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Review and Measurements of Benchmark AHB2 Amp

RichB

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Would I really notice a subjective difference going from my Bryston 3B-SST to the Benchmark (other than less heat)? Going from the Hypex NC400 to the Benchmark?

I like the different gain settings, they make a direct connection from a DAC with digital volume much easier.

There is a 30 day trial, so it will cost you shipping to find out.

- Rich
 

DonH56

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Would I really notice a subjective difference going from my Bryston 3B-SST to the Benchmark (other than less heat)? Going from the Hypex NC400 to the Benchmark?

I like the different gain settings, they make a direct connection from a DAC with digital volume much easier.

Maybe not but too many variables, like speakers, listening habits, etc.
 

tktran303

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They were voltage matched while driving the M20's with quick switching between amps.

Of course, neither (subjectively) sound like the AT4002:

https://www.audioholics.com/amplifier-reviews/ati-at4002

Also level matched.

- Rich

Hi Rich,

The speaker measurements are taken too close to the tweeter. (as shown by the null around 2.5KHz)

To show how the FR varies for the Revel M20, you should take measurements out in far field, so that tweeter and woofer sum correctly.

The correct minimum distance is 3-5 times the radiating diameter of the mid woofer, and at least 2 times the width of the speaker cabinet, whichever is the greater (so you capture the baffle step).

1 metre is the traditional distance for far field measurements, because it can also give you a sensitivity rating, and also gives a chance for the drive units to integrate (again avoid the null)
 
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John_Siau

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Wouldn't the wires that run closer introduce increased capacitance leading to a similar detriment?

I have some vintage 10AWG Monster speaker cable I planned to utilize, it must have awful inductance, but as I recently found I'm now deaf above 14kHz I should have little reason to worry about it.
The capacitance is not an issue. Capacitance forms an RC lowpass filter where R is the output impedance of the amplifier. C will be a few pF per foot (multiply this times the cable length to get the total cable capacitance). The Canare cable that we sell is 45 pF per foot. The output impedance of the amplifier can be determined from the damping factor if it is not specified directly. Use the damping factor at 20 kHz if this is available. The AHB2 has a damping factor of 350 at 20 Hz and 34 at 20 kHz. At a damping factor of 34, the output impedance would be 8/34 = 0.235 Ohms. The total cable capacitance will be far too low to have any impact when driven by such a low impedance. Series resistance and inductance are the important parameters with speaker cable. Inductance becomes a major issue with cables over 100 feet long.

Here is the calculation for 100 feet of Canare 4S11 Star Quad speaker cable driven by an AHB2 power amplifier:

R = amplifier output impedance at 20 kHz = 0.235 Ohms
C = 45 pF/foot X 100 feet = 4500 pF

f = 1/(2πRC) = 1/(2*3.14*0.235*4500E-12)= 150 MHz

Here is a tool for calculating RC low-pass filters:
http://sim.okawa-denshi.jp/en/CRtool.php

Our calculation shows that the cable capacitance will cause a -3 dB roll off at 150 MHz. Yes 150 megahertz! This number shows how insignificant the capacitance of speaker cable is in most applications! Our calculation was for a 100-foot cable. At 10 feet this cutoff frequency would be 10 times higher. Forget about speaker cable capacitance. It just doesn't matter.

edited - Thanks to DonH56 for catching my math error.
 
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DonH56

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It's "only" 150 MHz, not GHz... Still higher than most audiophiles are likely to hear.
 

777

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Martin Logan Neolith has 0.43 ohm at 20khz. In this case the damping factor will be 0.43/0.235=1.8 :) The capacitance of speakers cables is irrelevant indeed but what about speaker capacitance ? How AHB2 behave with speaker capacitance like 1uF or 2uF ?
 

D700

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I've read through the literature and am a little confused on the balanced versus unbalance. I get that XLR are better...what exactly do I lose using RCA>XLR cables with a source that only has RCA out?
 

John_Siau

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What about 3-ohms @John_Siau ? As there are often speakers that dip below 4 ohms for certain small ranges of frequencies.

Is it possible for @amirm to test at 3 ohms?
The amplifier will stay clean when driving very low impedances. The published rating is the long-term continuous output power and this is a function of heat sink area. Driving 3-Ohm, 2-Ohm or 1-Ohm impedance dips are not a problem. What is unique about the AHB2 is that is stays clean when driving these low impedances. There is virtually no rise in THD as load impedance drops. No other amplifier does this so well. Most amplifiers show a significant increase in THD when driving low impedances. In contrast, the feed-forward correction system in the AHB2 keeps the output distortion free.

Yes it is OK to test at 3, 2 and 1-Ohm loads. You cannot do damage to the AHB2 amplifier by very low load impedances or short circuits. It is fully protected and it will shut itself down before you can cause damage. You can even short it out while it is playing at full output.
 

John_Siau

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I've read through the literature and am a little confused on the balanced versus unbalance. I get that XLR are better...what exactly do I lose using RCA>XLR cables with a source that only has RCA out?
It is not about what you lose it is about what you gain by using an RCA to XLR adapter cable instead of having an RCA input on the AHB2.

Our adapter cables wire the XLR+ back to the RCA pin. The XLR- is wired to the RCA shield. A separate shield conductor connect the AHB2 chassis tot the RCA shield. This wiring allows the use of the balanced differential amplifier at the XLR input. This wiring provides significant rejection of ground-loop noise. An RCA to RCA connection provides no rejection of ground-loop noise.

The best solution is always balanced to balanced. Here is an application note that I wrote on this topic:

https://benchmarkmedia.com/blogs/application_notes/balanced-vs-unbalanced-analog-interfaces
 

John_Siau

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There is time and effort involved in sourcing, ordering the parts in addition to building them. That's not free when I could spend that time doing something else I enjoy. There are excesses in audio but this one is not it.
The people who make the cables live and work in Syracuse NY and they don't just make minimum wage. You can build them for 1/2 of the price if you have the time, skills and tools. You can probably get something from China for less than you would pay for materials. Connectors would be Chinese knockoffs, cable would be different. The biggest issue is the quality of the connectors.
 

Veri

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It is not about what you lose it is about what you gain by using an RCA to XLR adapter cable instead of having an RCA input on the AHB2.

Our adapter cables wire the XLR+ back to the RCA pin. The XLR- is wired to the RCA shield. A separate shield conductor connect the AHB2 chassis tot the RCA shield. This wiring allows the use of the balanced differential amplifier at the XLR input. This wiring provides significant rejection of ground-loop noise. An RCA to RCA connection provides no rejection of ground-loop noise.

The best solution is always balanced to balanced. Here is an application note that I wrote on this topic:

https://benchmarkmedia.com/blogs/application_notes/balanced-vs-unbalanced-analog-interfaces
Makes so much sense actually :) well done.
 

SEKLEM

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The amplifier will stay clean when driving very low impedances. The published rating is the long-term continuous output power and this is a function of heat sink area. Driving 3-Ohm, 2-Ohm or 1-Ohm impedance dips are not a problem. What is unique about the AHB2 is that is stays clean when driving these low impedances. There is virtually no rise in THD as load impedance drops. No other amplifier does this so well. Most amplifiers show a significant increase in THD when driving low impedances. In contrast, the feed-forward correction system in the AHB2 keeps the output distortion free.

Yes it is OK to test at 3, 2 and 1-Ohm loads. You cannot do damage to the AHB2 amplifier by very low load impedances or short circuits. It is fully protected and it will shut itself down before you can cause damage. You can even short it out while it is playing at full output.

All I can say is "Neat". I don't have a vast technical knowledge of how amplifiers work, but this sounds impressive. Does Benchmark have an eye to make any more power amplifiers or integrated amplifiers at the moment?
 

D700

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Our adapter cables wire the XLR+ back to the RCA pin. The XLR- is wired to the RCA shield. A separate shield conductor connect the AHB2 chassis tot the RCA shield. This wiring allows the use of the balanced differential amplifier at the XLR input. This wiring provides significant rejection of ground-loop noise. An RCA to RCA connection provides no rejection of ground-loop noise.
That's an elegant approach and a very clear answer. Thank you.
 

John_Siau

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Really interesting post, thanks for sharing. I guess you're referring to "Near field HD spectrum (20 mm distance)" from https://hificompass.com/en/speakers/measurements/accuton/accuton-s280-6-283n. I should probably give it a try myself too, but @RayDunzl is the man for such measurement, in case he haven't done it already. It would be nice to check H2...H5 for 90dB at 2-3 meters in front of speakers (listening point), of course...sinewaves (-1dBFS maybe).

Thank you!
There is no disagreement. I am confident that there is a logical, scientific and real-world reason for what I have heard and reported. However, so far, I do not know what it is. This is one reason why I participate in forums such as this and appreciate posts such as the one above from josh358.
The differences can be measured and the measurements often imply that the differences can be heard.

Whenever an audio product produces a distortion component that is reproduced at a level above the threshold of hearing, there is a chance that it will be heard. The reality is that most amplifiers produce distortion at levels that are significantly above 0 dB SPL in a typical system. In contrast, the distortion produced by the AHB2 is at or below 0 dB SPL (do the math if you doubt this statement).

The sound of an amplifier's distortion is not masked by room noise. We can hear tones that are as much as 30 dB lower than the ambient noise.

An amplifier's distortion may also not be masked by the music, especially when the distortion is separated from the fundamental by several octaves. In musical signals, low frequencies normally have much higher voltage swings than high frequencies. The high-order harmonics produced by bass content can reach levels that are comparable to high-frequency musical content. How loud is that flute that you hear and how loud are the harmonic cues that tell you that it is a flute?

Zero crossing distortion produces many high-order harmonics and it is most audible when playing at low levels.

IMD is particularly problematic because it does not resemble the harmonics that are produced by musical instruments. With IMD, the distortion does not occur at harmonic frequencies and it may be well separated from any musical content that would mask it. Many class-D amplifiers have IMD problems. Class-AB amplifiers can produce significant IMD when they have zero-crossing distortion. They can also produce significant levels of IMD when presented with a pair of high-frequency tones.

The distortion produced by electronics is different than the distortion produced by speakers. The differences in the spectrum can be seen on an FFT. Electronics are more prone to creating high-order harmonics which are not well masked.

Electronics produce harmonic distortion at perfect integer-related frequency ratios. Musical instruments do not. For example, the piano produces harmonics that are a bit further apart than integer ratios. The tuning of a piano is normally stretched to compensate for this non-integer harmonic spacing. The overtones produced by a piano string will beat against the integer-ratio harmonic distortion produced by electronics. A "warm-sounding" amplifier will make a piano sound like it is out of tune.

Harmonic distortion may change the sound of a musical instrument long before we can recognize the fact that the music is distorted. The ratios of harmonics to the fundamental give each musical instrument its unique voice. Any time you add harmonic distortion, you make changes to this voicing.

A non-linear phase response may be more audible than a non-linear frequency response. Errors in the phase response may create the impression that the frequency response is different. The frequency response of the AHB2 extends down to 0.1 HZ and up to 500 kHz so that the phase response is linear within the 20 Hz to 20 kHz audible band.

Many amplifiers distort when driving the low-impedance portions of the speaker's impedance vs. frequency curve. This doesn't show up in steady-state 8-Ohm and 4-Ohm tests. It is important to look at the distortion when driving very low impedances. The AHB2 stays clean when driving difficult loads and phase angles. In contrast, most amplifiers do very poorly into these difficult loads.

Here is an application note about a double-blind ABX listening test that we did between the AHB2 and a typical class-AB amplifier with decent specifications. This test was examining the audibility of zero-crossing distortion when playing a single tone through loudspeakers at a low level (0.01 watt producing 67 dB SPL at the listening position). It was very easy to hear the difference between the two amplifiers. I scored 25 correct out of 25 trials on my first attempt.

ABX listening test:

https://benchmarkmedia.com/blogs/application_notes/power-amplifiers-the-importance-of-the-first-watt
 

GrimSurfer

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Thank you very much for your last post, John. It clearly describes the differences in various acoustic and electronic phenomenon that can define an amplifier's "signature sound" (understanding that the goal should be to minimize or eliminate this altogether).

It would be an understatement to say how much informed audio buffs appreciate the performance of your products. Benchmark components offer, by all measures I've seen, truly exceptional performance.
 

John_Siau

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Try oversampling with different reconstruction filters in HQAudio Player. You'll be surprised at how some reconstruction filters bring out small details, and others seem to suppress them. I certainly was. And you'll really hear this difference between DAC's as well, as opposed to just the chip. Forex, closed form filters as in the Schiit DAC's or in HQAudio player seem to accentuate small details. I believe Schiit attributes this on their site to the fact that time domain performance of their closed-form filter isn't stochastic.

This stuff is endlessly fascinating. DACs sound far more different than they should, given their specs. In some cases, it's easy to see why -- poor jitter rejection, for example, gives some DAC's a "gray" coloration, or you have issues like departure from monotonicity, or they intentionally use underdamped filters with overshoot (ESS in at least some of their chips). In other cases, it isn't so easy. But while the difference between good amplifiers is subtle, I've found that DAC's can sound dramatically different. I mean, tone-control-level different, despite the fact that they're flat within a fraction of a dB. Comparing them has been a real ear opener.
Our ears are relatively insensitive to phase response on steady-state combinations of a fundamental and a series of harmonics. However, we are quite sensitive to phase response on transients. If the high-frequencies arrive first, they are accentuated. If the high frequencies arrive late, the impression will be that they are rolled off. An impulse, such as drum rim shot produces a broad spectrum of frequencies. A perfect impulse produces all frequencies simultaneously. The sound of this impulse will change when the phase response is changed. Many D/A converters use filters with a non-linear phase response. These filters change the way an impulse sounds. The changes will be most noticeable in the percussive content of the music. It has become popular to manipulate the sound with non-linear filters. These filters are an effect and some people enjoy having these effects applied to their music upon playback. The down side is that these effects are cumulative. Each pass through a non-linear phase filter will add more audible effects. In contrast, many A/D and D/A processes can be cascaded when the converters have a linear phase response. We have demonstrated this with some in-house listening tests that we conducted with cascaded converters.

The differences can be measured if you run the right tests. The differences will not show up in the frequency response (amplitude response).

Beware of time-domain response plots because they often look at only one phase relationship to the sample clock. An "improved" filter response may not look so good at other phase relationships to the sample clock. Filters that allow aliasing will pull transients toward the nearest sample clock edge (distorting the transient).
 

John_Siau

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I find that the difference between good amplifiers is subtle, and while some things stick out right away (the Parasound, for example, has very rich midbass), others are subtle and require long-term listening (forex, a cymbal on the Parasound sounds like noise, while on the AHB2, it sounds like, well, a cymbal -- it rings).
This is an interesting observation. MP3 compression destroys the phase relationships between fundamental tones and harmonics. When this processing is applied to music, percussive sounds tend to become bursts of noise instead of identifiable actions. I find that this is the most disturbing artifact produced by MP3 compression. You seem to be describing a somewhat similar effect. Accurate phase response is very important when reproducing transients.
 

maty

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Other day, I extracted the following from a wikipedia article

[ Overview

The advantages of DMM (hard surface material) over acetate lacquer cutting (soft surface material) are both sonic and practical: because of the rigidity of the master disc medium, no groove wall bounce-back effects take place after the cutting has been completed. This preserves the original modulation details in the groove walls much better, especially those involved with sudden fast attacks (transients).

The improved transient response, as well as the more linear phase response of DMM improve the overall stability and depth-of-field in the stereo image. In addition, disturbing adjacent groove print-through sounds (groove echoes) are reduced in DMM.

Also, there is no need to rush the finalized master disc directly into a refrigerator for groove preservation, as in conventional lacquer disc cutting, before processing the master disc to produce matrices for the pressing of the records... ]

@John Siau

Do you agree with the second pragraph?
 

RichB

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Would I really notice a subjective difference going from my Bryston 3B-SST to the Benchmark (other than less heat)? Going from the Hypex NC400 to the Benchmark?

I like the different gain settings, they make a direct connection from a DAC with digital volume much easier.

Listening to 2-channel music, there is more apparent detail than the AT6002 and AT522NC that I have directly compared.
This has been stated in many reviews. It isn't that the detail is not present with other amps, but individual sounds were not as easily recognized.
I suppose this is a byproduct of vanishing low distortion and accurate phase response.

Last night, I watched Serenity again with the AHB2's amplifying my 5-channel system. The soundtrack is excellent. Most notably, the ambient sounds produced an amazing surround experience.

Another advantage is the heat reduction that has permitted the disposal the temperature-controlled cooling fans. The ATI's produced 320 watts idling versus the 80 watts for the 4 AHB2's. The AT522NC idle at 38 watts, about double the AHB2.

- Rich
 
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