Superclocks

[AudioBang]

I appreciate the insight. Thanks.

[barrows]

17 hours ago, bibo01 said:

Is it worth considering the LT3045 regulator for it?

ADM 7150 with good implementation is actually the better regulator for an XO.

 

17 hours ago, AudioBang said:

Just curious, is your DAC using an ES9018 Sabre or ES9038PRO?

 

On a marketing sheet, ESS states that the 9038 is pin compatible with previous versions [9018] and I was wondering if the 9018 could be swapped for a 9038 PRO without any software considerations....

Just me contemplating more Hi-Fi stupid things to do 

No, 9038 Pro has different requirements of the I/V stage, you can however direct swap in the ESS 9028 for the 9018, but you will need to change the control software as well.

[mansr]

1 hour ago, barrows said:

ADM 7150 with good implementation is actually the better regulator for an XO.

Why is that?

[AudioBang]

On 8/5/2018 at 10:25 PM, AudioBang said:

Not sure if anybody is still out there on this thread - [one year later]...

 

I'm posting my experience upgrading the Crystek CCHD 575 100MHz oscillator [$25] in the Vega DAC to the Pulsar [$425] merely for perspective for anyone interested. Conclusion: In this application, reducing the power supply noise made a significantly bigger improvement over TXCO to OCXO upgrade at 10X the cost.

 

The actual measured phase plot provided with the Pulsar showed a 10dB improvement at 10Hz and 100Hz over the Crystek CCHD-575 generic datasheet published on their website.

Using the Pulsar, the soundstage did snap into another level of focus along with improved transient response. Without all the details about burn in time and listenable improvements,  my immediate thought was that the Crystek 575 is a great clock for $25! I guess that tells you in this case I was not totally WOW'ed for the money.  The improvement was definitely there though.

 

Two years later,  I added a gold Bybee purifier both in line with the ground to the Pulsar as well as in line with the 3.3V Vcc to the Pulsar.

That was when the soundstage really kicked up another two notches. That was a WOW.

I can't speak to the authenticity of marketing mumbo jumbo of Bybee, but in this application, they are definitely removing noise from the supposed Low-noise Pulsar voltage regulator http://www.pulsarclock.com/ds/Pulsar_Power.pdf . If I had to do it over again and I knew what I know now, ideally I'd have inserted the Gold Bybee  purifiers in line with the supply [and ground] to the Crystek.

 

I happened to have these lying around from 15 years ago and they are outrageously expensive today....  Tried them on interconnects and was not impressed.

 

I would really like to know what power supply noise was removed [if even measurable].

I found they work great after a bridge rectifier in a power amp but did not find a WOW in other basic applications. The Krell's I modified had vague soundstages that really opened up after adding the large Gold Bybees after the rectifiers.  

 

Hope this was useful.

 

Rich M

Supplemental :

Throughout all of my mods and equipment swaps, there  is one consistent theme  that has followed me to this point - that the highs and mids, remained consistently objectionable in certain recordings and brought my system to its knees. And until last night, I was never able to isolate it. While not related to the subject, I felt under the context of the heard differences of such enhancements, it is very relevant....

Long story short - I have been reading cable reviews and blogs going back to Art Dudley's JPS Labs Aluminata review in 2001 to Michael Fremer's more recent comments on Tara Zero with the ground isolation box. The consistent thread was that the cable shield induces noise into the conductors and blurs the sound and masks detail . Bill Whitlock of Jensen in his whitepaper points out that the diff amp on a balanced termination would cancel any noise out if the cable itself was constructed precisely enough for the noise  to be induced exactly the same in both balanced conductors.  The common thread between the JPS Labs and Tara is shield noise absorption/dissipation. I took a shot at trying to build a JPS Labs Aluminata balanced interconnect and did not succeed. Not an elegant solution for me and I sense some marketing liberties going on so I did not want to take a shot at V2.

I then  had the thought that perhaps a Bybee filter might act like a rare earth metal to reproduce the effect of the Tara grounding station.

So I inserted a gold Bybee  between pin 1 and ground  in a used Assemblage DAC  I just bought for $220 and left the $7 Hosa XLR ground lifters at the receiving ends and at last the whole soundstage opened up. All of the incremental tweaks done over the years certainly added up but the elimination of noise on the shield was by far an order of magnitude improvement. I then tried it on two different balanced digital cables and similar gains were realized. I don't know the delta between an $18K pair of Tara Zero Evolutions and what I tweaked last night but this addresses the Achilles heel that has been plaguing my system since day 1. Given the context of my post of having second thoughts about spending the money on the Pulsar, under this new discovery, despite that 20X the cost of the Crystek [and the Crystek is a great timing source!], I think the added space and dimensionality can now be more fully appreciated. 

 

 

System: 

 

Krell Evolution I amps running off 240VAC from dedicated lines all lubed with silver contact paste

Krell Evolution II preamps powered from EI core custom wound balanced transformers [replacing original torroids] fed by 240V/120V balanced EI core isolation

Dunlavy SC VI - with rebuilt external x overs using Duelund CAST caps/Inductors with RF/EMI filters near the drivers

Analysis Plus Silver Apex balanced cabling with open ground at receiving ends and Bybee Gold inserted between pin 1 and Gnd at source ends

Balanced 240VAC to 120V power using J&K Audio custom transformers

 

Assemblage Dac 1.5 modded to run off lead acid batteries, modded for balanced digital input and balanced analog outputs

using Jensen isolation transformers 

Auralic Vega Dac modded to replace internal torroid with separate EI core input transformers, Schottkys, Sparkos Regs, Pulsar Clock, 

Antipodes DX3 [upgraded]

Matrix USB -Spdif 2

Uptone Ultracap, Regen, various LPSs

 

 

 

 

[Ralf11]

On 4/26/2017 at 7:49 PM, monteverdi said:

Thats quite interesting statement considering how much Antelope Audio charged for their external clocks

 

Don't confuse consumer rip-off pricing with SQ.

 

Rubidium clocks are like graphite speaker cables.

 

search here for a more extensive thread on Rb

[AudioBang]

Half of my post for whatever reason got edited out....

My point was that in my system, the masking of detail, blurring, compression at certain frequencies/transients etc., from ineffectiveness of interconnect cable shielding severely handicapped DAC comparisons and minimized the audible effect of replacing the stock timing source in the DAC with a higher performing [10dB lower noise] OCXO.

 

By opening XLR pin 1 [ground] at the receiving ends and inserting a rare earth metal EMI absorber between pin 1 and ground at the source end, everything opened up significantly.

[barrows]

2 hours ago, AudioBang said:

Half of my post for whatever reason got edited out....

My point was that in my system, the masking of detail, blurring, compression at certain frequencies/transients etc., from ineffectiveness of interconnect cable shielding severely handicapped DAC comparisons and minimized the audible effect of replacing the stock timing source in the DAC with a higher performing [10dB lower noise] OCXO.

 

By opening XLR pin 1 [ground] at the receiving ends and inserting a rare earth metal EMI absorber between pin 1 and ground at the source end, everything opened up significantly.

Sounds like something is wrong with the design of your components or your interconnect.  With a true differential amplifier/receiver, and properly designed balanced cabling, no shields currents will matter to the differential signal pair, as that boise would be common mode and cancelled out.

Here is a great paper to study, which includes discussion of this, among other things.  (and I ma actually a Bybee believer in general)

The G Word.pdf

[Miska]

On 8/6/2018 at 5:25 AM, AudioBang said:

I'm posting my experience upgrading the Crystek CCHD 575 100MHz oscillator [$25] in the Vega DAC to the Pulsar [$425] merely for perspective for anyone interested. Conclusion: In this application, reducing the power supply noise made a significantly bigger improvement over TXCO to OCXO upgrade at 10X the cost.

 

The actual measured phase plot provided with the Pulsar showed a 10dB improvement at 10Hz and 100Hz over the Crystek CCHD-575 generic datasheet published on their website.

 

In many cases I've seen, jitter performance of DAC output doesn't change much by changing the clock part. Because the problems are more in the PCB layout design and other parts rather than the clock module itself. There's not much joy about fancy clock if it is spoiled before it reaches the DAC chip pin... And even if it reaches the DAC chip pin unspoiled, the it can still be spoiled by noise coming to any other pin of the DAC chip.

 

So I'd recommend to stop looking at datasheet plots and start looking at jitter measurement output of the entire DAC device instead. Because the two don't often correlate much...

 

[barrows]

7 minutes ago, Miska said:

 

In many cases I've seen, jitter performance of DAC output doesn't change much by changing the clock part. Because the problems are more in the PCB layout design and other parts rather than the clock module itself. There's not much joy about fancy clock if it is spoiled before it reaches the DAC chip pin... And even if it reaches the DAC chip pin unspoiled, the it can still be spoiled by noise coming to any other pin of the DAC chip.

 

So I'd recommend to stop looking at datasheet plots and start looking at jitter measurement output of the entire DAC device instead. Because the two don't often correlate much...

 

Jussi, the interesting thing is, swapping to a better specified clock has always worked for me in terms of subjective listening, with a clear improvement, but the J-test spectrum looks the same.  I suspect the J-test is insufficient to see what is really happening, after all the J-test was developed basically to to test (the rather poor) SPDIF interface performance, we need a better measurement.

My experience is with changing clocks with the same implementation as far as clock distribution to the DAC chip goes.

[AudioBang]

I am reporting actual listening experiences so despite specs and what I read, if the experience isn't there I wouldn't be reporting it. 

Putting a battery on the clock circuit of my Wadia 27ix years ago produced a very desirable improvement. 

 

My original post described a definite improvement substituting the Pulsar for the Crystek 575 in the Vega Dac.

My opinion regarding that experience was that it did not reflect 20X the cost of a $25 Crystek.

After altering the shield grounding in my system, the level of resolution was indescribable. 

I had no idea what I had been missing all these years. And under that context, the imaging/soundstaging  improvements were much less discernible before the said shield grounding change.

 

According to Jensen's Bill Whitlock, noise being induced from a shield to the balanced signal carrying conductors is not induced equally.

And that is the problem

He even goes on to describe how the different color dyes in the + and - signal insulation affect the dielectric properties of the cable.

His view is that the cable needs to do its job delivering  identical  noise on each balanced conductor so that the diff amp can do its job.

I guess anyone can say anything and this is just his view.

But the experience is undeniable.

 

 

 

 

 

 

 

 

[AudioBang]

BTW, thanks [Barrows] for the GWord PDF. I see Whitlock's Pin 1 problem was referenced in the paper.

When I spoke to Whitlock over the phone, I didn't get that we were talking the same language in terms of his use of the word "hum and noise" vs. the result I was after which was eliminating the gray haze that, until its gone, you never knew was there obstructing the detail, calm and ease and openness of the presentation.  According to JPS Labs' Joe Scuba, "the interconnect is where everything goes wrong".

I've been screwing with this "hit or miss -style" for a few years now. It was the Tara grounding box description that had me take the final step of draining the shield at the source end through the Bybee. 

 

Rather than alter my relatively new Analysis Plus interconnects, which came

highly recommended by MSB, I altered my cheap Assemblage DAC by combining both channels' Pin 1 to a single Bybee going to ground inside the chassis. 

 

For my balanced digital cables, I used a $8 Hosa ground lifter  and drilled small holes to insert ground leads to a Bybee Gold as the absorber. I used this on two different Balanced Digital Cables with very positive result.

 

Not sure if I want to rattle the cage yet with power cord discoveries.... 

 

 

 

[Miska]

7 hours ago, barrows said:

Jussi, the interesting thing is, swapping to a better specified clock has always worked for me in terms of subjective listening, with a clear improvement, but the J-test spectrum looks the same.  I suspect the J-test is insufficient to see what is really happening, after all the J-test was developed basically to to test (the rather poor) SPDIF interface performance, we need a better measurement.

My experience is with changing clocks with the same implementation as far as clock distribution to the DAC chip goes.

 

I think J-test is very good, since it tells if there's pollution on the clock. A DAC with fancy clock may perform poorer than another DAC that has regular clock. I prefer to pick the DAC that performs better, not the one that has fancier part inside.

 

Problem in changing clocks is that it is hard to place new clock properly in regards to PCB layout. For example if a DAC is originally using the NDK clock part and then you would like to swap to Crystek, no way it is going to fit there. If you put any wires there, then you are certainly big time spoiling the PCB layout and likely adding a lot of jitter. So at least the performance needs to be always verified by measuring, to check that it doesn't actually get worse.

 

Can you post some J-test24 measurements what is the baseline performance of the DAC where you are swapping the clocks and it makes difference sonically? I would like to see how close they end up realizing the clock performance in practice.

 

[barrows]

9 hours ago, Miska said:

Problem in changing clocks is that it is hard to place new clock properly in regards to PCB layout. For example if a DAC is originally using the NDK clock part and then you would like to swap to Crystek, no way it is going to fit there. If you put any wires there, then you are certainly big time spoiling the PCB layout and likely adding a lot of jitter. So at least the performance needs to be always verified by measuring, to check that it doesn't actually get worse.

Yes, I totally agree with this.  My experience with swapping clocks is in using the same interface for each clock, with impedance controlled u.fl micro coax as the clock line (same length for every clock <6cm) and same ultra low noise/ultra low impedance power supply to each clock.  So, no unshielded flying clock leads etc.  And I have heard differences/improvements with XOs featuring lower phase noise at 10 Hz and below.  But with the J-test, the spectrum generally appears perfect, even with a relatively "standard" XO...  

For a Sonore project a few years back we had a batch of NDK clocks measured and sorted for phase noise by a guy here in the US who does this for a few companies, so we had a range of performances of the same model clocks to try.

I wish I had access to an analyzer which can measure the phase noise, it would be very interesting to measure XO phase noise at the input pin of the DAC (chip), but the expense of that tool is not justified at this point.

 

One other note, I have a pretty good idea what lower jitter (measured), and lower phase noise XOs sound like, as back when I worked at PS audio we did some work correlating lower jitter (measured), with listening tests.  From that experience, I can say that in some system contexts it is possible that, sometimes, a listener might actually prefer the sound of higher jitter-so I could certainly see someone doing a rather poor "clock mod", using a better XO, but resulting in higher jitter (due to implementation), and then claiming way better sound.

 

One thing I saw recently, on a not to be named audio review site, was a very expensive Ethernet switch a company was selling, which included an "upgraded" OCXO.  they showed pictures of the internals and I was appalled.  The clock signal from the new clock board added was carried on a single unshielded wire, for some distance, and then soldered to the input pad of the prior clock, yikes!  This is a commercial product being sold for a not insignificant upcharge over the basic switch, and it looked like a poor DIY implementation.  Of course the review said it sounded amazing and lifted many veils, etc....  And of course, i have always been wary of companies offering external clocks to improve performance for a DAC, while this can be necessary to sync multiple devices in a studio environment, it is not a good idea for home audio: beware of these companies folks.  Best way to implement a clock in a DAC is to have it and any re-clocking, directly adjacent to the DAC (chip), with proper PCB layout, and not use an external clock going through a wire and at least two connections, etc.

 

I can have the J-test measurements you suggest made with different XOs, but the AP is in Florida and I am in CO so it takes some time.  I may get the AP sent here if/when I get an even better clock for comparisons (Pulsar).  At that point maybe I can get the AP sent here for some testing...  But from prior work, I suspect I will see a clean J-test with (excluding some very low level power supply related spikes) every clock I try.

[barrows]

13 hours ago, AudioBang said:

Deleted

[barrows]

12 hours ago, AudioBang said:

Not sure if I want to rattle the cage yet with power cord discoveries.... 

I would be interested in discussing some of your mods including power cabling etc...  Perhaps in a dedicated thread.  I am open minded about such things, but also believe that we can often be fooled by listening tests alone, and we need to be somewhat vigilant in understanding technically what may be going on.

RE the balanced cabling, a lot of audiophile XLR cables are made "wrong", as per Mr. Putzeys paper I posted, you should have a twisted pair for pin 2 and 3, surrounded by a  braided shield which then connects to pin 1, and then pin 1 goes to chassis (which can also be considered shield).  Lots of audiophile XLR cables have three wires plus shield, and this construction would likely result in uneven sharing of ground currents on the actual signal wires (pin 2 and 3), some wires also have "drain wire" added which could also result in uneven noise currents on the signal wires.  I check my cables, and my Nordost cables (Frey) are correct, with a symmetrical construction and only the braided shield carrying ground currents.  I also make sure pin 1 of the XLR shell connects direct to chassis.

 

I suspect that all the audiophile "grounding" products we see are "working" because they compensate for incorrect engineering of cables and components which is common in audiophile products (or to correct for problems inherent in single ended, RCA, wired components).  If the components are true differential circuits and wired correctly with pin 1 direct to chassis and the circuit ground to chassis at one other point, and we use correctly designed XLR cables, then there would be no "need" or advantage to the specialty grounding blocks, etc.

[Miska]

2 hours ago, barrows said:

But with the J-test, the spectrum generally appears perfect, even with a relatively "standard" XO...  

 

I'd be happy to see first DAC with perfect J-test figure! Please post!

 

So far, almost all problems I've seen are related to something else than the clock module itself. 99% of the problems are related to design of the surrounding circuits or the PCB layout (or just the PCB materials). So I'd rather spend the efforts improving where the problems are (PCB layout design and surrounding circuit design), rather than messing around with a component that has very minimal effect on the performance compared to aforementioned aspects.

 

Higher the clock frequency is, worse it gets anyway. So better to use 22.5792/24.576 MHz clocks with DSD512 without clock dividers (there are such DACs)! If you go for clocks near 50 MHz you already suffer about 10 dB in phase noise. Let alone clocks near 100 MHz needed for ESS Sabre. You could even optimize and use 11~12 MHz clock and DSD256 and still have flat noise floor up to over 100 kHz.

 

2 hours ago, barrows said:

I wish I had access to an analyzer which can measure the phase noise, it would be very interesting to measure XO phase noise at the input pin of the DAC (chip), but the expense of that tool is not justified at this point.

 

No reason to bother with such, because it would be irrelevant. Only relevant thing is what comes out of the DAC's analog output. Even if you have perfect clock in the clock input pin, but if you have USB packet ticking and a bit of extra thermal noise leaking to the reference voltage pin the entire result is still spoiled.

 

2 hours ago, barrows said:

One thing I saw recently, on a not to be named audio review site, was a very expensive Ethernet switch a company was selling, which included an "upgraded" OCXO.

 

I have to admit I never understood why mess with such in first place. If you use for example HQPlayer with NAA, or just a generic UPnP Renderer, the network traffic doesn't carry any clock information what so ever. When you listen to Tidal streams the jitter over internet is already in several milliseconds range, and still it doesn't matter at all. NAA can handle jitter up to about +- 500 milliseconds.

 

2 hours ago, barrows said:

I can have the J-test measurements you suggest made with different XOs, but the AP is in Florida and I am in CO so it takes some time.  I may get the AP sent here if/when I get an even better clock for comparisons (Pulsar).

 

Did you already mod the clock in AP too?

 

[barrows]

11 minutes ago, Miska said:

No reason to bother with such, because it would be irrelevant. Only relevant thing is what comes out of the DAC's analog output. Even if you have perfect clock in the clock input pin, but if you have USB packet ticking and a bit of extra thermal noise leaking to the reference voltage pin the entire result is still spoiled.

Well, i would say we disagree on this.  And checking for noise on the DAC's voltage reference is pretty easy, so easy to rule out as a source of problems.  USB packet noise is 8 kHz, I do not see it, there is pretty good isolation from that noise in this DAC with dedicated ADM 715x regs for each DAC Vin, and isolation between USB receiver and DAC I2S input.

BTW, I am not modding commercial DACs with alternate XOs, I am building DIY DACs and evaluating different XOs with the same implementation (for clock distribution).

As for "perfect" J-test results with regards to jitter spectrum, OK, perfect is in quotes!  But take a look at the Stereophile measurements of commercial DACs, these days a lot of them have virtually perfect results.  And a lot of these days have far from "perfect" XOs, or XO implementations.  That said, I still find audible differences going from, for example, a Crystek 575 or 957, to a NDK SDA series XO (not the SD series)(and depends on XO frequency, of course) in the same implementation.  Have not measured on the AP (yet as mentioned) but i highly suspect I would see virtually no difference on J-test.

[AudioBang]

If anyone is willing to travel to the central Connecticut  area I'd be happy to talk/listen audio for an hour or two .

Bring your own DAC if you want. I think it could be a great experience to meet and share...

 

Krell Evolution I mono amps - custom noise absorbing power feeds and DC Blockers, Gold Bybees on load side of power bridge rectifier

Krell Evolution II Dual mono preamps with EI-core Ultra-ISO transformers [J&K Audio-Singapore] replacing the factory torroids

Dunalvy VI with rebuilt crossovers using Duelund CAST and RF filters across each driver

Antipodes DX3 Music Server

Vega Dac with separate EI-core Ultra ISO transformers [J&K Audio] replacing stock torroid, SIC Schottky diodes, Pulsar clock replacing Crystek CCHD-575 [elevated about 5mm from same circuit board pads], Sparkos regs

Assemblage DAC powered from batteries w/Matrix USB SPDIF 2 and Uptone ISO regen (battery powered)

240VAC dedicated 10AWG lines to amps and front end - silver contact pasted on all connections back to the main breaker

240 to 120V balanced EI-core Ultra-ISO trans [J&K Audio] to each component before amps 

Analysis plus Silver Apex Balanced  interconnects - pin 1 open at receive ends, grounded through Gold Bybee Filter at source ends

Analysis Plus Big Silver Oval speaker cable

TP Link Ethernet to Antipodes fiber optically isolated from main router and powered from dedicated balanced EI core transformer.

 

Rich M

Avon, CT

 

 

[barrows]

I'll be in CT over the holidays, I know many may not have time for audiophilia then (I might not!) but I would be interested in checking out your stuff, shoot me a PM if you would like.

[Miska]

19 minutes ago, barrows said:

Well, i would say we disagree on this.  And checking for noise on the DAC's voltage reference is pretty easy, so easy to rule out as a source of problems.  USB packet noise is 8 kHz, I do not see it, there is pretty good isolation from that noise in this DAC with dedicated ADM 715x regs for each DAC Vin, and isolation between USB receiver and DAC I2S input.

BTW, I am not modding commercial DACs with alternate XOs, I am building DIY DACs and evaluating different XOs with the same implementation (for clock distribution).

 

Checking the actual DAC analog output is easiest and the only relevant thing. It tells the actual facts. Because there can be also poor/buggy chip designs...

 

I cannot comment on anybody's DIY DAC, only on commercial DACs. And the weakest point is usually not the clock module. And I have a lot of doubts about modifying commercial DACs. Replacing a clock in tightly packed SMD board of a commercial DAC is not straightforward if it is not exact footprint match. And even if it is, it could have some characteristic differences not taken into account in the original design. Blindly going and replacing a clock module to a "better one" on a commercial DAC and hoping it'll improve things is likely not going really do that.

 

And there are other funny things, like for example ESS' wandering noise bumps around the noise floor likely coming from the ASRC. These are actually hard to catch because they are time-varying and very low level.

 

19 minutes ago, barrows said:

As for "perfect" J-test results with regards to jitter spectrum, OK, perfect is in quotes!  But take a look at the Stereophile measurements of commercial DACs, these days a lot of them have virtually perfect results.

 

I always read though all of them, but somehow I haven't yet spotted any perfect result. I'm not interested in 16-bit J-test results because that hides a lot. 24-bit is the relevant one. Of course I don't read JA's comments, just what I see myself. I also was checking through all the HiFi-News (UK magazine) results when PM has been publishing the results. And of course I measure all the gear I happen to get my hands on.

 

[barrows]

33 minutes ago, Miska said:

And there are other funny things, like for example ESS' wandering noise bumps around the noise floor likely coming from the ASRC. These are actually hard to catch because they are time-varying and very low level.

I use ESS chips but the ASRC is not running.  Also I reduce variables as much as possible by having same clock distribution when evaluating different clocks and there is only 1 clock running (on the isolated side) of the DAC for 44.1 base sample rates only.  My understanding is that with the 9028/9038 designs they have eliminated much of the low level artifacts you mention although trying to figure this out from what ESS actually makes public (or even in the datasheets) is difficult.

Also I have DSC-2 balanced in progress, but not fully implemented (yet), very interested in this approach and your design.

 

But of course, in general, you are entirely correct that there are many aspects of any design which can be screwing up results and leading to wrong conclusions.  But, I have heard too many differences which were obvious, repeatable, and without question occurring, which do not show up in a standard set of measurements taken at the analog outputs of a DAC to agree that one can learn everything just from that measurement.  Do you not hear differences in various HQPlayer filters which produce the "same" analog output (in the audible bandwidth) in standard measurements?  Just curious. 

[Miska]

54 minutes ago, barrows said:

Do you not hear differences in various HQPlayer filters which produce the "same" analog output (in the audible bandwidth) in standard measurements?  Just curious. 

 

I have no problem analyzing/measuring differences between HQPlayer filters and modulators...  It is not right/wrong, just different.

 

It is not something that would elude measurements but be audible subjectively. Sometimes the differences are easier to analyze than to hear.

 

My design flow is such that I do objective evaluation first and only once it has passed, I start to listen. I don't even want to listen to something that doesn't look as expected on analysis.

[barrows]

1 hour ago, Miska said:

I have no problem analyzing/measuring differences between HQPlayer filters and modulators...  It is not right/wrong, just different.

You have measured differences within the range which would be audible?  That is <20kHz and above -110 dB?  Really?  I mean, I get that an early rolloff filter could be considered a measurable/audible difference (although not theoretically for my hearing)-but what about minimum phase vs. linear, for example...

 

[mansr]

3 minutes ago, barrows said:

You have measured differences within the range which would be audible?  That is <20kHz and above -110 dB?  Really?

Making a filter/modulator that causes such differences is trivial. Then again, so is making one that doesn't.

[barrows]

2 minutes ago, mansr said:

Then again, so is making one that doesn't

That is what i mean.  HQPlayer has many filter options, some of which could be shown in measurements restricted to the audible bandwidth O(and level) to have differences and some not, yet many have their favorites.