Hydrogenaudio Forums

CD-R and Audio Hardware => CD Hardware/Software => Topic started by: Joe Bloggs on 09 September, 2002, 07:33:45 AM

Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 09 September, 2002, 07:33:45 AM
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 (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.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 09 September, 2002, 08:33:26 AM
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?
Title: Help! Sacd Good Or Bad?
Post by: Peter on 09 September, 2002, 08:50:08 AM
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.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 09 September, 2002, 09:15:40 AM
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:

)
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 09 September, 2002, 10:16:35 AM
This is an example of Sigma-Delta, or Pulse Density Modulation:



Is DSD actually just PDM code?
Title: Help! Sacd Good Or Bad?
Post by: Jasper on 09 September, 2002, 11:29:13 AM
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.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 09 September, 2002, 11:39:31 AM
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??

Quote
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).


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
Title: Help! Sacd Good Or Bad?
Post by: Jasper on 09 September, 2002, 11:54:26 AM
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.
Title: Help! Sacd Good Or Bad?
Post by: 2Bdecided on 09 September, 2002, 12:49:39 PM
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.
Title: Help! Sacd Good Or Bad?
Post by: n68 on 09 September, 2002, 01:24:08 PM
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 $$....



Title: Help! Sacd Good Or Bad?
Post by: mcg1969 on 09 September, 2002, 06:42:12 PM
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.
Title: Help! Sacd Good Or Bad?
Post by: Frank Klemm on 09 September, 2002, 08:25:19 PM
Quote
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.

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.
Title: Help! Sacd Good Or Bad?
Post by: crazyboy_T on 09 September, 2002, 11:22:09 PM
> - 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 (http://www.sfu.ca/sonic-studio/handbook/Binaural_Hearing.html) and Discrimination of the Ear (http://vaxc.middlesex.ac.uk/~nicholas15/pa2020/lecture1.htm)
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? 
Title: Help! Sacd Good Or Bad?
Post by: n68 on 10 September, 2002, 05:44:33 AM
Quote
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.

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)


Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 10 September, 2002, 06:47:52 AM
Quote
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.

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 !
Title: Help! Sacd Good Or Bad?
Post by: Frank Klemm on 10 September, 2002, 06:57:47 AM
Quote
> - 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 (http://www.sfu.ca/sonic-studio/handbook/Binaural_Hearing.html) and Discrimination of the Ear (http://vaxc.middlesex.ac.uk/~nicholas15/pa2020/lecture1.htm)
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? 

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.
Title: Help! Sacd Good Or Bad?
Post by: 2Bdecided on 10 September, 2002, 07:36:27 AM
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.
Title: Help! Sacd Good Or Bad?
Post by: petracci on 10 September, 2002, 09:53:04 AM
Quote
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.

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
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 10 September, 2002, 10:01:13 AM
Quote
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.


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.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 10 September, 2002, 10:09:34 AM
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?
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 10 September, 2002, 04:27:55 PM
Quote
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.

I don't think brickwall filters are problematic today. On the other side, SACD is problematic in other ways, as explained here.
Title: Help! Sacd Good Or Bad?
Post by: bryant on 10 September, 2002, 05:33:39 PM
Quote
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?

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?
Title: Help! Sacd Good Or Bad?
Post by: Kim_C on 10 September, 2002, 09:51:03 PM
Quote
Quote
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.

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


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 (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 (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/47dac.html)
http://www.sakurasystems.com/products/shigadac.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 (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 (http://www.audioasylum.com/scripts/t.pl?f=digital&m=18753)
Title: Help! Sacd Good Or Bad?
Post by: Artemis3 on 10 September, 2002, 11:19:30 PM
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.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 11 September, 2002, 12:42:14 AM
Huh? Plain DVD does 24/96? What's DVD-A for then?
Title: Help! Sacd Good Or Bad?
Post by: bryant on 11 September, 2002, 12:52:26 AM
Quote
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 (http://www.chesky.com)
www.classicrecs.com (http://www.classicrecs.com)
www.hiresmusic.com (http://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.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 11 September, 2002, 01:45:15 AM
Any DVD-A records with DRM?
Title: Help! Sacd Good Or Bad?
Post by: bryant on 11 September, 2002, 02:12:40 AM
Quote
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.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 11 September, 2002, 02:37:26 AM
Oh, I mean, any DVD-As without audible watermarks?
Title: Help! Sacd Good Or Bad?
Post by: n68 on 11 September, 2002, 03:08:16 AM
Quote
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.

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.



Title: Help! Sacd Good Or Bad?
Post by: bryant on 11 September, 2002, 04:24:20 AM
Quote
Oh, I mean, any DVD-As without audible watermarks?

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).
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 11 September, 2002, 06:46:12 AM
Quote
Audio Note Dacs are most famous of them: http://www.audionote.co.uk/dacs/dac_index.htm (http://www.audionote.co.uk/dacs/dac_index.htm)

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 $/€
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 11 September, 2002, 08:33:38 AM
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.
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 11 September, 2002, 09:26:35 AM
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.
Title: Help! Sacd Good Or Bad?
Post by: Frank Klemm on 11 September, 2002, 09:54:31 AM
Quote
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.
Title: Help! Sacd Good Or Bad?
Post by: user on 11 September, 2002, 09:55:20 AM
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.....
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 11 September, 2002, 10:11:14 AM
Quote
I don't understand any word.

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 ?
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 11 September, 2002, 10:29:18 AM
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 (http://www.tnt-audio.com/sorgenti/audionote11kit_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...
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 11 September, 2002, 10:37:53 AM
.. 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 .....
Title: Help! Sacd Good Or Bad?
Post by: Kim_C on 11 September, 2002, 11:43:54 AM
Quote
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 $/€


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).

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


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 (http://www.zandenaudio.com/english/con-5000.html)

Morgan Audio Deva Cd-player
http://www.morgan-audio.co.uk/deva.htm (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 (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 (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 (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 (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 (http://www.sakurasystems.com/products/47dac.html)

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

Quote
.. 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, this is correct.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 11 September, 2002, 11:51:24 AM
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
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 11 September, 2002, 03:12:52 PM
Quote
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?


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...

Quote
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?


yes
Title: Help! Sacd Good Or Bad?
Post by: Frank Klemm on 11 September, 2002, 03:49:50 PM
Quote
Huh? Plain DVD does 24/96? What's DVD-A for then?

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)
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 11 September, 2002, 07:45:15 PM
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 (http://www.audio-illumination.org/forums/index.php?act=ST&f=1&t=2957&st=13)
Title: Help! Sacd Good Or Bad?
Post by: Kim_C on 12 September, 2002, 08:53:39 AM
Quote
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.


Ryohei Kusunoki says this on the interview (http://www.tnt-audio.com/intervis/kusunoki_e.html):
Quote
KUSUNOKI SAN:
There is a slight possibility that a digital filter-less DAC's intrinsic quantizing noise, existing beyond the audible range, can badly influence the sound. In my experiments, however, the noise is effectively eliminated with a first-order low-pass filter.

The original Compact Disc format was based on the assumption that a "human can hear up to 20kHz" in essence. So why bother oversampling and cutting off the "inaudible sounds" generated by oversampling? I hope my readers to be skeptical on this methodological inconsistency.
So, what is the sampling frequency in essence? Sampling the sound with 44.1kHz means that the CD can "differentiate the sound up to 25 microseconds." Raising the sampling frequency to 96kHz, for example, should not be considered as an extended frequency range up to 48kHz; it should be regarded as an "enhanced precision - over time domain," instead.

TNT-AUDIO:
There are studies showing the human ear sensitivity is extended to frequencies higher than 20kHz, at least in dynamic situations. This seems to contradict your theory. Our ears, anyway, tell us that you cannot be far from being right. What do you think about this?

KUSUNOKI SAN:
My theory is based on the assumption that our audible range is limited to 20kHz, as I have explained in the Audio Amigo interview. Therefore, if we can hear the sound beyond 20kHz and be influenced by it, this would be inconsistent to my theory.

TNT-AUDIO:
On the other side there is an audible loss in high frequencies, a few dBs at around 20KHz. According to you, is the drop in high frequency response an advantage or a shortcoming?
If you consider this a limiting factor, have you ever tried to solve it?

KUSUNOKI SAN:
Certainly there is such loss when you measure the frequency response. However, the loss can be detected only by those people who are very sensitive to high-frequencies, and most listeners cannot differentiate the attenuation of sound. The loss is not favorable, but I think it is not that important.


Quote
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.


Quote
KUSUNOKI SAN:
I have been paying attention to the digital filters these days. I have described my DAC design as a "non-oversampling" in the MJ articles, and the appellation got out of control thereafter. Among those DAC components, the digital filters should be more important -- that's what I think at this moment.


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 (http://www.tnt-audio.com/clinica/convertus1_e.html)

Index of related TNT-audio articles:
http://www.tnt-audio.com/clinica/solidstate.html (http://www.tnt-audio.com/clinica/solidstate.html)
Title: Help! Sacd Good Or Bad?
Post by: 2Bdecided on 12 September, 2002, 09:33:29 AM
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.
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 12 September, 2002, 09:49:58 AM
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.
Title: Help! Sacd Good Or Bad?
Post by: Kim_C on 12 September, 2002, 10:20:05 AM
Quote
This doesn't mean that if you let pass lots of high level ultrasonic garbage, 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.

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

Quote
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.


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.

Quote
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.


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.
Title: Help! Sacd Good Or Bad?
Post by: Kim_C on 12 September, 2002, 10:52:50 AM
Quote
P.S. maybe we could say that an oversampling DAC gives "tonal purity" while one without a filter gives " transient accuracy".


Yes, here is comment from Audio Asylum thread "brickwall filter vs no digital or analog filters" (http://www.audioasylum.com/scripts/t.pl?f=digital&m=42318) which is on same lines:

Quote
1. Brickwall filter
Implemented as a digital filter. Usually used with synchronous oversampling. You cut off all information from the the digital signal after c. half the (over)sampling frequency. Pitfalls: ringing - with a simple single pulse the pulse will start to sound before it is played, because the filter "sees the signal before it is played". Most implementations also ring after the pulse is played. Some call this smearing in the time domain. One could say that this implementation is more correct in the frequency domain.

2. Filterless DAC
You don't cut off (or even attenuate) the alias images predicted by Nyquist theorem by filtering after c. half the sampling frequency. You get alias images right after half the sampling frequency and multiples of that. With CDs these mean ultrasonic noise in the analog signal. If implemented properly, you don't have ringing like with digital brickwall filters, but you do get ultrasonic noise. One could say this implementation is more correct in the time domain.

So it's either frequency or time artifacts. Pick your poison.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 12 September, 2002, 11:20:20 AM
Quote
P.S. maybe we could say that an oversampling DAC gives "tonal purity" while one without a filter gives " transient accuracy".

Where can I find the :insane: smiley from old Hydrogenaudio?
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 12 September, 2002, 11:51:26 AM
Quote
"brickwall filter vs no digital or analog filters" (http://www.audioasylum.com/scripts/t.pl?f=digital&m=42318) which is on same lines:

So it's either frequency or time artifacts. Pick your poison.


Sorry, I don't buy this time artifacts are relevant at all:

1.- they happen on very rare occasions with real music (only if there are high levels and of impulsive type at a very specific frequency around 22 KHz)

2.- they happen also at the AD stage, because the signal is first brickwall-filtered at this stage. At the DA stage, the ringing may not occur at all, depending on how the filtering was done at the AD stage (due to possible filtering at the AD stage of the precise frequency that "excites" the ringing at the DA stage... but also possible that the ringing from both stages adds up, I have to investigate more on this (*1).). Any ringing may appear is very likely to be already recorded on the cd, and cannot be avoided just by not filtering at his stage.

3.-when they happen, are not audible at all due to (*2) its short duration and backwards temporal masking.


So there's a clear pick in my opinion.



(*1) Edit: I've done a quick test with CEP filters, and seems that the ringing doesn't add up.

(*2) Edit: Also, of course, due to the frequency of the pre-ringing (~22 KHz) which falls out of the audible band.

Edit: this nasty quotes break whenever you edit the message  (Garf: this better? KikeG: Sure  )
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 16 September, 2002, 07:41:14 AM
Quote
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.

The guy from TNT audio interviewing this japanese says it, and you should listen closely :

That is not said at all !!

There haven been a number of publications in the past 10 years about the capability of the Human ear to 'hear' sound > 20 KHz ( dont ask me for any links, i dont have any avaialble  )

The trick is to differentiate between

- static ( sine )

and

- dynamic ( transient, impulses )

'hearing' .

I did tests with my own ears, almost 5 years ago :

- I was not able to 'hear' sine tones > 16.5 KHz ( maybe even lower now )

but

-  in a blind ABX test i could easily detect a low pass filter of 4th order ( 24 db/octave , linkwitz ) with a -3 dB at 20 KHz on a high quality Stereo chain, with normal music ( hitrate : > 85 % ) .

Why ?

There are theories taking into account that we dont have only one ear, but two of them. Human brains can localize sound sources very precisely, especially if the sound is transient, means no sine tones. Dont ask me how its done exactly, if i remember correctly its a kind of measuring the time difference between the signal arriving on the one ear and the other time short after ( a few µs resolution ).

This theory, if i remember correctly, now states that this time resolution works better the higher the frequency of the transient is. A common example will be that its easier to localize the sound produced by a metal hammer hitting metal, compared to a soft drumstick hitting a drum.

Means you cant really her 'sound' or 'tones' > 20 KHz, but the missing/existence of sound > 20 Khz can deteriorate sound localisation. BTW, its not said that sound > 20 Khz cant be detected at all because of the inner ear. We have a small membrane in our inner ear ( dont ask me for its name ) being positioned before cochlea, and its function is still not 100% clear to science.
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 16 September, 2002, 08:03:24 AM
I would also think that you're most likely to hear ultrasound during transients if you can hear ultrasound at all.
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 16 September, 2002, 01:56:27 PM
Quote
There haven been a number of publications in the past 10 years about the capability of the Human ear to 'hear' sound > 20 KHz ( dont ask me for any links, i dont have any avaialble   )

I only know of one study done by japanese researchers, where they "measured" different brain waves when the test subject was exposed to high ultrasonic levels. However, the subject didnt hear anything, and the different brain waves were present even after the utrasonic sound ceased. No one was able to repeat this experiment, iirc.

If you know of another REAL study about this, it would be interesting to know about it. But not of the type "I've heard" or "I've read somewhere" myths.

Quote
did tests with my own ears, almost 5 years ago :
- I was not able to 'hear' sine tones > 16.5 KHz ( maybe even lower now )
but
- in a blind ABX test i could easily detect a low pass filter of 4th order ( 24 db/octave , linkwitz ) with a -3 dB at 20 KHz on a high quality Stereo chain, with normal music ( hitrate : > 85 % ) .
Why ?


Maybe your filter with -3 dB at 20 KHz was something like -1 dB at 13 KHz, I don't know. This would be quite audible with harmonic-type musical signals.

The thing is that I can hear up to 18 - 18.5 KHz with sine signals, but can't differentiate an impulsive clip such as castanets attack brickwall-filter lowpassed at 15 KHz from non-lowpassed. Even have some difficulties with a 12 KHz lowpass. Just try at http://www.pcabx.com/technical/low_pass/index.htm (http://www.pcabx.com/technical/low_pass/index.htm). Just try to ABX any of the 18 KHz lowpassed clips, and tell me if you are able to hear them sound different. Go down into the lowpass until you can hear the difference.

This is because with this more complex real world signals, masking comes into effect, and the lower high frequencies easily mask the higher high frequencies. This doesn't happen with pure tones (sines).

About localization, what happens is that ear is very sensitive to interaural delays, but if you lowpass both channels the same way, the interaural delays remain the same. Even when ear is so sensitive to interaural delays, this doesn't has much to do with the upper frequency limit of hearing.
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 16 September, 2002, 06:25:49 PM
Quote
-  in a blind ABX test i could easily detect a low pass filter of 4th order ( 24 db/octave , linkwitz ) with a -3 dB at 20 KHz on a high quality Stereo chain, with normal music ( hitrate : > 85 % ) .

What kind of filter was it ? Analog or digital ?

How much ABX trials were there ?
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 17 September, 2002, 05:39:47 AM
it was an analog filter i had done using a few op-amps. The interesting thing was that the filter was always in the signal path, but only cut-off frequency was changed from 20 to 40 Khz with a normal switch.

I made my brother switch to any state without telling me, and did 3 runs alltogether, each being 20 repetitions.

A 24 db linkwith filter shouldnt have a 1 db attenuation at 13 KHz.

About the ABX test : i dont have any suitable stereo system connected to my PC for the time being ( and no plans to do so ), so i cant participate in any tests right now. I just moved to another house recently, with a much bigger living room, and my 'good' stereo system will hopefully soon have a 2nd spring. Now the only question is how i could connect a decent 24/96 soundcard ( which i also dont have for the time being ) to this system ?

To make this clear ( i maybe wasnt before ) :

The comparison test mentioned above was done with analog material coming from a Linn LP 12, Karma pickup, amplified with Muscial Fidelity Preamp, and playing direct cut material.

Of course, this test was useless with CD material, as it doesnt contain anything > 20 Khz . The only sensible way to make such a test on a PC would be with a decent 24/96 soundcard and music recorded at this very sampling rate.

You cant hear what i decribed with test signals ! The recording was also very important IMHO, as most studios destroy the original stereo information by recording every instrument with a dedicated microfone and giving its 'position' in the stereo image using the 'balance' slider .... so there is no runtime difference between left and right ear, but simply an amplitude difference ...
Title: Help! Sacd Good Or Bad?
Post by: fewtch on 17 September, 2002, 05:49:50 AM
Sheesh... all of this started from an original question, "is SACD good or bad?" (LOL).  Well, why not just arrange to listen to one (with a good recording and proper equipment) and decide that way?  End of thread. 
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 17 September, 2002, 06:32:02 AM
Quote
giving its 'position' in the stereo image using the 'balance' slider .... so there is no runtime difference between left and right ear, but simply an amplitude difference ...

You can't say "most" studios use this. Or just give an example (well, ok, this is easy since fewtch just posted one yesterday  )
Among my 200 CDs of mostly electronic or pop music, I think I can find two or three recorded in mono, one or two in pure stereo, and maybe one or two with a "balance slider stereo" as you describe.
Most of them use an artificial DSP stereo with delay, spatialization, reverb and all.
Title: Help! Sacd Good Or Bad?
Post by: fewtch on 17 September, 2002, 07:12:40 AM
bleep
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 17 September, 2002, 07:32:13 AM
Quote
A 24 db linkwith filter shouldnt have a 1 db attenuation at 13 KHz.

You're right, I didn't realize it was a 24 dB/octave filter when I replied.

However, other things could have affected your test. Non perfect filter response, for example (higher or lower gain on the passband depending on the cutoff, big phase aberrations, etc.), because I don't believe a 20 KHz lowpass can be easily audible at all.

Try when you can the test I propose, the results are quite convincing. If you can't hear a 19 KHz tone, and you can't hear a 16 KHz lowpass on a signal with many transients (castanets), it does not seem logic that you could hear a 20 KHz lowpass on that same signal.

You can try on more critical signals, possibly such as some cymbals (maybe castanets has too much low frequency impulsive content) , I believe Pio has some available, lowpass them with something like 18 KHz an see if you can tell the difference.

Also, 85% hitrate doesn't mean much by itself, it would be necessary to know the number of trials to see how significan is that number.

In the same website (pcabx) there was a test similar to yours, but comparing 24/96 musical signals with 16/44.1 signals, unfortunately, it seems to be offline now. As I know, nobody had been able to identify those different formats in an ABX test.
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 17 September, 2002, 08:31:59 AM
Quote
In the same website (pcabx) there was a test similar to yours, but comparing 24/96 musical signals with 16/44.1 signals, unfortunately, it seems to be offline now. As I know, nobody had been able to identify those different formats in an ABX test.

What kind of 'test signals' were used here ?

If it was a kind of single instrument i dont think you could 'hear' what i could differentiate in my comparison, being the 'air' around instruments, the precision of positioning and the 'depth' of the recording studio .

Chesky recordings, again, are very accurate in stereo positioning and delays ( maybe they use a good old stereo microphone assembly, like a Jecklin configuration, at least for some recordings like Ana Caram )
Title: Help! Sacd Good Or Bad?
Post by: Joe Bloggs on 17 September, 2002, 11:01:35 AM
He said analog filter. The most likely explanation for him ABXing the lowpass is phase distortion. QED
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 17 September, 2002, 12:17:38 PM
Quote
If it was a kind of single instrument i dont think you could 'hear' what i could differentiate in my comparison, being the 'air' around instruments, the precision of positioning and the 'depth' of the recording studio .

The 'air' around instruments has not much to do with the upper high frequency range, is usually more a thing of middle and upper-middle frequencies.

Try dowloading the "rach_original.flac" file from http://www.ff123.net/samples.html (http://www.ff123.net/samples.html)

It is a quality, 'airy' piece of classical music from Telarc records. If you analyze it, you'll see that it has practically no content above 10 KHz.

About your test, vinyl can go up to 50 KHz in some cases, but there is just garbage at there. I beleiev max. "usable" frequency is below max. usable frequency at cd (which goes up to near 22 KHz).

If it was so easy to detect a 20 KHz lowpass, why is it so universally agreed between audio engineers and researchers that our hearing only goes up to that frequency, in best cases? Why are there no experiments *clearly* proving that we can "feel" (=hear) things above 20 KHz?
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 17 September, 2002, 05:40:27 PM
Quote
You can try on more critical signals, possibly such as some cymbals (maybe castanets has too much low frequency impulsive content) , I believe Pio has some available, lowpass them with something like 18 KHz an see if you can tell the difference.

Back online :

(https://hydrogenaud.io/imgcache.php?id=378f06fb1e9a21a45e2c555eb033df9e" rel="cached" data-warn="External image, click to view at original size" data-url="http://pageperso.aol.fr/lyonpio2001/test/lowpass/cosmicbaby.jpg)

Right-click and save target as cosmic.pac (524 ko) (http://pageperso.aol.fr/lyonpio2001/test/lowpass/cosmic.pac)

(https://hydrogenaud.io/imgcache.php?id=dec859e980af8fdb70aa688b1238d0d2" rel="cached" data-warn="External image, click to view at original size" data-url="http://pageperso.aol.fr/lyonpio2001/test/lowpass/transwave.jpg)

Right-click and save target as transwave.pac (991 ko) (http://pageperso.aol.fr/lyonpio2001/test/lowpass/transwave.pac)

Air liquide.pac was impressive in the 14 kHz range, it must mask higher frequencies. I didn't bring it back.

Nobody tried these samples yet. I don't know if they are good test samples for lowpass, but at least they seem to.
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 19 September, 2002, 08:56:42 AM
Quote
It is a quality, 'airy' piece of classical music from Telarc records. If you analyze it, you'll see that it has practically no content above 10 KHz.

KikeG,

my knowledge about audio engineering can certainly not compare to yours, but if you are telling me that it is impossible for humans to 'take audible information' ( maybe better then using the expression 'to hear' ) from the frequency range > 20 Khz by looking at a track with a spectrum analyzer, than it seems i failed to communicate the idea behind my original statement, being why frequency components in 'sound transients' maybe could be helpful for human hearing 'system' ( thats is including brains ) to support sound localisation, and thus to improve 'spatial' impression of music ....
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 19 September, 2002, 05:18:38 PM
Well, I was just intending to explain that this "airy" sensation between musical instruments, as far as I know, has not much to do with the upper frequency range of hearing.

About looking a track with a spectrum analyzer... well, sound at last is only a signal, some of whose properties can be easily analyzed. If a classical musical piece has no content over 10 KHz, you can bet it won't have over 20 KHz.

About frecuencies over 20 KHz... well, if we are *totally* deaf, unsensitive,  to a ultra-high level signal of 21 KHz, I can't see how, much lower levels that can be found in regular music at these frequencies, could have an effect over our perception.
Title: Help! Sacd Good Or Bad?
Post by: DickD on 11 February, 2003, 08:41:46 AM
Regarding the airy sensation, I concur with KikeG.
I've also created it while trying ways to add ambience to a dry mono vocal recording at 11.025 kHz sampling rate, 16-bit/channel without comb-filtering or colouration of the sound.

This sampling rate allows only frequencies of 5.5 kHz and below, and I've resampled properly to 44.1 kHz (bandwidth limited) after adding the effect to guarantee no aliasing, and the effect is still there (not an artifact of my soundcard operating at low sample rates where it's not anti-aliased.

It's a kind of airy quality - a gentle whisper and feeling of space around the sound. It feels slightly grainy and analogue, and much more like being in an auditorium - I guess that's a sort of "presence".

That wasn't audible in the original sound, yet all frequencies above 5.5 kHz have been filtered out, so despite the hissy nature of the "grain" or "air" that suggests a wide frequency response, it's not actually a wide response at all, it's somehow in the ambient relfections I aritificially added and can be represented within the frequencies below 5.5 kHz.

Incidentally, I have some other ideas from this exercise - including the use of complex numbers (Argand diagrams) to represent stereo waveforms, and how this might allow an alternative, safer method of creating monaural recordings from stereo sound. If I get time to work this through, I'll share it.

In the mean time, I'll open up a new thread to describe the ambience effect.

Dick Darlington
Title: Help! Sacd Good Or Bad?
Post by: robUx4 on 03 March, 2003, 09:00:51 AM
Quote
About frecuencies over 20 KHz... well, if we are *totally* deaf, unsensitive,  to a ultra-high level signal of 21 KHz, I can't see how, much lower levels that can be found in regular music at these frequencies, could have an effect over our perception.

Here is my little theory about this. The human animal evolved from others animal that used to live in water. And in water the sound you can "hear" is much higher in the frequency range. The same way as our brain have a reptilian part (this is the technical name), maybe our body have some reminiscents of our origins.

Now about how we could hear transients. If you take a constant sound at 15kHz that you can hear. The internal parts of your hear will move with approximately the same movement/speed. We basically can't "hear" much higher because of the mechanical inertia of the moving parts in our ears... But if you have a short click (a few milliseconds) at 30 kHz. There will be the moment, when the sound starts, that your ear will move. But it won't be able to follow the speed. That doesn't mean your ear will not move. Especially if the frequency is not 100% constant (it will produce modulation at low frequencies). The movement of the ear, might not be at all like the movement of the air we "receive", but in the end there is one. And I think we can feel it.

Maybe that's all bullshit, but I think it's unfair to say that we cannot feel at much higher frequencies. I believe Christian was fair with his tests and that it somehow proves that the frequency range is more complex that what we currently know. (and I don't say that because Christian is my working partner )
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 03 March, 2003, 09:35:34 AM
Quote
We basically can't "hear" much higher because of the mechanical inertia of the moving parts in our ears... But if you have a short click (a few milliseconds) at 30 kHz. There will be the moment, when the sound starts, that your ear will move. But it won't be able to follow the speed. That doesn't mean your ear will not move. Especially if the frequency is not 100% constant (it will produce modulation at low frequencies). The movement of the ear, might not be at all like the movement of the air we "receive", but in the end there is one. And I think we can feel it.

Yes, my prof in uni always was trying to make clear to us that there is a HUGE difference between the hadnling of the ear for static ( sine ) and dynamic ( transients ) signals.

BTW, high frequency hearing tests with earphones will fail completely to be able to find the effects i am proposing, as most test signals are mono, so there is no form of time delay between L and R ....
Title: Help! Sacd Good Or Bad?
Post by: 2Bdecided on 03 March, 2003, 10:02:00 AM
Most of the ideas that are being kicked around here could be proved or disproved in a well planned experiment.

And some people here have the equipment to try such an experiment.

1. 2496 sound card
2. decent headphones
3. Cool Edit


We've been discussing these things for ages. When these ideas were first kicked around in the 1990s, no "normal" people had the equipment to be able to test the ideas. Now, many people do. It seems to me, rather than discussing hypotheticals, we could also come up with some ways of proving/disproving the hypothesis, and then carrying out the experiment.


As my own contribution, the 1st hypothesis seems to be:

24/96 material sounds better than 44.1/16

The approach should be
a) get 24/96 material (David Chesky sells some nice DVDs)
B) convert to 44.1/16 (Cool Edit will happily do this as well as anything)
c) compare (blind test)

You'll have to remove all confounding problems - for example, I can hear when my audiophile24/96 switches sample rate, which makes a blind ABX test rather difficult! Putting some silence at the start of tracks (and not switching mid-track) may remove this problem.


If, and only if, someone can hear a difference, then we can start trying to figure out what mechanism is causing that difference.

Any takers?

Cheers,
David.
P.S. - When I had decent headphones, I didn't have the 24/96 sound card - now I have the card, I no longer have the headphones. However, I'm working on it! I've never had a quiet listening environment.
Title: Help! Sacd Good Or Bad?
Post by: user on 03 March, 2003, 10:04:56 AM
http://www.hfm-detmold.de/texts/de/hfm/eti...ten/seite2.html (http://www.hfm-detmold.de/texts/de/hfm/eti/archiv/diplomarbeiten/seite2.html)


"..............dass durchschnittlich Frequenzen ab 16-17 kHz nicht mehr wahrgenommen werden. In seltenen Fällen können Personen 20 kHz und darüber detektieren. Diese Angaben beziehen sich aber nur auf die Eigenschaften des Gehörs bei periodischen Stimuli. Da Musik und die meisten Schallereignisse aber nicht nur aus periodischen Vorgängen bestehen, sondern durch unperiodische Einschwingvorgänge charakterisiert werden, muss das Gehör auch auf die Aufnahme und Verarbeitung solcher unperiodischer Signale hin überprüft werden.

Untersuchungen, die speziell mit unperiodischen Signalen und steilen Impulsen als Reiz arbeiten, sind vergleichsweise jung und werfen noch viele Fragen auf. Impulshafte Signale, wie etwa ein Trommelschlag, können in ihrem scharfen Einschwingvorgang sehr hohe Frequenzen über 20 kHz im Frequenzbereich enthalten. Es ist also durchaus denkbar, dass während geräuschhafter Einschwingvorgänge der auditiven Wahrnehmung die Möglichkeit gegeben ist, indirekt auch Frequenzen über 20 kHz zu verarbeiten und zu einer bewussten klanglichen Unterscheidung zu führen.........."


Short translation of this part of a Diploma work:

average listeners may listen up to 16-17 kHz, not more !
Some persons up to 20kHz and a little bit more.
This refers to abilities of "ears/brain" regarding periodic stimulation.
As music does not always consist of periodic stimulation, but contains aperiodic "Einschwingvorgänge = first-time-oscillation", the hearing has to be tested to aperiodic signals.

Tests, surveys regarding aperiodics, are young and result to new questions.
Impulse signals like drum, have a sharp "first-time-oscillation" (aperiodic) , and may contain much higher frequencies than 20 kHz.
It may be possible, that "listening" of frequencies higher than 20 kHz may be possible, during these "aperiodic first-time-oscillations".

So, they tested the high-sampling frequency of 96 kHz vs. 48 kHZ with practical music, with the result, that 48 khZ sampling is enough, indishtinguable from original sound.
Test-listeners were a good group of students becoming recording-ingeneers.
Title: Help! Sacd Good Or Bad?
Post by: ChristianHJW on 03 March, 2003, 10:28:29 AM
Quote
So, they tested the high-sampling frequency of 96 kHz vs. 48 kHZ with practical music, with the result, that 48 khZ sampling is enough, indishtinguable from original sound.
Test-listeners were a good group of students becoming recording-ingeneers.

They were using a tweeter fast  enough to reproduce transients containing high frequencies, like a Manger or an airmotion transformer, in the listening tests ?

Note that a normal studio monitor is just not good enough for these kind of experiments .... good earphones, like some nice STAXes, should be fine .....
Title: Help! Sacd Good Or Bad?
Post by: Frank Klemm on 03 March, 2003, 01:25:04 PM
Quote
Most of the ideas that are being kicked around here could be proved or disproved in a well planned experiment.

And some people here have the equipment to try such an experiment.

1. 2496 sound card
2. decent headphones
3. Cool Edit


We've been discussing these things for ages. When these ideas were first kicked around in the 1990s, no "normal" people had the equipment to be able to test the ideas. Now, many people do. It seems to me, rather than discussing hypotheticals, we could also come up with some ways of proving/disproving the hypothesis, and then carrying out the experiment.


As my own contribution, the 1st hypothesis seems to be:

24/96 material sounds better than 44.1/16

The approach should be
a) get 24/96 material (David Chesky sells some nice DVDs)
B) convert to 44.1/16 (Cool Edit will happily do this as well as anything)
c) compare (blind test)

You'll have to remove all confounding problems - for example, I can hear when my audiophile24/96 switches sample rate, which makes a blind ABX test rather difficult! Putting some silence at the start of tracks (and not switching mid-track) may remove this problem.


If, and only if, someone can hear a difference, then we can start trying to figure out what mechanism is causing that difference.

Any takers?

Cheers,
David.
P.S. - When I had decent headphones, I didn't have the 24/96 sound card - now I have the card, I no longer have the headphones. However, I'm working on it! I've never had a quiet listening environment.

Wrong.

You must compare 96/24 and 96/24->44.1/16->96/24 audio.

96 kHz and 44.1 kHz often sounds different (with problems on the 96 kHz
side), because the frequency response is very different.

1st series of tests:  Compare 96/24 with 96/23...96/10.
2nd series of tests: Compare 96/24 with 96/24->64/24->96/24, 96/24->48/24->96/24, 96/24->44.1/24->96/24, 96/24->40/24->96/24, 96/24->32/24->96/24.
Title: Help! Sacd Good Or Bad?
Post by: robUx4 on 03 March, 2003, 06:09:55 PM
That would be a very interresting stuff. How many people here have the necessary hardware and willing to make that test ?
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 03 March, 2003, 07:06:58 PM
I've got a 24/96 soundcard, and HD 600 Headphones.

30 kHz at -3 db
38 kHz at -10 db

But I've got no 96 kHz material.

Robux4, what you describe about the transient response is perfectly right. But it doesn't support your point. A delayed slow response to a 30 kHz transient is nothing else than a clean lowpassed version of the transient. The pulse has a flat frequency content  from 0 to 30 kHz. The delayed version has a flat frequency content from 0 to 16 kHz only. The removal of the inaudible frequencies acts exactly like inertia.
See a more detailed explanation at http://forum.cdfreaks.com/showthread.php?s...1864#post376461 (http://forum.cdfreaks.com/showthread.php?s=&threadid=61864#post376461)

However, the "periodic" behaviour against "transient" one is a valid question. In mathematical terms, we ask if the ear is a "linear" device. Linear meaning roughly that the effect of the sum equals the sum of the effects, and therefore a pulse being a sum of frequencies from 0 to N, if the ear is linear, it should react as a lowpass filter, the resulting lowpassed pulse actually heard being the "sum of the effects", that is, the sum of all reactions to audible content in the pulse.
But if the ear is not linear, it won't necessarily lowpass the pulse. The pulse heard won't necessarily be the sum of the effects that each frequency component in the pulse would have.

The most basic test is the intermodulation one. Say that you can't hear above 16 kHz. Generate a 6 kHz square wave in cooledit. And for god's sake, perform a spectrum analysis of it... I tried 3 times to generate square waves in Cool Edit before actually getting a true one. A true square of frequency N must have a frequency content of N, 3N, 5N, 7N ect... and nothing else ! The square generator in CoolEdit is completely wrecked. But it works at 6 kHz for a 48 kHz sampling rate.

Then, you have a 6 kHz sine, plus a 18 kHz one. If you compare it with a 6 kHz sine alone, all you'll hear is a volume difference (the square being louder because the level of the N frequency is above the peak level of the square).
Rising the volume, a ghost 12 kHz tone should appear. It's intermodulaton distortion in the ampli or speakers.
Now, playing a 6 kHz sine in one speaker and a 18 kHz one in the other, there is no intermodulation, no 12 kHz tone in the room.
WARNING : it is said that 18 kHz sines can fry tweeters !!!
Check with a microphone, that your speaker actually plays a 18 kHz tone.
If you hear a 12 kHz tone anyway, check with a microphone, that it is not in the room, and if it's not, you're hearing the effect of inaudible frequencies.
I've tried it, and could hear nothing at a level that would have been 100 db if it has been a CD playback. So tell me about 40 kHz @40 db !
I can't find the report of the other people that have tried the same experiment and failed too, but Nika says it's in there :
George, Watch this!!!....(96k) (http://www.musicgearnetwork.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=3;t=000822) @George Massenburg, abstract in page 33

Quote
Actually, according to Paul Frindle........no. According to experiments that were described in pages past, this phenomenon does not happen with the ear if either of the two fundamental tones is higher than the ear can hear. I have actually substantiated this myself as well. My hearing caps off at around 17.5kHz. I played a 16.5kHz signal and an 18.5kHz signal through two separate pair of speakers in a room and listened for the 2kHz tone and it never shows up. Therefore, limiting the signal to above what humans can hear does not inhibit our ability to capture the performance as naturally as we would hear it live.


And last, there is Oohashi's experiment ( http://jn.physiology.org/cgi/content/full/83/6/3548 (http://jn.physiology.org/cgi/content/full/83/6/3548) ) in which people say they couldn't hear the difference, but the electroencephalogram showed one (delayed). They say also that people said they were more comfortable when the ultrasonic content was played, but they don't say how many people said it. This should be analysed like an ABX result : if you ask people to choose which performance they prefer, and that a little more than half the people tell they prefer the ultrasonic one, it can very well be pure chance. They don't state the level of confidence of the correlation between people's statement and the performance played (=same kind as the level of confidence between your ABX choice and the samples really played).

I've just thought of a non linear behaviour of the ear... The ear, as a lowpass filter, should act like an IIR filter isn't it ? But a true linear behaviour would be the one of an FIR (well, a symmetric IIR one, with pre echo).
Wouldn't there be something here to discuss ?

IIR/FIR filters explanations :
The audio engineer's approach to understanding digital filters. For the idiot such as myself (http://www.cadenzarecording.com/papers/Filters.pdf) by Nika Aldrich
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 04 March, 2003, 03:46:40 AM
I agree pretty much with Pio's explanations and Frank proposals.

However, about the first, I think you don't have to confuse linear response with impulse/causal response. IIR filters, which are causal, can perfectly be linear in its behaviour. Linear refers usually to amplitude linearity, whilst the difference between IIR and FIR is mostly on the phase/time response.

I think that our ear, as a mechanical semi-linear minimum phase (IIR-style) device, won't differ much in its response with transients or with continuous signals, but I don't know enough about this to talk with authority. That's why I set up the lowpass audibility blind tests with impulsive high-frecuency contents inside a musical signal: http://www.hydrogenaudio.org/forums/index....4894&hl=lowpass (http://www.hydrogenaudio.org/forums/index.php?act=ST&f=1&t=4894&hl=lowpass)

About Frank proposal, I think that as a first test, 96/24 compared to 96/24 -> 44.1/16 -> 96/24 would be enough.
Title: Help! Sacd Good Or Bad?
Post by: KikeG on 04 March, 2003, 03:50:30 AM
Quote
... but I think it's unfair to say that we cannot feel at much higher frequencies. I believe Christian was fair with his tests and that it somehow proves that the frequency range is more complex that what we currently know.

That's why I set up a similar test, but with a digital lowpass and critical musical signals, at the link at my previous post. Only one person I know of has proved to hear the 20 KHz lowpass, none a 21 KHz lowpass.
Title: Help! Sacd Good Or Bad?
Post by: ExUser on 04 March, 2003, 04:15:58 AM
This is completely off-topic, but I just wanted to say that I always feel enlightened when the major names around HA come out for a high-signal-to-noise discussion. This is one of those times. Thanks, guys.
Title: Help! Sacd Good Or Bad?
Post by: robUx4 on 04 March, 2003, 04:41:51 AM
I just wanted to add something to my paleonthological (?)/biological theory. We are mostly made of water, which conduct ultrasonic waves very well. So there is a chance that we would be responsive to these even though it doesn't come direcly from the ear. The same way as we "feel" a low bass sound instead of hearing it.

So that would mean a headphone is probably not enough to conduct this test.
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 04 March, 2003, 06:29:47 AM
Quote
The human animal evolved from others animal that used to live in water. And in water the sound you can "hear" is much higher in the frequency range. The same way as our brain have a reptilian part (this is the technical name), maybe our body have some reminiscents of our origins.


Easily cheched : just to compare the hearing ability of a fish with the human one. But whatever result is got, it won't make human hear above 20 kHz...

But you're right about ultrasounds in the body. According to this paper (http://www.nottingham.ac.uk/physics/ugrad/courses/mod_home/f33ab5/notes/us/US001intro198.doc), the human body conducts ultrasounds better than air.
Title: Help! Sacd Good Or Bad?
Post by: user on 24 March, 2006, 09:26:21 AM
interesting study and paper about difference between DSD & PCM 24 bit/176,4 kHz carried out scientifically with abx/double blind,

here the short English paper pdf :
http://www.hfm-detmold.de/eti/projekte/dip..._paper_6086.pdf (http://www.hfm-detmold.de/eti/projekte/diplomarbeiten/dsdvspcm/aes_paper_6086.pdf)

German complete websites with pics etc, partly English citings:
http://www.hfm-detmold.de/eti/projekte/dip...m/xdslindex.htm (http://www.hfm-detmold.de/eti/projekte/diplomarbeiten/2004/dsdpcm/xdslindex.htm)

German complete Theses as pdf, 99 pages:
http://www.hfm-detmold.de/eti/projekte/dip...rbeit%20neu.pdf (http://www.hfm-detmold.de/eti/projekte/diplomarbeiten/2004/dsdpcm/pdf/Gesamtarbeit%20neu.pdf)

short German pdf with pics:
http://www.hfm-detmold.de/eti/projekte/dip...vergleich_d.pdf (http://www.hfm-detmold.de/eti/projekte/diplomarbeiten/dsdvspcm/hoervergleich_d.pdf)

I hope, it wasn't posted before.

145 test subjects, all trained listeners in various ways, could not abx at all, they felt very frustrated finding no differences. Counting all samples, it was ca. 1450 vs. 1450, so pure guessing which sample was dsd and which was pcm 24/176,4

Only 4 guys could find difference under headphone conditions with guessing probability less than 5%.
at stereo mode, nothing at surround.


Find more studies or papers at http://www.hfm-detmold.de/eti/ (http://www.hfm-detmold.de/eti/) , select German or English version http://www.hfm-detmold.de/eti/indexen.html (http://www.hfm-detmold.de/eti/indexen.html) and look under Theses.
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 24 March, 2006, 10:27:48 AM
I remember having read this. It must have been discussed somewhere already.

Their statistical model is flawed.

The statement that 2.76% of the completed tests yelded results within the range of "critical probability" (i.e. less than 5% probability of guesswork) is very funny.

If things went completely according to chance, 5% of the completed tests should have yelded results within the range of critical probability, since by definition, critical probability is 5% !

They give detailed information about the results they got. We should be able to see if some results are really significant. But we need to define what is a success and what is a failure.
Implicitly, their definition of a success is "one test at least got a result equal or better than a given score".
We must then specify a probability threshold according to the tested hypothesis. For example, if the possibility that there is an audible difference between DSD and PCM is high, we can use p<0.05. But if we consider this hypothesis as very unlikely, we have to demand a much lower p, so that the tested hypothesis, though unlikely, is still much more probable than chance.
Then we need to compute the minimum individual score s, so that the probability, for random guessing, that at least one people get a score superior or equal to s is inferior or equal to p.

I did not do this, but one of the 145 tests at least was significant, the one who scored 20/20.
They explain that a small click at the beginning of the playback might explain this success, though this click was not conciously audible.
Title: Help! Sacd Good Or Bad?
Post by: SebastianG on 24 March, 2006, 10:58:32 AM
I wonder why they have chosen 176,4 kHz at 24 bits and not something that has a similar data rate like 16 bits. Garf once linked an interesting paper that compared DSD to PCM 176,4 kHz at 8 bits. The PCM thing still seemed to be superior in terms of SNR and linearity.

Sebi

edit: wording
Title: Help! Sacd Good Or Bad?
Post by: Pio2001 on 24 March, 2006, 11:36:19 AM
Probably to stay close to real-world applications.