kHz sine wave.

Having only 128 steps translates into having only 7 bits in PCM. But that would be 7 bits 2.8MHz PCM--so in one dimension it has much higher resolution, in another dimension it has much lower.

However, this would resemble 'delta modulation' as detailed here http://www.cs.tut.fi/~rosti/1-bit/ whereas it is my understanding that DSD resembles a 'raw output' from a delta-SIGMA A/D converter. The Delta-Sigma coding is covered further down the page, and I still don't understand how it works, but it seems that it's supposed to offer higher resolution at the same bitrate.

The usual description given for DSD is more along the lines of delta modulation but I suspect this is misrepresentation or oversimplification to the uneducated public (of which I am a member )

This is an example of Sigma-Delta, or Pulse Density Modulation:



Are you supposed to reconstruct the sine wave by running an appropriate lowpass filter over the modulated signal? Unbelievable

Is DSD actually just PDM code?

The general idea is correct, but SACD quality does NOT resemble the quality of 6 or 7 bits PCM (although it would also be incorrect to say it resembles 64bit PCM).
DSD works by determining for each sampling moment wether the sum of the previous errors is above or below a certain level and encodes that in a 1 or a 0 (at least that's more or less what I understand of it). If one would have a signal in which the signal rises from 0 to full-scale in 1/44100th of a second, then DSD would indeed have a BIG problem with that. Luckily for DSD most music doesn't do that, in fact for most music it would be enough (only just, mind you) to encode the difference between this sample and the previous one in 8 bits (assuming 44100Hz and 16bps).
All in all DSD probably sounds better than PCM (at 44100Hz with 16bps), but it does need a lot of noise-shaping and other tricks to work well (to mask all the little errors incurred by the relatively coarse resolution of the steps), which is one of the reasons why I personally don't like the system. But then again I have never really listened to a SACD.

Also keep in mind that the stepsize for DSD could be different than for 16bps PCM, so also because of that it isn't really possible to compare 1bit in DSD with 1bit in 16bps PCM.

Oh, and for those thinking about the very high sampling-rate of DSD and thus assuming it encodes signals upto 1.4MHz, sorry, but it doesn't, in practice the signal is limited to 50 or 100 Khz (because of some necessary filtering, among other things).

BTW, DSD uses delta-sigma modulation and not delta modulation, delta modulation simply looks at the difference between the current value and the previous one, while delta-sigma modulation looks at the sum of the differences and tries to make that as small as possible.

So the encoding system is really just plain delta modulation? Not delta sigma? Why is that--delta sigma (the article seems to say) has better information density, and can be directly utilized by the final stages of a delta-sigma 1-bit DAC...

On the other hand, if the idea is correct...

You can only get 16 bits equivalent at 43Hz and below!
If you want to encode up to 11.025kHz fullscale sine waves you are still limited to 8 bits resolution!

Surely this is not right??



So does DSD use 1 bit or 8 bits in each time step?

Oh, I get it! You mean in practise it is sufficient to use 8 bits effective resolution and expect there to be no full scale sine waves above 11.025kHz, is that what you mean?

I suppose this could vary from recording to recording, e.g. for hard rock with crazy cymbals you have to go back to 6 or 7 bits

Sorry to have confused you, I didn't have a good look first (I have edited the post now). DSD does use delta-sigma modulation.

As for the thing about using 8 bits, it was just to illustrate that one usually doesn't need all the 16 bits to encode the signal, even when using simply delta modulation (or rather some kind of differential PCM, as the term delta modulation is apparantly only used for differential PCM using a 1 bit quantizer) and a sampling rate of 44100 Hz.

IIRC, the difference between delta and sigma-delta is basically this:

delta: you integrate (sum) the output
sigma-delta: you integrate (sum) the input


The (not too obvious) result is that, while delta modulation can suffer from slope overload (which is what you're describing - the 22kHz sine wave is moving far too quickly for the digital staircase to track it), sigma-delta modulation does not. However, the higher frequencies are full of noise.

Think of it this way: in a basic delta modulator, if you draw a graph of frequency against allowed amplitude, then you are allowed high amplitudes at low frequencies, but only lower amplitudes and higher frequencies (otherwise you push it into slope overload).

However, in a sigma-delta system, it's as if the signal is "equialised" on the input to reduce the higher frequencies (to prevent slope overload), and then re-equalised on the output, to bring the higher frequencies back to their correct level. The result is that the higher frequency ranges contain much more noise - the modulator has an intrinsic noise, and it's amplified at higher frequencies on the output.


This is a very badly remembered way of thinking about it, and it's not quite how it works in reality. But I hope it helps!


So, basically, SACD has about 20 bits of equivalent resolution in the audio band, but terrible amounts of high frequency noise. It also has excellent time resolution. It sounds very good - whether all that high frequency noise is a good idea is a different matter entirely!

Cheers,
David.

yup...


call it what you wan`t...

(i assume your talking hybrid sacd..)

1. this is not real 5.1 audio...
2. not by far (qualety-wise) as good as dvd-a...
3. say you wanna make backups/samplers from sacd..
you can`t ripp...a sacd unit can`t read a cdr/rw/dvdr..

4. conclusion.. just a waste of $$....

I don't think you can fairly say that SACD is "not by far as good as dvd-a".

Both SACD and DVD-A are overspec'ed, albeit less so for SACD. You don't need 24 bits at 192kHz to get a clean representation of an audio signal; or 24 bits at 96kHz for that matter. So there is no reason whatsoever why perfectly mastered SACD and DVD-A versions of the same music should be distinguishable from each other.

Now of course, the problem with ripping is understandable, but hey in that sense DVD-A PCM can't be ripped either.

As for understanding how SACD works, for me what has sufficed is to note that the signal-to-noise ratio of a DSD signal is lower than 0dB---in other words, there is more noise than signal---but that the vast majority of that noise has been shaped into the inaudible frequency range. So within the audible frequency range, you get all the SNR and dynamic range you want.

Some of the technical flaws of SA-CD:

* If you want to do some digital post processing, you must convert
it to PCM. If PDM has any advantage, this advantage is removed.

Digital post processing can be
- digital filters for loudspeaker/room acoustic equalization
- digital filters for splitting the signal for 2/3-way loudspeakers
- digital filters for sound control (more complex equalizers)

* PDM is also not suitable to directly drive digital power amplifiers.
It switches too often so you have to much switching losses.
So PDM must converted to PCM and then to PWM.

* PDM is may be suitable to built low cost head phone DA/C+Amplifiers.

* PDM is very sensitive to asymmetries between switch on and switch off.

* Best possible converters (noise + linearity at low levels) do NOT use PDM, but
- PWM
- 4...16 PDM convertes in parallel
Both can not be generated by PDM, but by a PCM.

*The frequency response of the output filter of a SA-CD is not defined

- So it is not possible to compensate the effect of the output filter in the recording
- It is very likely that manufacturs built gadgets with extremely wide frquency response
and huge amounts of HF noise to boast with a extremely wide frequency response.

A proposal of such a filter should look like:

* 4th order LR-filter with f_{-6dB} = 48 kHz
12 kHz: -0.03 dB
24 kHz: -0.53 dB
48 kHz: -6.02 dB
96 kHz: -24.61 dB
192 kHz: -48.20 kHz
384 kHz: -72.25 kHz

Depending on the high frequency amount the frequency overall-frequency response
can be linearized up to 60 kHz (Pop) or 80 kHz (Classic).

-------------------------------------------------------------
Another point:
I have serious doubts about the need of a higher time resolution.
- Modern perceptial encoder has lime resoution between 1.5 and 5 ms.
- Very critical signal needs time resolution of 2...0.5 ms for f=10 kHz.
- CD has something in the range of 0.2...0.3 ms for f=20 kHz and 0.04 ms for f=10 kHz.
- DVD-A (96 kHz) has something in the range around 0.015...0.02 ms for f=20 kHz and
0.01...0.015 ms for f=10 kHz.

> - Very critical signal needs time resolution of 2...0.5 ms for f=10 kHz.

Is .5ms the finest needed for stereo depth imaging cues ? I heard of a hearing capability called "interaural time difference", which trained humans can resolve to around 30 microseconds. I'm fairly clueless about this (hearing time resolution), so could somebody set me straight about how that parameter is relevant to stereo/binaural music reproduction? The references I found:
Binaural_Hearing and Discrimination of the Ear
seem to only mention impulse noises such as clicks. I can't say that I know of many people who listen to recordings of clicks for pleasure...panning cymbal crashes, maybe?

yup...


a. do a comparison between a "normal/low-fi" cd-player.. that can read a hybrid
sacd.. and a good dvd. then you hear it.. (depends on your ears though..)
on a hybrid sacd.. the content is ordinary red book.. and the
channel info.. lies in a separate dsd layer.. in dvd-a.. the info lies in
the track itself.. ac3/pcm. not vob.. (in a vob file.. there is additional info..)

b. yes it is duable to ripp. 5.1 pcm.. but not a feature soft-developers
put in a history/change.log
(dvd-a pcm tracks is obsolite)

But in this case, it must be compared to the interchannel time difference of the CD, that might be well inferior to the global time resolution of the CD !

It is the time needed to jump from silent to loud.

Inter channel time resolution of the CD is in the pico second range, for the DVD-A
in the femto second range.

This is much much much below the technical limits of the rest of the transmission chain.
Note that an uneven temperature distribution in your listening room generates interchannel
error which are 10^6 times larger. Also if you are moving some millimeters.

It's possible to detect an interaural time delay of ten microseconds at a frequency of 1 kHz. That's about the limit.

CD can store this accurately. It cannot store a waveform with a period of 10 microseconds (that would be a 100kHz sine wave!) - but it can accurately represent a time delay of 10 microseconds between two 1kHz waveforms.


Whether any particular DAC reconstructs the waveforms with enough accuracy to detect this time delay is another matter entirely. The ones I've tried do.


Cheers,
David.


P.S. Frank - I'm well aware of the downsides of SACD. At an AES conference two or three years ago, representatives from Sony would not even enter the same room as my tutor: He was sitting on the "High Resolution Audio committee", discussing many of the issues that you raised. They didn't like it. But format wars are not won or lost on technical matters, or even quality issues.

The idea is to shift most of the quantization noise power (noise shaping) to higher frequencies, combined with high oversampling, and filter them out afterwards. So the final output does not contain a lot of high frequencies.

The equalization of the noise that you describe is IIRC more reminiscent of Dolby noise reduction

I thought the reason to go hi-res is pretty much universally agreed on: to move the Nyquist frequency further from the upper ceiling of human hearing so that lower order filters can be used instead of brick wall filters, which are problematic any way you design them.

OT:
I'd finally figured out what the correct grammar should be for my title. It should be 'Ought SACD (to) sound like crud?'

Should the 'to' be in there?

I don't think brickwall filters are problematic today. On the other side, SACD is problematic in other ways, as explained here.

No, the 'to' should not be in there.

Also (at least in the USA) 'ought' would normally be replaced with 'should' in a question, so this would be better:

Should SACD sound like crud?

Finally, the negative version sounds even better because you are questioning the normal assumption:

Shouldn't SACD sound like crud?

If you really want to use 'ought', you can just put a question mark on the declaritive:

SACD ought to sound like crud?

Well.... they still might be problematic...

Ryohei Kusunoki published article on 1996-97 at Japanese MJ magazine about Non-Oversampling and Digital-Filter-Less DAC Concept.
He states that: "the issue is not either it is Non-Oversampling or Higher-rate-sampling, but the use of the digital filter can cause smearing in the time domain".

Arcticle is available here in english: http://www.sakurasystems.com/articles/Kusunoki.html

Here is interview from 1999 where he tells of his further research on subject: http://www.tnt-audio.com/intervis/kusunoki_e.html

In interview he says:
"I found the answer after listening to a DAC using eight DAC ICs to bring about 8-times oversampling without digital filter. The DAC's sound clearly indicated that oversampling was not the culprit of sound degrading, but the real offender was the digital filter."

"Digital filters cut off signals beyond 20kHz with a very steep curve, but needs around 2msec of time to calculate the enormous data. I think this is the reason of "diffusion of sound coherence", the characteristic tonal quality of the oversampling DAC"


47 Laboratory DAC's Model 4705 Progression and Model 4715 Shigaraki implement Kusunoki's ideas by using zero oversampling and they don't have any filter at all, no digital or analog. Here is info about them:
http://www.sakurasystems.com/products/47dac.html
http://www.sakurasystems.com/products/shigadac.html


There are some other High-End DAC's which follow this "school of thought" by using 0-oversampling and analog filter instead of digital one.
Audio Note Dacs are most famous of them: http://www.audionote.co.uk/dacs/dac_index.htm

On Audio Asylum thread "Non-oversampling DAC concept", Audio Note's Peter Qvortrup says that they started developing the 1xoversampling DAC concept in 1995, first functional prototype was tested in early 1996 and Audio Note UK built and shipped the first 1xoversampling DAC5 in August 1997, a full three months before Mr. Kusunoki's article was published. Thread is here:
http://www.audioasylum.com/scripts/t.pl?f=...digital&m=18753

So in other words, the answer to the thread's question is: sacd is not good

Must be a marketing trick to pass a DRM compliant format to the market.

You know, unfortunately, no matter how much "advantages" the "DVD-A" has, it also comes with some unwanted DRM "features".

So for me, the plain original DVD (not the planned DVD Audio) is enough.

PCM 24/96 is supported in the original DVD spec, and DVD burners are becoming popular. Ppl with 24/96 boards are capable of producing their own HQ discs, with no DRM crap.

Of course the DVD spec also supports dolby digital and mpeg audio layer 2.

All current DVD players can play these. No need for new "DVD-A" aware units either.

Huh? Plain DVD does 24/96? What's DVD-A for then?

Yup. Here are 3 sources for 24/96 music recordings on standard DVDs:

www.chesky.com
www.classicrecs.com
www.hiresmusic.com

Another advantage of these is that the digital data is available on the S/PDIF outputs of many DVD players so you can use high quality DACs (I have a 24/96 MSB DAC II).

DVD-audio goes to 24/192 (with optional lossless packing) and also has better multi-channel support. But, mostly, it has DRM and won't allow the high resolution audio out of the player in digital format.

Any DVD-A records with DRM?

All DVD-A discs are [currently] impossible to rip and force the digital outputs of the players to be downsampled. To me this is generally called DRM, but perhaps there is a specific meaning to DRM that I am not aware of.

Oh, I mean, any DVD-As without audible watermarks?

yup...


hence... dvd-a is just as possible to ripp.
but will.. in most cases be downmixed.. in other words..
it is possible to ripp. a dds track.

ex: the x2cd (ac3/vob/ifo/pcm) i think will downmix it.. i not shure.
there are for shure.. several apps. that >can< do it.

in theory... a multi track sequenser.. should be able to do it..
as long as you go thrue a dds. soundcard.

I don't know if DVD-A discs (or SACD discs for that matter) have audible (or inaudible) watermarks on them. They obviously claimed to want to do this but don't get a lot of favorable press about it. Does anyone know the latest?

The SACD has a sort of physical watermark to prevent playback in an unsecure device (i.e. ripping).

For what it's worth, Audio Note is the most expensive hifi manufacturer in the world.
Their top amplifier (100W vacuum tubes) is more than 150,000 $/€

So would you agree with this kind of design?

What kind of output does it give? The original big- and small- staircase waveform?

e.g.

The one on the LEFT???

Under the picture in the page:
"One could think of just leaving the side band information in the spectrum, because we can't hear them..."

Sorry for stepping is so late .... DVD-A vs. SACD is one of my favourite subjects .... of course i prefer the DVD-A.

In fact, standard DVD-Video specs have a 24/96 mode and it was again David Chesky and his team who were launching the first recordings ( Sara K. amongst them, one of my favourite ) as DVD-V with 24/96 audio ( no video ) more than 5 years ago.

I remember to had the chance to play them on a hi-quality chain on one of the first players with a real 24/96 DAC, a Kenwood, but the results were lousy :-) , especially when being compared to a fully modificated HK or better CD Player !

With some positive thinking one could estimate to hear advantages in some respect, but the overall quality of the CD was much better .. thanks to all the mods done on the CD player, like battery powered power supply, meachanical damping of the drive, etc.

Since then i was waiting for some good mods based on DVD-A players in my former 'scene' , but with no results i hate to say. Many of the freaks were waiting for the first DVD-A to appear, but the stupid format wars between SONY with their crappy SACD thingies and PANASONIC leading the DVD-A side was dooming both formats to being unssuccessful in the end, at least it seems to be this way.

Today PCs are my hobby and i guess i wasnt able to hear the differences between good CD Players and well modified DVD-As, as my hearing isnt trained anymore.

About SACDs inability to playback full scale high frequency signals :

If i am not completely mistaken the same is valid for CDs also, as red book forbids 0 dB 20 KHz signals AFAIK .. else no modern CD player could fulfil red book with a normal 1 bit DAC, even at high internal clock.

I don't understand any word.

DVD-Video:


Theoretical PCM possibilities for DVD-Video:

bit/kHz

16/48 - up to 8 channels
20/48 - up to 6 channels
24/48 - up to 5 channels
16/96 - up to 4 channels
20/96 - up to 3 channels
24/96 - up to 2 channels


These are specs of DVD-V , not DVD-A !
Conclusion: DVD-A is (same with sacd) a kind of copy prohibition.
The industry could provide us with excellent stereo or multichannel music on DVD-V.....

Never mind Frank,

i sometimes dont understand my own shit !

I was convinced ( but havent read red book to investigate, sorry for that unprecise information policy ) that CD red book does not allow to put a full scale 0 dB 20 Khz signal onto a CD when mastering them .... and vice versa that no CD player complying with red book needs to be able to play such a signal. I also estimated that most 1 bit DACs as used in cheap ( and our days even expensive ) CD players wouldnt be able to reconstruct such a signal, given the fact that they had to use a very high internal clock ( 65534 x 44100 = 2.9 GHz ? ) to do that .....

Sorry if the info about red book is not correct or if i didnt understand the basic idea of a 1 bit DAC with noise shaping . My understanding of these DACs was to have a high speed 1 bit DAC feed a capacitor with small portions of current at a high frequency ( about 1 - 2 MHz ?? ) to reconstruct teh original signal, in oder to spare the (expensive ) analog anitaliasing filtering after the DAC and also to overcome linearity problems of the DACs in mass production ?

If this is wrong, how is it ever possible that a 1 bit DAC working at 1 - 2 Mhz can output a full 16 bit signal at 20 Khz ?

OK, so non-oversampling DACs use analog filtering exclusively...

I took a look at this review of an AudioNote DAC 1.1...
http://www.tnt-audio.com/sorgenti/audionot...ote11kit_e.html

And they say it was able to produce this analog filtering:

frequency relative level (dB)
1kHz 0
10kHz +0.25
20kHz -2
30kHz -10
40kHz -17
50kHz -22
100kHz -39
200kHz -57
500kHz -80
1MHz -99

And the phase remains 'essentially linear' up to 20kHz.

Sounds good... the analog filtering should not lead to pre-ringing, unlike digital...

Can somebody tell me once and for all whether pre-ringing is actually an artifact or it is just the natural product of the calculations?

E.g. if you made a 2-way IIR brickwall filter (not 1-way, that would of course give no pre-ringing but wouldn't have correct phase response) that processes the whole track offline, would it still have pre-ringing?

I mean it's funny how with phase correct filtering you inevitably get a symmetrical output from an impulse input... which by definition has added sound content BEFORE the impulse...

.. arent most modern DACs using a combination of digital and analog filtering to overcome antialiasing probs ?

All the 'oversampling' stuff coming up in the early 90is had nothing to do with what oversampling really is ( during recording ), but the attempt to replace (expensive ) analog filtering with ( cheaper ) digital filters, thus making the analog filters more or less redundant .....

Yes, their stuff is very expensive, but they have also very reasonably priced lower range available. Mainly Zero and One series (which has non oversampling 1.1 DAC).



Here is more 0-oversampling DAC's which use analog filtering if you are interested:

Zanden Audio Model 5000
http://www.zandenaudio.com/english/con-5000.html

Morgan Audio Deva Cd-player
http://www.morgan-audio.co.uk/deva.htm

DAC100 HIBARI (which is PCM56 based dac by Kondo)
http://www.audionote.co.jp/digital/index.htm

Other related information:

For Do-It-Yourself people, articles about building 0-oversampling DAC. Very interesting and worth a read.
http://www.tnt-audio.com/clinica/solidstate.html

Audio Asylum discussion thread "Brickwall filter vs No filter or Analog filters"
http://www.audioasylum.com/scripts/t.pl?f=...digital&m=42318

Audio Asylum discussion thread "Building DAC, Filter of Filterless?"
http://www.audioasylum.com/scripts/t.pl?f=...digital&m=24610


Personally i'm interested on 47 Laboratory DAC's. They don't oversample at all and they have not any filtering. Despite absence of digital or analog filtering, reviews of them are positive. Model 4705 Progression is part of my "dream system" which i'm going to buy sometime in far, far future.

47 Laboratory Model 4705 Progression
http://www.sakurasystems.com/products/47dac.html

47 Laboratory Model 4715 Shigaraki DAC
http://www.sakurasystems.com/products/shigadac.html



Yes, this is correct.

What troubles me is that I don't even know what's wrong with oversampling and digital filtering sonically to give non-oversampling an edge in sound quality (if indeed this is true!~ ) I suspect it has something to do with the pre- and post- ringing introduced by digital filters. Do contribute to the new thread I started about this

It is a natural product of the calculation.
It is the calculation itself that is an "artifact" A brickwall filter is an artificial DSP.
Natural (analog) filters are not phase linear...



yes

DVD-D can only uncompresed LPCM up to 6 MBit/s.

96 kHz x 24 bit x 2 (4,6 MB/s)
48 kHz x 24 bit x 5 (5,8 MB/s)
48 kHz x 16 bit x 5.1 (4,6 MB/s)


DVD-A can also compressed LPCM up to 10.2 MBits/s.

192 kHz x 16 bit x 5.1 (4...6 MBit/s)
96 kHz x 24 bit x 5.1 (4...6 MBit/s)

About non-oversampling filters, I have to say that any DAC is supposed to filter as much as possible over half the sampling rate, otherwise you get lots of aliasing above this frequency. This is ultrasonic sound not audible by ear, but if at high levels, may intermodulate (because some equipment is more non-linear above audible range) and cause products that fall into audible range.

So, proper DAC MUST filter over half the sampling frequency (over 22.050 KHz for CD), otherwise the DAC is lacking the necessary reconstruction filter.

About superiority of analog filters over digital filters, etc, any filter response, including analog, can be realized using digital filters, in a more easy and less problematic manner. That means that you can use digital filters at a DAC that show no pre-ringing the same way that an analog filter woud do, but with the consecuence that it won't be linear phase anymore, as analog filters.

And about problems with brickwall filters on CD, time smearing, etc, I think this is not problematic at all for the reasons I explained here: http://www.audio-illumination.org/forums/i...=1&t=2957&st=13

Ryohei Kusunoki says this on the interview:






So, yes digital filters are more practical but CD-player and DAC manufacturers should concentrate on making better quality digital filters.


I again recommend TNT-Audio article about building a 0-oversampling DAC anyone who's interested about the concept. It discusses the theory and technical charactistics of 0-oversampling DAC and mentions Kusonukis DAC filterless concept as "purist":
http://www.tnt-audio.com/clinica/convertus1_e.html

Index of related TNT-audio articles:
http://www.tnt-audio.com/clinica/solidstate.html

I've heard David Chesky's 6 channel 24/96 stuff (that he intended to release on DVD-A) on demo at the AES, and it is amazing. Much better than crappy 5.1 which was never designed for music anyway.


As Frank said, DVD-A allows for lossless compression of the audio data, which enables 6 full bandwidth channels at 24-bit 96kHz. The DVD-V format did not include this lossless compression, so is restricted as described by user, because the maximum data rate allowed on a DVD is about 9-10Mbps (someone will jump in now with the exact figure!). The lossless compression is required to bring the 24/96 6-channel data down below this limit. search: Meridian Lossless Packing, or MLP


Cheers,
David.

P.S. maybe we could say that an oversampling DAC gives "tonal purity" while one without a filter gives " transient accuracy". All the theory of digital sampling goes out the window without a reconstructing filter, but since the ear low passes (typically) somewhere below 20kHz, you could say that is the reconstruction filter.

There is still theory to say that a filter should be included (because of intermodulation in the audio system causing the ultra-sonics to get back into the audio band) - but in addition to the quotes in this thread, I've heard a mastering engineer say that nyquist low pass filtering is a BAD THING when converting higher sample rates to 44.1kHz for CD release.

In reply to Kim_C:

It is true that human ear cannot hear much above 20 KHz, and this is specially true for dynamic situations. Most people can't hear anything over 19 KHz with steady signals, and maybe over 17-18 KHz on dynamic signals.

This doesn't mean that if you let pass lots of high level ultrasonic garbage (that was not in the original analog signal at all!), due to the lack of a reconstruction filter, there is not going to be problems over the audible range. Most amplifiers have a passband that goes up to 100 KHz. If you let happen that ultrasonic garbage to pass from 22 KHz to 100 KHz, there is a good probability that with such high level ultrasonic signals, the usually greater nonlinearity of the amp at such high frequencies causes intermodulation products that fall into the audible range.

Analog reconstruction filters that filter properly this ultrasonics are very difficult to build, and its properties (stability, phase response, passband ripple) are suboptimal in comparison, that's why they were dropped many years ago. On the other hand, using a simpler low order filter that has good characteristics, it is not possible to filter much of this ultrasonic garbage.

Again, I don't see any problem with today's cd oversampiling digital brickwall filters, in theory. In practice, if some phenomena not taken into account happens, well, show me some proper ABX or double blind tests that prove there is really a problem.

In my opinion, Kusunoki is another of those "illuminate" people who sell expensive esoteric solutions to solve unexisting problems. Same happens with things such as upsampling external DACs, SACD, even things like green pens, cd demagnetizers, cable holders, silver cables, etc.

I am aware of this. In this case amplifier needs to be able to cope with this.



No disagreement here, IMHO modern cd-technology is very very good and many ways "good enough" that there is not a need for DVD/SACD audio on customer side, except for the multichannel audio.



Might be, might be not. It's another point of view for sure...

OTOH, i'm not a engineer and don't know much about these things.. i'm just interested on different technologies and how well they work.

Yes, here is comment from Audio Asylum thread "brickwall filter vs no digital or analog filters" which is on same lines:

Where can I find the :insane: smiley from old Hydrogenaudio?