Sutherland KC Vibe MK2 Phono Stage Review

[solderdude]

The high voltages are not for obtaining increased overload capabilities. Phono front ends using parallelled MOSFETs or BJTs will operate at high voltages to run the devices in their most linear area.

This pre-amp has regular opamps in it.
The high voltage is internally 'substantially lowered' acc. to the manufacturer.
By the looks of it using 2 zeners ?

Ticks and pops may or may not be better served with less of a swing in any case, depending on how the stage clips and recovers when overloaded.

Yep, that's what I said.

 

[solderdude]

This is what i was wondering about when people discuss headroom and the clicks becoming much more audible when clipping (note that i have no idea about these things, but somehow the case solderdude mentions also seemed logical). I would appreciate further discussion or explanation for noobs. Thanks.

Opamps can have terrible overload errors with so-called rail sticking. In such case the output voltage 'sticks' to the voltage rail longer than it should be and thereby effectively elongating (widening) the input signal. On top of that, after the rail sticking it takes some time for the output to reach/follow the input signal again.
This will make a 'tick' wider and thus lower frequency and more audible.

 

[sarumbear]

They put more value on giving dimensions of the device than its electrical characteristics!

This!

 

[sarumbear]

Probably would have needed to buy the full rack $2000+ version which was battery powered to get the good headroom. The Southerlands are nice pieces, and if you're already buying an audiophile turntable, high end arm, and an MC cartridge, an additional $1-2 for an accompanying phone stage is not an unreasonable expenditure. If you're into vinyl, that is. I, for one, no longer am.

Have you not understood the issue here or you are just going with the manufacturers "vibes"? How do you expect a battery operated amplifier be better than the one that is operated at 48VDC?

 

[sarumbear]

Interesting company. I never heard of them but I'd like to see reviews of the products higher up in their lineup. The Vibe MK2 is their entry level product. Their top of the line model is only $4000--which for this type of product is reasonable if it performs significantly better.

How will you know if it is better or worse? Also, what is your criteria to judge a phone amplifier's quality? I really like to know.

 

[Cars-N-Cans]

Ticks and pops may or may not be better served with less of a swing in any case, depending on how the stage clips and recovers when overloaded. Having a massive swing available just means your clicks and pops are less distorted and have a greater amplitude- something that is not necessarily better. I have preamps with MM stage overloads of >320mV@1kHz and capable of 20V out (+/-32V rails). IOW, they are practically impossible to overload with a cartridge, but you'd better have a sensible person on the volume control.

For the prices they charge for these you would think they would include sensible limiting circuitry. The two options seem to be just let the phono stage slam up against one its rails, or arguably worse, sail away and drive the speaker amplifier into hard clipping, which is likely to be the slowest thing there if it’s a linear, and this assumes it doesn’t have protection circuitry that will try and intervene first. Much nicer would be to just clip the transient to some sensible level that keeps all the circuitry between the cartridge and the speakers in it’s normal operating region. But instead it’s just the bare minimum circuit with some of these more niche ones, which look no different than stuff I roll myself. Ok for me, but not too swift for a commercial product with any random person as it’s potential end-user.

Edit: Come to think of it, there are all sorts of nifty things you could do, esp. if one were to internally digitize the signal from the cartridge. Then you could do things like simply remove the noise, or even include predefined “noise” samples like random, but pleasing pops and clicks that could be inserted in at that location once the digitized waveform has had the offending impulse or noise removed. But, I guess all the “budget” goes to the creative buzzwords and farcical claims in the marketing material instead. I mean yeah there is software processing that does this, but it would be a nice change of pace to have some real time processing built in to at least justify the cost.

 

[AudioSceptic]

As a phono stage has a fixed gain (when in use), there's no practical difference between input or output overload, operationally, the phono stage just overloads at a certain point. In this case, at 26mV, so inadequate for some, but not all LPs. I have a PPM on the output of my phono stages, and most of my LPs stay within 12dB or so of 5cm/sec, so this stage would just be OK for most MM cartridges. However, that's just for modulation, scratches and dirt causes clicks which can easily exceed the peak modulation, as so could overload the stage. That turns a click into a splat, much more audible.

If a cheaper stage like the ProJect, or CA can have much better overload characteristics, if phono stages 30 or 40 years ago could, why can't an expensive modern phono stage do too?

S

Agreed that it makes no difference overall whether the input or output overloads.

You say "within 12dB or so of 5cm/sec". That would be only about 20 cm/s, which seems low compared with Shure's figures. Do you have a feel for how large scratch/click peaks are? Higher than the 70 cm/s peak modulation that Shure found? I was thinking of measuring levels and wondered if a DMM would suffice (I would play a 12" 45 as they seem to have the highest levels), so I'd like to know more about your PPM (peak programme meter?): cost, spec., etc.

In any case, there's no excuse for the poor overload performance of many current phono stages, and especially any costing more than a couple of hundred £/$.

 

[restorer-john]

This will make a 'tick' wider and thus lower frequency and more audible.

Well, let’s test some transient overload characteristics of phono stages shall we?

Standard EIA toneburst over max rated input or…what do you suggest?

Overload at 1kHz is the standard for RIAA stages.

 

[anmpr1]

I don't know about the other stuff, but a Phillips head screw to the chassis for a grounding point really speaks to high tech sophisticated ergonomics. Last time I encountered that sort of thing was on a Dyna PAT-4. But the Dyna only had a slotted head screw, so for your thousand dollars Sutherland gives you one more slot for your screw driver.

 

[sarumbear]

Come to think of it, there are all sorts of nifty things you could do, esp. if one were to internally digitize the signal from the cartridge.

Once you digitise you can analyse the waveform. Ticks have fast rise and fall times. In music almost always the fall times are slow due to reverberation, acoustic or artificial. It is perfectly possible to detect such fast decaying pulses (ticks) and limit their amplitude without using a general limiter. Such process will not affect the music and should be transparent.

There are various audio software that does that as an offline process but I have not heard any hardware that does it in real time. If I will be paying 3+ digits I demand this.

PS. Come to think of it, the DSP in various digital processors should be able to do the job.

 

[sergeauckland]

Agreed that it makes no difference overall whether the input or output overloads.

You say "within 12dB or so of 5cm/sec". That would be only about 20 cm/s, which seems low compared with Shure's figures. Do you have a feel on how large scratch/click peaks are? Higher than the 70 cm/s peak modulation that Shure found? I was thinking of measuring levels and wondered if a DMM would suffice (I would play a 12" 45 as they seem to have the highest levels), so I'd like to know more about your PPM (peak programme meter?): cost, spec., etc.

In any case, there's no excuse for the poor overload performance of many current phono stages, and especially any costing more than a couple of hundred £/$.

Shure's figure of 70 were for the outliers, the main bulk of their measurements was clustered around 10-20cm/sec. That stacks up with what my PPM is indicating on most of my LPs. The PPM I use is a DIY design, using the display of a Velleman VU indicator with the rectifier and time constants changed to match the Type 1 PPM spec. This has a faster rise time than the Type 2 PPM in general use in the UK, and is the more commonly used one in Europe. It is still a quasi-peak indicator rather than true-peak, so will underindicate on very short peaks, especially at high frequencies.

I've set the output level of my phono stages to give 0dBu (0.775v) out at 5cm/sec recorded velocity at 1kHz, and my PPM goes to full scale at +15dBu, and only a few LPs ever get to that level, most peak at around +10dBu, so something around 15cm/sec. It does go to show, however, that unlike digital that has a fixed 0dBFS maximum, LPs can be cut at whatever level the cutting engineer decided was a good compromise between playing time, noise and distortion. I have a few compilation LPs, Sounds of the '60s, that sort of thing, that cram 30 minutes playing time onto one side, and the level is quite low, with the corresponding noise quite high, but it doesn't matter much considering the compressed nature of the music.

As there's no standard for maximum cutting levels, and given Shure's measurements, an absolute minimum of 20dB overload would be indicated as necessary, with 25dB preferable.

S.

 

[sergeauckland]

Once you digitise you can analyse the waveform. Ticks have fast rise and fall times. In music almost always the fall times are slow due to reverberation, acoustic or artificial. It is perfectly possible to detect such fast decaying pulses (ticks) and limit their amplitude without using a general limiter. Such process will not affect the music and should be transparent.

There are various audio software that does that as an offline process but I have not heard any hardware that does it in real time. If I will be paying 3+ digits I demand this.

PS. Come to think of it, the DSP in various digital processors should be able to do the job.

The Garrard Music Recovery Module was one such hardware box. As I recall, it used an analogue delay line and threshold detector to cut out the clicks. I've never heard one in operation, but they never particularly caught on.
S.

 

[anmpr1]

I wonder why these often expensive phono stages don't offer a switchable low cut filter? That used to be standard in the heyday of LP records. It can't be because of expense, for what they charge.

IMO anyone seriously interested in phono reproduction, especially if they are collecting older records, should have a preamp with a mono (A+B) switch, a separate A and B switch is often handy, plus a balance control, in addition to the low filter. Channel reverse could be helpful for troubleshooting some problems. Not many modern preamps have those features, anymore.

 

[restorer-john]

The Garrard Music Recovery Module was one such hardware box. As I recall, it used an analogue delay line and threshold detector to cut out the clicks. I've never heard one in operation, but they never particularly caught on.
S.

I have an SAE-5000 here...

Tell me, what do you wanna see?

This is the bypass, vs in-line, zero setting. Notice the infrasonic filter, but an awesome flat response in general.

 

[AudioSceptic]

I don't know about the other stuff, but a Phillips head screw to the chassis for a grounding point really speaks to high tech sophisticated ergonomics. Last time I encountered that sort of thing was on a Dyna PAT-4. But the Dyna only had a slotted head screw, so for your thousand dollars Sutherland gives you one more slot for your screw driver.

View attachment 234117

Effective, but cheap or what?!

 

[restorer-john]

forum is going a little bit nuts. Duplicating posts etc...

 

[dachada]

The used opamps are +/- 18V max.
The input voltage = 48V and this is +/-24V.
There is at least 6V available for regulation but maybe even more.



The reason for the rather low dynamic headroom is the method used.
Passive RIAA correction, which some believe is the 'best sounding' method.
With active RIAA the gain varies with frequency. Passive ones have a huge gain (full BW).
This basically means the highs have the largest amplitude (that's why RIAA is needed) and thus these clip first.
Below the basics for this type of RIAA.

When the overload character of the used opamps is 'nice' ticks that far exceed the max. input level are actually less audible (as they are limited in amplitude by the clipping) than when there would be a huge headroom.

The build costs of this amp are below $ 200.- when built in small quantities so that's quite a markup.

I Use the same design with my custom Phone Pre. The A1 has a dual JFET PN4393 amp and the A2 has a very low noise/low distortion opamp. with a dual 15V PS i have very nice dynamic headroom

 

[solderdude]

Well, let’s test some transient overload characteristics of phono stages shall we?

Standard EIA toneburst over max rated input or…what do you suggest?

Overload at 1kHz is the standard for RIAA stages.

One could ask @amirm to introduce a needle pulse (through inverse RIAA) and drive it to a certain level.
I don't think he is going to implement such a test as it isn't a standard test and there will be discussions about levels.
Perhaps a high voltage needle pulse + 1kHz sine (at normal level), or better yet... a continuous sine and in level increasing needle pulses ?

Personally I am not fussed about vinyl as vinyl is highly subjective oriented anyway and not about signal fidelity but rather about 'sound' and synergy between cartridge, tone arm, load (pre-amp), RIAA correction etc.

 

[anmpr1]

Effective, but cheap or what?!

How about a knurled knob? I have three or four phono stages hanging around. None at a thousand dollars. And all but a Dyna uses a knurled knob for a grounding point. I think a couple of them even have a hole in the post for bare wire. Very easy to manipulate.

Sure, it's a little thing. But for a Grover Cleveland, I expect something a little more user friendly than a Phillips head screw direct into the chassis. That exudes cheapness. And I'm guessing they don't even throw in a screwdriver for that price!

 

[anmpr1]

The Garrard Music Recovery Module was one such hardware box. As I recall, it used an analogue delay line and threshold detector to cut out the clicks. I've never heard one in operation, but they never particularly caught on.
S.

Packburn was the professional's choice. I am surprised to find they are still around. Stanton 500 on the home page!

Packburn Electronics

The Finest Noise Suppression Hardware Available

www.packburnelectronics.com