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Author Topic: World's first NOS 24/384 filterless DAC  (Read 256433 times)
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Ava12
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« Reply #15 on: December 02, 2008, 12:44:15 am »

God that sounds like a dream to me. drool
I have to build this one someday!
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« Reply #16 on: December 02, 2008, 08:15:56 pm »

Hi Peter,

How do you think to connect a SACD transport to the NOS1? maybe via HDMI?. I have 3 SACD players, OPPO DVD player, PS3 and Cary SACD 306 Professional, I think that only the Oppo and PS3 can output the SACD digital signal, and only via HDMI, and I don't know SACD drives for computer player for use firewire connection. 
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« Reply #17 on: December 02, 2008, 09:31:30 pm »

Good question and an honest answer : currently I don't know yet.
I think anyway the Oppo is known to be able to capture the DSD output and the ESS Sabre for sure is able to digest DSD.
But as with all features from my second list above, they have to be created explicitly. So, that is a theoretical list and I can tell you ... to implement them all may take many weeks. Besides that, some connections just go over Firewire, and others require dedicated inputs + input switches. So please remember : what I listed is indeed theory only and allowing for it all at the same time would require a "switchboard" which may look virtually imposant, but which may be undoable at the same time. This is exactly why this "routing stuff" goes by software these days, which ... requires the programming of that. It may look all nice for you, but is a huge task for me at the same time. I think ...

Peter
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« Reply #18 on: December 03, 2008, 03:25:01 am »

Hi Peter,

How do you think to connect a SACD transport to the NOS1? maybe via HDMI?. I have 3 SACD players, OPPO DVD player, PS3 and Cary SACD 306 Professional, I think that only the Oppo and PS3 can output the SACD digital signal, and only via HDMI, and I don't know SACD drives for computer player for use firewire connection. 

Does even the PS3 output sacd through even hdmi?   I thought there was no way to get that SACD digital signal... If I play my SACDs through my ps3 they get downsampled through the digital optical output, I know that ... in any case my SCD-1 and SACDs lie dormant as I've played with computer audio for low these many years.... wait, guess it hasn't been that long.  Man, so much has changed though, been very exciting this last year.
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« Reply #19 on: December 03, 2008, 08:57:46 am »

Officially (at least that's what I know of it) the raw DSD stream can't be output from an SACD player (it's not allowed, whatever). But, from some players it can be picked up, and the Oppo (don't know the type) is known for it. It you Google around a bit, you will find it.
You'd have to take up the solder iron though ... yes

Btw, I don't own an SACD player, and there was nothing in my mind that planned to use the DSD input. But about theoretical options ...
Peter
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« Reply #20 on: December 03, 2008, 12:02:29 pm »

Hi Peter,

How do you think to connect a SACD transport to the NOS1? maybe via HDMI?. I have 3 SACD players, OPPO DVD player, PS3 and Cary SACD 306 Professional, I think that only the Oppo and PS3 can output the SACD digital signal, and only via HDMI, and I don't know SACD drives for computer player for use firewire connection. 

Does even the PS3 output sacd through even hdmi?   I thought there was no way to get that SACD digital signal... If I play my SACDs through my ps3 they get downsampled through the digital optical output, I know that ... in any case my SCD-1 and SACDs lie dormant as I've played with computer audio for low these many years.... wait, guess it hasn't been that long.  Man, so much has changed though, been very exciting this last year.

Hi, only the two first generations of PS3 can play SACD, and unfortunately the PS3 can't output a pure DSD signal through HDMI, it convert DSD to PCM 24bit/176khz.
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« Reply #21 on: December 03, 2008, 12:08:51 pm »

Officially (at least that's what I know of it) the raw DSD stream can't be output from an SACD player (it's not allowed, whatever). But, from some players it can be picked up, and the Oppo (don't know the type) is known for it. It you Google around a bit, you will find it.
You'd have to take up the solder iron though ... yes

Btw, I don't own an SACD player, and there was nothing in my mind that planned to use the DSD input. But about theoretical options ...
Peter

HDMI from versions 1.2 and up can transmit DSD securely with HDCP (High-bandwidth Digital Content Protection).

I just asked to Oppo if its players can output via HDMI pure dsd signal, I am waiting its response. I have read that the DV 980HD can do it, but the DV 983HD converts DSD to PCM, I don't know about the DV 981HD model that I own.

Also there is some Pioneer, Sony and Denon player that has a SACD digital interpace call iLink, that it is firewire, but I have read that Pioneer convert DSD to PCM.
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« Reply #22 on: December 03, 2008, 01:18:45 pm »

scratching ... Ok, apparantly I must dive into this. My, say, personal problem is that I derive much from the movie world, where getting audio over HDMI seems to be a tough job, and no one rule seems to exist (mind you, this is related to multi channel vs. 2 channel (where things *do* work) but in relation to AC3 and DTS *and* that 2 channel is not an option there obviously.

I could also say : do you know (definetely !) about a soundcard with HDMI input, which ... well, just works. From what output ? ahh, I don't know, because where is the CDPlayer with HDMI output ? and would there even be a reason for it ?

Right now two external DACs spring to my mind which officially support DSD meant to be read from files (computers), and both live in the recording world only (no, I didn't say Pro world !). So, an example is DXD (which is 352800) which nowadays often is used for mastering DSD. Also it goes the other way around : DXD can be used to edit DSD (DSD can't be edited, or it is too cumbersome to do it).
From this, I am fairly sure (but not 100%) that where DSD is converted to PCM without any losses, it would be 352800 PCM. Now, you might have a DAC that can do it (but you won't), but now where is the soundcard to pass it through ?

If I keep on typing, I just as well might find the solution automatically, but I only wanted to indicate this is all not so easy (at all).

Note that many DAC chips (!) are able to receive DSD, just because they're delta sigma, and because of the high oversampling rates needed for that principle, it's relatively easy to support DSD as well. This does not mean that "we" can use such chips for SACD material, which for 100% sure won't be playable from within a PC anyway (although mr. Putzeys created one in private). Thus, those chips are meant for being in SACD players ...

If you are looking for an external DAC being able to play the contents of e.g. your Oppo, you must catch the DSD stream (I think several external DACs exist with DSD input, just like I could provide it with the ESS Sabre).
If you can only catch the PCM stream it should not be downsampled, and if I'm well informed (see above) it's a dead end, unless you have a 384000 (352800) DAC.

All 'n all, right now, I don't see where HDMI comes into play. But hey, I sure can't know everything, so if anyone knows more, please tell it and consider this as (well meant) BS. Happy

Peter
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« Reply #23 on: December 03, 2008, 07:26:48 pm »

Reply from Oppo:

The only player which we have which supports native DSD transportation is the DV-980H. All other products convert DSD to PCM.
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« Reply #24 on: December 04, 2008, 01:03:22 am »

Personally I would like my future "ultimate" DAC to feature a BNC clock input.
I belive an external clock with huge PSU can be good. I am not an engineer, so I wonder if this option is easy to implement in your new DAC prototype?

An external clock shouldnt be needed. If done well (really well) the DAC has to be the master clock.

Quote
Another thing is that most DAC's are working from x48kHz samplingfrequency. (48, 96, 192, etc). But the CD is 44,1kHz. So inside the DAC, there has to be a samplerate convertion (SRC) chip in front of or integrated inside the DAC chip.

A NOS design like this (Peter correct me if I'm wrong) does not change the sampling rate, whichever you put in, that will be out. It is filterless. The beauty of peter creation is that it can accept 48, 88.2, 96,176,4 and up to 192k. Just to cite all available sampling rates Happy

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I believe this SRC converters are "not good".

No, they are cr*p, they ruin the signal and add jitter, to avoid like all digital filters.

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« Reply #25 on: December 07, 2008, 10:28:56 am »

Just some background on "non oversampling" (NOS) for those interested :

For me, unlike the first explicit provocer of it (Peter Qvortrup working for Audio Note) it is all about the squariness of waves. In music "square waves" exist all over, and if it isn't a synthesizer which can exhibit them as exact as can be (when no analogue devices where in the mix at recording), it is the coincidence of matters. An example of the latter in nature is thunder, which can be visualised by looking at the front of a surf water wave. And, while thunder is an example of the large level (think bass like) of squarish waves, instruments like a trumpet exhibit them at a more detailed level (higher frequencies). A tick on something like the edge of a snare or tom drum would be another (kind of) example, where the "square" exposes as a very fast (read : steep) transient.
At least in 44K1 sampled WAV data, transients exist ranging easily over 2/3 of the total voltage range, meaning that the voltage (for a 2V RMS output DAC) will change from e.g. -1.3V to +1.3V in one go ! So, in analogue form this cannot exist, because it would imply an inifitly fast rise time, but with 44K100 samples per second there's just no more "resolution" to catch the steps which may be there in (analogue) reality.
Note that with e.g. 176K400 samples per second there's 4 times more room for in between voltage steps, and a transient which is captured at 44K100 exposing one go, could show 4 steps at 176K400. Or still one, because the sampling rate is way way too low to mimic analogue.

Above I mixed two principles :
1. When a steep transient (or square sound) is really there, it should be expressed like that (first part of the above);
2. When a not so steep transient (or less square sound) is there in reality, it should not be expressed more square than reality was (second part of the above).

The second part is theoretically solved by a higher sample rate;
The first part is solved by just not rounding the squares.

The story becomes confusing, knowing that the first part will be emulated (!!) by oversampling. Note that oversampling as such, is nothing else than adding interpolated samples which are not real, but do round squares which were not perfectly square in the first place. In order to understand this principe, do this :

Draw a square 2d wave with the corners not being 90 degrees exactly. This mimics what happens in sound. Those corners must be drawn in digital steps and not in an analogue round fashion. The steps you draw are the samples take.
Now adjust thee drawing, by interpolating the steps with a factor of 2. Thus, each step becomes 2 steps. This must be done at the outside bondaries of where the rounding of the corners started. Thus, looking upwards, the rounding goes downwards. And, looking sideways (the top of the wave) the rounding goes sideways.
After this one step of interpolation, you can already see that at doing it again, the square wave becomes more and more rounded. Do this an infinit number of times, and a pure sine will be left !

The latter is what this is all about : oversampling DACs make sines out of original squares, and the sound is destroyed. Oh, it will be less harsh because of less squares, but "harsh" is relative, and when a trumpet shows the "harshness" of just that instrument, it should stay like that.

It is obvious that this all can be related to transients, and that steep transients will become less steep, and dynamics will be destroyed.
Whether we talk about squares as such or about dynamics, both are the exact same subject.

As a side note, keep in mind that when an exact square is fed to the DAC, oversampling doesn't matter a thing, because there will be no steps to interpolate (draw an exact square wave now, and try !).

When you understand the above, you will see why some people like oversampling DACs and others like non oversampling DACs;
Both contribute to something which could be worked out for the better, but when the oversamplig DAC works out for the better, this can only be because of too few samples in the data. With this I refer to the exmple of a transient being recorded too steeply, because there was no room to have more steps, those steps always being there in (analogue) reality. Thus, the higher sample rate allows for those steps which are better for theory on one side, but since the steps are no reality at all (the real steps were different), it works out worse at the same time.

Now here is the important part :
An oversampling DAC creates less reality because a too steeply captured transient will be flattened in an unrealistic way;
An oversampling DAC creates better reality because a too steeply captured transient, which is unrealistic by itself, will be flattened.

Got this ?
There will be no scientific answer to which of each "features" works out for the better, and both have an unreal result.
BUT :
There's also the matter of the real transients and squares which are reality from the start, and the oversampling DAC will get those out of the way just the same ! And thus :
The "feature" of the non oversampling DAC to at least pertain those transients and squares comes out of the equation as a positive.
And thus all 'n all the non oversampling DAC wins with 2-1.

There is quite some more to it, like oversampling shifting the nyquist frequency and therefore requiering less filtering, and for example of 352K800 it is said that just no filtering is needed at all. 44K100 just officially needs the filtering, which will operate in the audible domain.
The NOS filterless DAC just doesn't do that, and where filtering by itself will again destroy sound, the artifacts (if audible at all) from "aliasing" are taken for granted.
Again, "if audible at all", while at the same time synthesizer music is completely destroyed by oversampling. And this is audible for everyone.


Having said this all, the phenomenon "oversampling" needs some additional clarification;

An oversampling DAC, in 95% of cases is so, because it can't operate without it. The sigma-delta DACs are the example, with DSD (= SACD) in thee same line of working.
These 1 bit principles can operate only by heavy oversampling, meaning 256 times or more, up to MHz's.

The other 5% of cases, are about multi bit DACs which just do not need the oversampling in order to operate, but, it is done anyway for the reasons of shifting the nyquist frequency as briefly referenced to above, and besides that will need filtering which is always needed just *because* of the oversampling. In this case we talk about oversampling to e.g. 176K400 which is 4 times only. Note though that 176K400 actually is a strange number, and 192K is more normal, but since 192K can't be divided by 44K100, first a common denominator has to be found, and from there on heavy oversampling (underway) emerges again.

Although the definitions do not exist explicitly, one could say the "oversampling" is the heavy kind, here posed as a pure negative, while "upsampling" is going from e.g. 44K100 to 176K400 directly.


Lastly, and to get a bit from the abouts, at 44K100 sampling rate, a 22050Hz pure sine is represented as a pure square. There's just no more (sample) room to make it less than pure square. Similarly, a 11025Hz pure sine is represented as a square built from 2 steps. This is just audible, knowing that squares produce a ton of harmonics.
Now, upsample this 1 time, and the anomaly (for what it's worth) at the very audible 11025 is shifted to 22050 which is not audible (for most).
This is why heavy "oversampling" is beneficial by itself, because it shifts the anomaly into the inaudible area. But keep in mind : in a fake fashion.

All together you may get the grasp of the importance of a 192K non oversapling DAC, meaning :
When 192K native material is played, the anomalies have been shifted "over 4 times better" into the inaudible area anyway (but now with real samples !), which makes non oversampling "over 4 times more legit".

Peter
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XXHighEnd Mach II Stealth LPS PC -> Xeon E5 2640v4 with Hyperthreading On (20 cores) @~720MHz, 32GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = *14/-/1/1/1/* Q1Factor = *1* / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = *0.9*  (max 120) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = 43 / Nervous Rate = 100 / Cool when Idle = n.a / Provide Stable Power = 0 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filter Low 705600 / -> USB3 from MoBo -> *Lush USB 1m00* -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> B'ASS Current Amplifier -> Blaxius Interlink -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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PeterSt
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« Reply #26 on: December 07, 2008, 11:53:20 am »

Ok, after a week of playing with the DAC (I still don't think is is burned in sufficiently), I want to express some feelings about it :

I think I said it before : For a longer time I regard my own system as a whole to be the best I ever heard. Of course this doesn't say much, and a chance exists that many of you just have it better. Only statistics of me being around at other places tell that the chance is little.
Small conclusion : whether or not I have this "best system" or not, it should be better than average, and this is my reference.

The above was with my old 18/96 nos DAC.

Now, with the 24/192 in NOS non filterless mode, and if I had to apply a mathematical figure to it, I'd say it is 10 times better. Whoops, that's a lot.

I don't think that I, right now, can find commonly known wordings for it. I mean, there's nothing like spatiousness, wider stage, instrument positioning. That is, I don't recognize this or can't as off yet. You could also say : this was good already before, and improvements must be in other areas (at talking about "10 times better" Happy). But I'll try a few expressions as far as I can recognize them;

First of all, I think I can now at last hear the benefits of higher sample rate recordings. Remember, I could not do this before because of lacking bits 18 only) at 96Khz, and not being able to play 192Khz at all. Of course I could use my oversampling sigma delta (Fireface), but this is apples and oranges to begin with, and btw nothing good at all. So :
There appears to be a phenomenon to my judgement, which is related to being able to differentiate in voices singing in parallel. Remember, I'm talking about the higher samplerates, and how to hear they are better.
It doesn't matter where I'm in the room, as soon as background voices (male or female) start to sing, you can kind of count with how many they are. I don't talk about where they are, but with how many. So, a matter of being able to hear person A, B and C separately.
Maybe this is nothing new (for you), but so far I couldn't manage to find the absolute difference because of the apples and oranges thing.
In other areas I don't perceive a difference. Or not yet.

Back to normal 44100 playback;

The longer I play with the DAC, and the better album examples I can find for it, the more amazing the bass becomes. I have no words for it, but the sheer difference of hearing beautiful bass before (which already was a tough job for me, and is much related to XXHighEnd versions), with complete life-like basses from now ... I can't express it differently. I think earlier I already talked about the ability to hear the wood of a bass, but right now I'd say that someone with knowledge should easily be able to tell the "manufacturer" of the bass instrument. I could also say : before I was very happy when I could hear the strings of low bass vibrate (good resolution in that area), but now each (double) bass and cello just sound different. Give me a few months and I will be able to tell the artist by listening to the bass only ! (and I don't mean the means of how it's played, just the sound of it).

There are also strange things going on;
My last tweak being the SSD with the OS on it, already brought the clear vibration of strings being pulled too loudly. So, the metal itself makes the sound in this case, and while this is a very profound sound (dzziinnggg) I never heard it before from a speaker. I checked it with others to be sure, but it really is so. Why ? I'm puzzled, just because the sound is so profound, and by the way so very naturally. But, I never missed it before, so how the hell could I tell those strings were pulled so hard on the specific recordings ?
Now, with this DAC, it appears that it will be very hard for those musicians *not* to pull those strings too hard. They just do, and they do it always. Man man man, I get tears in my eyes at writing this, because it is really a complete new dimension in audio playback.
I don't care anymore whether there's a stupid clarinet showing off as often with jazz albums, it's the bass man (and often woman) making the sound !

A few days ago I bumped into James Cotton and his album "Deep in The Blues", and besides I seem to be able to clearly hear he is playing an acoustic bass guitar throughout the album, he is playing that guitar "toggling" (5 individual fingers) like a spanish guitar would be played without chords. But now imagine the deep sound of a bass guitar, and that those fast individual bass notes just work out as intended (instead of a smeared bassy sound). Man man man.

I think I said earlier that everything seems to be supported by bass. This is literal;
This works so much throughout, that it merely looks like an anomaly. I kept on paying attention to it though, and the only conclusion I can draw is that - like the wood from the bass - it is the recording room/space I am hearing. Think of a room and bass measurement, and the knowlegde that the room adds bass to your sub woofer, just like the cabinet the bass driver is in does. Now I hear the same throughout, and this wasn't there before *at all*. It is strange though, because you are not used to it. Even voices can express some sub low which sure can't come from the voice itself, but with some hall and reverberation it just sounds natural.
Besides this general phenomenon, I experienced quite some times the whole house started rambling because of sub low output, and this is on recordings I never heard it before.
Added to this that I have a couple of tracks to trim the bass vs. sub woofer output where the subwoofer output should stay normal in all cases, these recordings do not exhibit more sub woofer output. To me this is the most strange, but proves all is still right, and it is not just "more bass output". It merely looks like slow waves with not too much amplitude just being able to express now, where they were killed before.

Where I was keen before on having just directional bass output - knowing that much of that is caused by higher frequencies around that bass fooling you - this is now just "completely directional". And this is the most interesting, because if you now can hear where the man playing the bass really is, or what about two of them, this is just again another dimension. But keep in mind what I said above : when the metal of the strings becomes audible, this is the base of the "guideance" of the directional bass !

About my stories about standing waves disappearing when things are all right : this again vastly improved. But, know in a very understandable way;
As a strange example I want to mention Madonna with the ever accompanying synth bass. I used to know this as "bass" of which was audible it is a synth. Strangely enough now this isn't so much of "bass" as such anymore. The synth now expresses it's short pulsed output, and where things got smeared before, it's now just the pulses you are hearing, and the deep sunding coming from it disappears.

Read the latter again; it is contradictionary to all the other phenomena around the bass. Everywhere we have more bass, and here we have less ?
Well, in fact I just explained it. The waves are less smeared, and the individual vibes are expressed all over, and now the "individual vibes" and the standing waves come into play : the more the vibes can be expressed individually, the less they interact in space, the less anomalies come from that.
That in the mean time bass output itself is "higher" (read : better) is just another phenomenon I think, caused by the dedicated PSU. So, the better bass output seems PSU related, and the better accuracy is DAC related (though the PSU will enable to follow the accuracy).

Talking about accuracy, well, this expresses all over. So, now we're up to the higher frequency regions, and may it be "speed" (slew rate of the DAC) or the better translation of it because of the PSU, there is detail which ... well ... can't be expressed properly either.
This too is strange, because normally I would be able to give examples, talk about hi-hats and cymbals etc., but somehow here too other definitions are needed. I don't think hi-hats and cymbals etc. are better. I do think brushes are far better and it might be the only thing I could reason out why. The other things express in things which can't be told. Not by me, not yet;
Yello is always my example whether the NOS principle acts as intended (because synthesizers in an interesting fashion only). One can't talk about synthesizers and how they should behave and all. One *can* say though, that many more interesting sounds were heard, and the better music reproduction becomes, the more value those two guys put into their music. Might it be some sub low intended (!) reverberation, or a dazzling sound sweeping from left to right, neither is audible when the playback system isn't up to it.
My point is merely that these gagdets (when audible to start with) have less or more fragility. With synthesizers I think you can say that - besides things being audible or not - it is the fragility with which the sounds are expressed, that matter. Just think about some triangle and saw tooth waves interacting with eachother, and what might come of it. You can bet though that the Mini Moog guys from the early days, as well as the modern synth people from today, try to create interesting sounds, as long as they don't try to emulate violins. If you are not into synthesizers I am sure that this is because you never obtained the so much interesting intended sounds coming from them. Try Momita (uhhm, emulating volins) and combine classical with synthesizers, but in the mean time try to imagine how in the world the guy was able to create all those orchestra sounds from his synthesizers, all to be programmed right from the base.
But don't try your PC speakers ...

Lastly for now something commonly known, but happening here just the same :
Look at the picture below. For me this doesn't invite much to sit down and listen to it. Of course I tried, but she sings exactly as she looks like there. Yesterday - at testing various hires albums - I had to try her too (this is 24/88K2). Well ... since I am a man and thus can't cry I, didn't. But it was very hard to keep the tears inside. Whether it is the superbly played bass or just she herself, I imagined a woman with a history all laid in her songs on this album. Before it came to me as some shattering around, but now it just worked (as intended I suppose).
As said, a commonly known phenomenon, but for me in this case a radical change.
Of course, when the bass is not there (as it wasn't before) the "technical" fun has gone already. So, wanting to listen to the next track for the bass already, makes you dive into this album more than before. But still ...


Enough said for now. As you can see I don't have much to talk about, except for that bass. I guess it is too overwhelming to be ready for other merits at this moment. Later ...

Peter


* Barb01.png (131.22 KB, 505x502 - viewed 2569 times.)
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For the Stealth LPS PC :
W10-14393.0 - September 2017 (2.08)
XXHighEnd Mach II Stealth LPS PC -> Xeon E5 2640v4 with Hyperthreading On (20 cores) @~720MHz, 32GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = *14/-/1/1/1/* Q1Factor = *1* / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = *0.9*  (max 120) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = 43 / Nervous Rate = 100 / Cool when Idle = n.a / Provide Stable Power = 0 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filter Low 705600 / -> USB3 from MoBo -> *Lush USB 1m00* -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> B'ASS Current Amplifier -> Blaxius Interlink -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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« Reply #27 on: December 11, 2008, 10:00:52 am »

hello P
talking about system:
what amps and speakers do you use?
would be useful for my own reference to evaluate the "value" of your opinions
not that I don´t trust them,but useful to know in what context U talk about  "best bass" etc
if you e.g. had minimonitors Happy,that would be a useless statement to me
best
Leif "curious" Christensen(LMC)
Norway

and Yes Pedal´s friend waiting for Buffalo Dac (ESS SABRE) is probably me.
it left US y-day and me going to Florida the 19th , I won´t build it until early jan.
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PeterSt
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« Reply #28 on: December 11, 2008, 11:54:11 am »

Quote
if you e.g. had minimonitors Happy,that would be a useless statement to me

Of course not, that would be extraordinaire ! whistle

Let me first try to explain that good deep bass hardly is related to the surface of the bass drivers' diaphragm. Oh, it might, but it can hardly make a difference in the living room. The only thing what matters IMO is less distortion with more drivers, just because their excursion doesn't need to be so high per driver.
Btw I tell this, because I have in mind that you have those double BD15 red bass cabinets ... but maybe I'm wrong. And I only want to express : I just cannot for te life of me imagine ANY "bass setup" be better than mine ... with which I only want to say : a tad smaller (than a double BD 15) is way enough ...
But theoretically, of course, more will be better, but IMHO merely in the area of less distortion at high output.
My room is 290m3 and my listening level is always around 100dB @ 4 meters. Careful here, because I use the BD-15 Ultra with Orphean, hence rather directional, and @1m this is only 110 or so.
The port of the BD-15 Ultra is closed.

Btw, maybe hard to imagine, but with the better control of the bass the excursion gets less and the output more. Think of a straight 30Hz tone, making expand and distract the diaphragm 30 times per second, displacing the amount of air the surface of the diaphragm incurs for. Now make this 10 Hz because of improper control, but keep in mind it should be 30. Theoretically the excursion must be 3 times more now, and of course it will be fumbling bass.

Ok, this lot is driven by 4 33W GainClones, and mentioned 100dB SPL is at -30dB and 1.5V DAC output (no preamp).

Each channel bears a SVS PB12+ 550W active subwoofer capable of 12Hz driven by the LS output of the bass amp, and customly crossed at 40Hz (the BD15 goes straight to 27Hz).
Since I closed all the ports, output will drop off under approx. 16Hz (16 still works perfectly, which I can tell because it is the resonance frequency of the doors surrounding the living room -> they ramble).

I can tell you, the low B of a 5 string electric bass, when played somewhat more profoundly, keeps on vibrating on your stomache, giving the unary feeling I talked about earlier, just because that 60Hz (or whatever it is) is so powerful (at each vibration !).

Quote
and me going to Florida the 19th , I won´t build it until early jan.

I'm afraid I will be earlier then (expect it any day now). Of course I will again express my honest judgement about it.

Peter
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For the Stealth LPS PC :
W10-14393.0 - September 2017 (2.08)
XXHighEnd Mach II Stealth LPS PC -> Xeon E5 2640v4 with Hyperthreading On (20 cores) @~720MHz, 32GB, Windows 10 Pro 64 bit build 14393.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = *14/-/1/1/1/* Q1Factor = *1* / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = *0.9*  (max 120) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = 43 / Nervous Rate = 100 / Cool when Idle = n.a / Provide Stable Power = 0 / Utilize Cores always = 1 / Time Performance Index = Optimal / Time Stability = Stable / Custom Filter Low 705600 / -> USB3 from MoBo -> *Lush USB 1m00* -> Phisolator 24/768 Phasure NOS1a/G3 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> B'ASS Current Amplifier -> Blaxius Interlink -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere (also from the PC).

For a general PC :
W10-10586.0 - May 2016 (2.05+)
*XXHighEnd PC -> I7 3930k with Hyperthreading On (12 cores)* @~500MHz, 16GB, Windows 10 Pro 64 bit build 10586.0 from RAM, music on LAN / Engine#4 Adaptive Mode / Q1/-/3/4/5 = 14/-/1/1/1 / Q1Factor = 1 / Dev.Buffer = 4096 / ClockRes = 1ms / Memory = Straight Contiguous / Include Garbage Collect / SFS = 0.10  (max 60) / not Invert / Phase Alignment Off / Playerprio = Low / ThreadPrio = Realtime / Scheme = Core 3-5 / Not Switch Processors during Playback = Off/ Playback Drive none (see OS from RAM) / UnAttended (Just Start) / Always Copy to XX Drive (see OS from RAM) / All Services Off / Keep LAN - Not Persist / WallPaper On / OSD On / Running Time Off / Minimize OS / XTweaks : Balanced Load = *43* / Nervous Rate = 1 / Cool when Idle = 1 / Provide Stable Power = 1 / Utilize Cores always = 1 / Time Performance Index = *Optimal* / Time Stability = *Stable* / Custom Filter *Low* 705600 / -> USB3 *from MoBo* -> Clairixa USB 15cm -> Intona Isolator -> Clairixa USB 1m80 -> 24/768 Phasure NOS1a 75B (BNC Out) async USB DAC, Driver v1.0.4b (4ms) -> Blaxius BNC interlink *-> B'ASS Current Amplifier /w Level4 -> Blaxius Interlink* -> Orelo MKII Active Open Baffle Horn Speakers.
Removed Switching Supplies from everywhere.

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leifchristensen
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« Reply #29 on: December 11, 2008, 12:34:22 pm »

You only miss analogue then! Happy
interesting; I seriously considered changing my Trio´s for a similar setup with custom hornloaded double ultra with Orpheans in D´Appolito
less floorspace!
my listening area is of similar volume and I play at about same levels at same distance
I backed out because of little secondhand market for trios and some serious considerations about the modified BMS driver.
I ordered the Buffalo with the I/V stage incl.x-formers and psu s but obviously one could use Borbely´s discreet buffer
however this way I get up testing quicker and can alway develop it further.
by the way, it´s only the usb drivers that prevent the usb interphase from sending hi-rez.couldn´t someone rewrite one for audio use.or else we´ll have to wait for the Buffalo firewire I2S bus or USB 3.0
best
Leif
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