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Old 1st April 2012, 01:12 PM   #111
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Quote:
Originally Posted by abraxalito View Post
..... he's got skin in the S-D game......
You're talking about S-D, that is delta-sigma (D-S)?
Bruno is fully into pulse width (class d amps) and digital d-s; I met him some weeks ago, and in his (non-debatable) opinion NOS has serious flaws; maybe one more engineer who also needs to listen instead of lending his ears to the oscilloscope.
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Old 1st April 2012, 01:29 PM   #112
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Quote:
Originally Posted by abraxalito View Post
Or perhaps he has a proprietary fix for it?
Time will tell.

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Originally Posted by pieter t View Post
...maybe one more engineer who also needs to listen instead of lending his ears to the oscilloscope.
Yes, because HiEnd clearly needs more "engineers" who don't know 1st year university math and design by "ear".
Moreover Bruno's approach clearly doesn't work because his designs sound terrible (pro audio and hiend market alike will agree) and not "analogue" enough.

Oh, be sure to visit a Shaman instead of a proper doctor when you have health issues (moderators please move this to the vendor's area if you feel I'm promoting my "trade" ).
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Old 1st April 2012, 01:40 PM   #113
qusp is offline qusp  Australia
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yeah because designing Class D amps by ear would be really effective....

haha beaten to that line by TheShaman hehe

i'm curious about his views on convolver based DSP though, it does make me wonder if its just because he cant fit one effectively in the box. in my experience this type of DSP produces audio that is unlike anything ive ever heard.... at audio meets and shows included. i dont believe it can or should be used to completely replace competent speaker enclosure and room design, but it can sure help to augment those that live in the real and less than perfect world with less than infinite wallets

Last edited by qusp; 1st April 2012 at 01:49 PM.
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Old 1st April 2012, 02:14 PM   #114
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Quote:
Originally Posted by pieter t View Post
You're talking about S-D, that is delta-sigma (D-S)?
Yeah - some people put the sigma first, others like the delta to come before the sigma.

Quote:
Bruno is fully into pulse width (class d amps) and digital d-s; I met him some weeks ago, and in his (non-debatable) opinion NOS has serious flaws; maybe one more engineer who also needs to listen instead of lending his ears to the oscilloscope.
No, people like me need people like Bruno not to listen so as to create our own market niche. I celebrate his refusal to discuss NOS's benefits
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Old 1st April 2012, 02:32 PM   #115
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Originally Posted by abraxalito View Post
No, people like me need people like Bruno not to listen so as to create our own market niche. I celebrate his refusal to discuss NOS's benefits
I know what you mean.
I just couldn't keep up with the more agile of my mates in basketball nor football, so I focused on body-building/gym-related stuff where I was better than most and I felt good about it.

qusp, I believe he's just trying to point out DSP shouldn't be expected to work miracles and you still have to design a proper loudspeaker and treat the room. They already utilize DSP in Hypex and Grimm.
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Old 1st April 2012, 02:52 PM   #116
qusp is offline qusp  Australia
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Originally Posted by TheShaman View Post
I know what you mean.
I just couldn't keep up with the more agile of my mates in basketball nor football, so I focused on body-building/gym-related stuff where I was better than most and I felt good about it.
well played
I used to play power forward in representative and state level basketball till under 16, but then everyone became larger than me, soi worked on my outside jumpshot and still played good ball, i never did play starting 5 at state level under 18 though....

Quote:
qusp, I believe he's just trying to point out DSP shouldn't be expected to work miracles and you still have to design a proper loudspeaker and treat the room. They already utilize DSP in Hypex and Grimm.
yeah i know that confused me initially; however i dont think they use convolvers, only digital crossover/EQ and maybe some delay, not full room correction. This way the XO can still be built into the box, while a convolver would mean they had to build the PC into the box, or shift the XO to the PC. the way i understand it the PC connection is just for uploading filter coefficients in their products.

I agree with it the way you state it though.

Last edited by qusp; 1st April 2012 at 02:55 PM.
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Old 1st April 2012, 02:58 PM   #117
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Originally Posted by abraxalito View Post
Excellent - thanks a lot! I really love such detailed descriptions of comparisons Real science.
I take offence in that afirmation. SCIENCE represent experiments that are controlled, measured and repeted by peers with the same results.
This is a bunch of blablabla, subjective oppinions, you can call it poetry if you like, but is NOT science.
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Originally Posted by abraxalito View Post
Yeah - some people put the sigma first, others like the delta to come before the sigma.
Some would say that the Sigma block comes before Delta, so they call it Sigma-Delta. Some say that functionality is primarely Delta and Sigma is applied to the Delta product, so they call it Delta-Sigma.

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Old 1st April 2012, 03:54 PM   #118
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Originally Posted by SoNic_real_one View Post
I take offence in that afirmation. SCIENCE represent experiments that are controlled, measured and repeted by peers with the same results.
This is a bunch of blablabla, subjective oppinions, you can call it poetry if you like, but is NOT science.
Your hearing impared agenda is showing, yet again.
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Old 1st April 2012, 03:58 PM   #119
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Your wishful thinking agenda is showing again. For rest of us, Unicorns don't exist just because you wish to...
As for my hearing - I didn't see you chipping in when somebody posted here two similar files for a listening test. I did and I detected the right answer.

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Old 1st April 2012, 04:23 PM   #120
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Do you recall the part number of the filter - presuming it was an off-the-shelf chip you were using? Curious to know stuff like passband ripple (for time-smearing) stop-band rejection and OS ratio (4X. 8X?).
It was an AD1896 ASRC chip, which contains a half-band FIR sinc filter that can be bypassed on-the-fly. The oversampling ratio was x4. Although, to be fair, this wasn't a totally controlled comparison between NOS and OS modes because the OS mode also features jitter suppression, which the NOS bypass mode does not.

Quote:
As regards 3) yes again in agreement. What I suspect might be happening with cymbals in the OS case is that the half-band filter is aliasing above 20kHz. Bruno Putzeys has commented on how aliasing smears out HF transients so they appear to come from the speakers rather than from the recorded space. The solution is to abandon half-band filters and use apodizing ones.
I've always wondered about the notion of aliasing in the DAC reconstruction filter. By definition, a D/A converter can only reproduce baseband signals at the Nyquist rate of the conversion frequency. Aliasing artifacts can only occur symmetrically around integer multiples of the sampling frequency, all of which are above the baseband. Baseband signal aliasing would seem to only be possible at A/D conversion, and downsampling conversion. I believe that fully apodizing filters (fully in the stop-band by 19-20kHz) sound more natural for the same reason that NOS sounds more natural. They remove either the A/D or the D/A sinc filter response - see my further thoughts on this at the bottom.

Quote:
1), were you subjectively aware of the loss of HF on NOS compared to OS?
While I was aware of the HF loss, the most obvious differences between OS and NOS seemed to involve midrange clarity and soundfield clarity, and the absense of listening fatigue even on music with little apparent upper register content.

Quote:
3) I relate back to the half-band aliasing effects - smearing out of HF tends to mask the cues which allow us to listen in to the ambience of a recording space. The smearing could also have the effect of subjectively widening the soundstage, but this is a bit of a long-shot
My theory as to why redbook CD has traditionally dissapointed many audiophiles follows, taken from my recent posting on the subject in an AudioAsylum thread:

The technical problem I have with CD is that the intended signal (music) contains time-domain sensitive information. The 44.1ksps CD channel rate was designed with the frequency-domain requirements of music in mind. As far as I can tell, little to no consideration was given to the time-domain implications of the medium, either for recording or playback. The root of the time-domain problems are two-fold, as I see them. First, the fact that CD's channel bandwidth of 22.05kHz is so close to the recorded information bandwidth creates the requirement for very sharp anti-alias (recording) and anti-image (playback) filter responses. Which necessarily have a poor time-domain response, manifesting as the now fimiliar high-Q filter ringing. As it turns out, this severe ringing is fundamental to producing an accurate frequency-domain reconstruction of the original signal upon playback. Yes, I'm also aware that such ringing occurs at the edge of the ultrasonic range and should be pretty much inaudible, but that is only half the story, to which I'll shortly return. If the information to channel bandwidth ratio were wider, as it can be with high-res. digital, then both the anti-imaging and anti-aliasing filters could be much less sharp, with greatly reduced time-domain distortion. Mike Story of dCS has published a paper concluding that to be one of the reasons why high sample rate digital sounds superior to CD.

My own empirical experiments lead me to suspect that a second, non-obvious mechanism is also at work. I suspect that the time-domain problems extend to the dynamic inter-action, or intermodulation, of the near ultrasonic ringing responses of the multiple SINC filters utilized from recording through to playback. Resulting in artifacts within the audible range.

One of my self designed experimental DACs contains a programmable digital SINC filter. This programmable filter has enabled me to empirically evaluate oversampling, non-oversampling, and apodising digital filters. Here's what I heard. As is well known by now, non-oversampling, aka, NOS - which eliminates the playback SINC filter but does not affect the recording and mixing SINC filters - produces the natural and non-fatigueing sound so typically lacking in CD. Apodising - which retains the playback SINC filter, but removes the affect of the recording and mixing SINC filters - sounds equally natural and non-fatiguing as NOS. Oversampling - with all SINC filter responses in place - on the other hand, produces the typically fatiguing and course CD sound.

My hypothesis is that the primary source for what we have come to know as digititus, or traditional CD sound, is the dynamic intermodulation of the multiple SINC filter time-domain (ringing) responses of the standard CD recording and playback chain. Eliminating ONE OR THE OTHER of these sinc filter responses, so that only a single sinc filter reponse remains, greatly restores the natural and non-fatiguing quality otherwise absent. This hypothesis would also explain why high sample rate audio is often disappointingly not completely rid of unpleasant CD type artifacts. The use of SINC filters across a high sample rate chain could still produce time-domain filter response interaction which are audible. I will surmise that high sample rate recording-playback chains which take advantage of the extra channel bandwidth not for increased frequency-domain signal capture, but for utilizing anti-alias and anti-image filters having much less ringing in their time-domain responses, will subjectively provide the best high sample rate audio quality.

I've not yet developed an experiment to test this hypothesis, so it may prove faulty in so far as the exact distortion mechanism responsible is concerned. The empirical results, however, have been consistant and very obvious.
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Last edited by Ken Newton; 1st April 2012 at 04:52 PM.
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