Review and Measurements of Benchmark AHB2 Amp

[helom]

It seems the buzz is due to some less-than-ideal synergy between my Yamaha and the Benchmark. I tried my DAC/preamp and it was silent.

Anyone here using the AHB2/ LA4 combo?

 

[GXAlan]

Last night I spent a couple hours powering my Maggie .7s with my single AHB2. I played a wide variety of tracks, at SPLs averaging in the mid to high 80s, with dynamic peaks around 95b, which is plenty loud for most listeners and louder than I typically listen. I sit ~9 feet/ 3 meters from the speakers.

The amplifier's clipping LED flashed only once, during the crescendo of a very dynamic orchestral recording.

What was the track? What were you other mono blocks? (I have MG-IIIA’s which are even less efficient.)

 

[helom]

What was the track? What were you other mono blocks? (I have MG-IIIA’s which are even less efficient.)

I have yet to sell my outgoing amps, so in the interest of them retaining value I prefer not to divulge the brand and model. Suffice it to say they are of a well-made, well-respected brand.

Track is 'La Campanella' - Concerto Per Violino E Orchestra In Si Minore No. 2 - III. from Diabolus In Musica - Accardo Interpreta Paganini

Clipped only during the largest crescendo of that recording.

I'm not familiar with MG-IIIAs. All I know is a single AHB2 provides enough current for my needs with the .7s.

 

[RichB]

It seems the buzz is due to some less-than-ideal synergy between my Yamaha and the Benchmark. I tried my DAC/preamp and it was silent.

Anyone here using the AHB2/ LA4 combo?

I have the LA4 driving a pair of AHB2's and they are absolutely quiet.
For 2-channel:
Gigabyte I3 running Roon
Oppo UDP-205 (via USB WASAPI or ASIO)
LA4
2 x AHB2's
2 x Salon2's

For HT:
HDMI Source: ATV4K, TiVo Bolt+, Oppo UDP-205,
Emotiva RMC-1
2 x AHB2's for Voice2 and Studio2's
LA4 Fixed volume driving the 2 AHB2, one for each Salon2.

In HT mode (RMC-1), I can maybe here something with my ear pressed to the tweeter. My daughter can.
With the RMC-1 off and LA4 on she cannot hear anything. So the source is the RMC-1 but negligible.
The LA4/AHB2 measurements confirm these are not the source and the RMC-1 is a very quiet HT processor.

All analog components are connected via XLR.

In the past and before the Benchmark gear, I had hiss.
Once from a faulty power connector on the AV8801. I could jiggle the power cord and change the hiss profile. United radio repaired twice, once for this and a blown trigger and another time for just a blown trigger.

Other hiss cases were cured by moving to balanced cables.

In any event, I would connect on the AHB2 to the speaker, then test with the LA4, then add a source component and observe when the hiss is introduced.

In the past, I also had buzz from the center channel and rears with Parasound A31, ATI AT3000. That went away with the ATI 6000 amps.
However, the ATI amps had audible transformer hum from my listening position.

The AHB2 has no transformer hum, no buzz, no hiss, runs cool, and is one of the cleanest amps on the planet.
I love them!

- Rich

 

[RichB]

I did some loud listening sessions yesterday with tracks I have played over the years.
I use the LA4 is used for two channel music.
The LA4 at +7 (low gain mode) clipped (clip indicator only, so not current limited) at +7.5 but seldom at +7.

This is not anywhere near where I listening today (while writing this email): -23 on the LA4.
What's that 1/1000'th the power? The Salon2's are basically 4 Ohms speakers so if they clipped at about 200 WPC, so today I am peaking at .2 watts

Last night, we snuck in a close family friend and watched 1917 UHD without masks. Horrors!
This is a different audio path using the RMC-1 and the LA4 suppling fixed gain to match the other channels.
At the RMC-1 volume -15, this movie was IMO theatrically loud. The first explosion in the tunnel made us all jump.
There was no clip indicator anytime during this film. I could have gone louder plenty of headroom to play this at uncomfortable levels.
With two channels playing the same tracks as above the RMC-1 clips the AHB2's at -10, so about 5 dB of headroom.

The AHB2's bi-amp the Salon2's and are not bridged (although, that option is available should I decide endanger my hearing).

When not bridged, I can clip the AHB2s if I try since the Salon2's can handle the power, but I just don't need it.
I could clip them bridged, but that would just shorten the useful life of my hearing.

I recommend those interested, try one. Want more, then buy another, if budget permits.

I suspect all critical listening occurs below the first watt and not when my ears are in loud mode.
After listening at +7, I can barely hear -23 Today, it is just fine.

- Rich

 

[GXAlan]

I'm not familiar with MG-IIIAs. All I know is a single AHB2 provides enough current for my needs with the .7s.

Well, you got the clipping light to turn on, so by definition, you'll need two AHB2's to run as bridged monoblocks to listen at the same level. The MG-III are a larger panel with true ribbons. They are less efficient. (https://www.stereophile.com/floorloudspeakers/868/index.html). They are great for vocals, but are hard to drive dynamically.

I did a quick analysis of the track. It's definitely got impressive dynamic range. In my software, I see average SPL at -26 dB with a peak sample at the end of -5.74 dB (and if you average 1 second, maybe -15 dB peaks). The average SPL includes the peaks, so if I just pick a 30 second range without peaks, it's recorded at -34 dB.

If we assume it's a 28 dB swing between average and peak levels, your 85 dB average is probably actually pushing 113 dB+ for a single peak which triggered your clipping even though it's probably closer to a 20 dB swing

SO, at 5W, you're getting ~87 dB.

At 190W, you're getting 103 dB. (need 106dB, or even 113 dB).
To get to 106 dB, you need 400W. (Assuming that the 0.7 can handle that).

This is probably why people like Tannoy for classical. 96 db/W efficiency with power handling of 550W peak for a Canterbury.
That gets you almost 118 dB at your distance.

Although you could argue that it's not a good idea to listen that loudly.
https://www.sciencemag.org/news/2014/09/sounds-you-cant-hear-can-still-hurt-your-ears#

 

[helom]

Well, you got the clipping light to turn on, so by definition, you'll need two AHB2's to run as bridged monoblocks to listen at the same level. The MG-III are a larger panel with true ribbons. They are less efficient. (https://www.stereophile.com/floorloudspeakers/868/index.html). They are great for vocals, but are hard to drive dynamically.

I did a quick analysis of the track. It's definitely got impressive dynamic range. In my software, I see average SPL at -26 dB with a peak sample at the end of -5.74 dB (and if you average 1 second, maybe -15 dB peaks). The average SPL includes the peaks, so if I just pick a 30 second range without peaks, it's recorded at -34 dB.

If we assume it's a 28 dB swing between average and peak levels, your 85 dB average is probably actually pushing 113 dB+ for a single peak which triggered your clipping even though it's probably closer to a 20 dB swing

SO, at 5W, you're getting ~87 dB.

At 190W, you're getting 103 dB. (need 106dB, or even 113 dB).
To get to 106 dB, you need 400W. (Assuming that the 0.7 can handle that).

This is probably why people like Tannoy for classical. 96 db/W efficiency with power handling of 550W peak for a Canterbury.
That gets you almost 118 dB at your distance.

Although you could argue that it's not a good idea to listen that loudly.
https://www.sciencemag.org/news/2014/09/sounds-you-cant-hear-can-still-hurt-your-ears#

Thanks for that analysis. I figured the track has a wide dynamic range but didn't think it was that wide. Still, that's much louder than I normally play the .7s, and one of the most dynamic tracks I listen to, so not really a practical concern.

 

[anmpr1]

If we assume it's a 28 dB swing between average and peak levels, your 85 dB average is probably actually pushing 113 dB+ for a single peak which triggered your clipping even though it's probably closer to a 20 dB swing... SO, at 5W, you're getting ~87 dB. At 190W, you're getting 103 dB. (need 106dB, or even 113 dB). To get to 106 dB, you need 400W. (Assuming that the 0.7 can handle that).

This is probably why people like Tannoy for classical. 96 db/W efficiency with power handling of 550W peak for a Canterbury. That gets you almost 118 dB at your distance. Although you could argue that it's not a good idea to listen that loudly.

Your point about tolerating a very loud loudspeaker in a domestic environment is important. The late Drew Daniels (former JBL engineer) argued that it is the SPL/distortion combination that must be considered when designing loudspeakers intended to play at realistic SPL. His low distortion DIY project (which can be found on-line doing a name search) consisted of highly efficient speakers (reflex/horn) and two thousand watts of tri amplification. His advice:

"...play music at no more than realistic levels. You will get the best representation of the original sound if you play the reproduction at the original sound level. Playing too loud is as detrimental to fidelity as playing too softly.

If you play predominantly rock music, there is no such thing as an original sound level-since all the recorded material comes out of a little electronic box or was derived by sticking a microphone somewhere you would never purposely put your ears.

In either case, you need to keep a sound level meter handy. ...don't ignore this advice. These speakers make so little distortion that you will be tempted to believe that the 120 dB sound you are listening to is only playing at 90 dB. This is not good. You will lose your hearing. Don't let this happen."

I've posted this before, but it is relevant and a simple explanation (IMO). During a discussion with the late Peter Aczel about an article I wrote for his Webzine, he once explained:

"The dynamic range of the human ear is more than 120 dB. The dynamic range of 16-bit digital recording is theoretically 98 dB. The difference between the absolute softest audible music in a concert hall and the loudest climaxes is of the order of 60 to 70 dB because of the ambient noise floor. Let us say you need 1 milliwatt of amplifier power, in a given installation, to play the softest passages (I am just guessing), then 70 dB above that would come to 10,000 watts. Any domestic loudspeaker would go up in smoke with that kind of input.

With extremely high-efficiency horn-type theater speakers the numbers change; it is actually possible to produce levels of 110 or 115 dB or even more in a single installation, and here’s the remarkable thing—you can tolerate it because the distortion is low. We tend to judge loudness by the amount of distortion we hear, not by SPL! You wouldn’t adjust the volume control if you heard no distortion. So, you could have your “too good” 98-dB balls-to-the-wall digital recording without compression, if the efficiency and power-handling capability of your system were adequate—which they generally are not."

In an early commercial magazine test of Bob Carver's Phase Linear 700 amplifier driving AR LST speakers (fairly low distortion and able to 'absorb' a lot of amplifier) J. Hirsch noted that the listening panel tended to drive the amp into clipping, which was not done with other amps. The reason? Other lower powered amps (iuncluding the Crown DC-300) distorted badly when over driven (i.e., playing the speakers at very loud levels), causing them to be turned down. The Phase was so powerful (remember, this was the early '70s) that short term clipping was tolerated because the overall program SPL remained undistorted.

 

[Inner Space]

"You will get the best representation of the original sound if you play the reproduction at the original sound level."

Yes - Peter Walker, Quad founder ("The closest approach to the original sound") said any recording has only one correct replay level.

"We tend to judge loudness by the amount of distortion we hear ... "

Yes again - because of applied FX and reverb, the guitar on Led Zeppelin's "You Shook Me" sounds shatteringly immense, in a huge space. In fact, numerically, it's one of the quietest tracks on the recording.

 

[anmpr1]

Yes - Peter Walker, Quad founder ("The closest approach to the original sound") said any recording has only one correct replay level.

The question to my mind is whether this 'correct' level has anything to do with the actual sound supposedly captured at the recording venue. Or is this correct level simply an artificial constraint necessitated by obvious real-world limitations--something that has little to do with reproducing an actual performance?

I would modify the idea by saying that an all acoustic recording could sound 'realistic' if it is played at a level matching the original instrument's SPL. However, for various reasons the realistic SPL of musical instruments is generally impractical to achieve within a domestic environment. Nor would we want it even if it could be attained--at least on a consistent listening basis. The example I presented to Peter Aczel was simple: take a well known John Coltrane recording. Four instruments: Coltrane on soprano or tenor saxophone; McCoy Tyner on acoustic piano; Steve Davis on acoustic bass; Elvin Jones on drums. And let us imagine (with our unlimited imagination) that the recording engineer is able to realistically (in all aspects) record the session.

With that in mind, can anyone expect to play the said recording at a 'realistic' level in one's living room? Move the plants, coffee table and chairs, and imagine these four musicians in your living room (even if it is large enough), playing at moderate volume. Possibly you could deal with it for a short time; at least until John brings in Pharaoh Sanders, and instructs him to wail away. That sort of SPL would quite frankly drive you out of your house. The live music would, however, possess essentially unlimited dynamic range (instrument delimited, of course) plus zero distortion.

Now, imagine this being reproduced by any normal sound system. First, loudspeakers would never sound as natural--neither as loud, dynamic nor as distortion-free. Certainly not Peter Walker's Quad--at least in the dynamic SPL area. But neither would any box speaker, regardless of how Klippelized it be. You'd need something like Drew Daniels homemade monster (with thousands of watts) set-up. Commercial horn loudspeakers and fewer watts would approach the SPL and do it with a certain low distortion, but the chances of a horn speaker actually sounding as natural as the instruments is a big question.

We move on. Imagine a symphony orchestra. Or a live Grateful Dead concert. At that point and at that level this entire mind experiment breaks down to the point of ridiculousness. Which means we have to substitute 'the reproduction of the original sound' with simply some measure of a reasonable facsimile, one necessarily having numerous inherent limitations.

 

[Inner Space]

" ... we have to substitute 'the reproduction of the original sound' with simply some measure of a reasonable facsimile ... "

I'm sure Walker accepted - as we all must - that to record is to compress and to miniaturize, and therefore his notion of "one correct replay level" must have been scaled down to a level he found consonant with the size of the in-room image presented. Clearly he knew no equipment - least of all his - could reproduce live levels. So I agree with your point completely. In fact, all my equipment choices could be characterized as "a relentless pursuit of a reasonable facsimile" ... not a great marketing slogan, but an inescapable reality.

 

[RichB]

I have listened to a trio consisting of a baby grand piano, standup bass, and trumpet in my home.
The room is small 16x16x12 feet and had a fair number of people sound absorbers.

The trio was playing conservatively, since it was intended to be enjoyable.
I do feel that I could achieve similar levels in my main room 30x15x12 with the Salon2's driven by the AHB2's however, I don't have any recordings that can match a pianist playing the baby grand.

Recently, the piano tuner, who is a gifted player was riffing after completing the tune. He is also into audio gear and loved my sound of my system. But honestly, nothing I could demo for him cam close the the experience of listening to him play. The best demo was multi-channel recording "Dave Matthews & Tim Reynolds: Live at Radio City Music Hall".

https://www.amazon.com/Dave-Matthew...&sprefix=dave+matthew+and+tim+,aps,137&sr=8-1

- Rich

 

[cjm2077]

The question to my mind is whether this 'correct' level has anything to do with the actual sound supposedly captured at the recording venue. Or is this correct level simply an artificial constraint necessitated by obvious real-world limitations--something that has little to do with reproducing an actual performance?

I would modify the idea by saying that an all acoustic recording could sound 'realistic' if it is played at a level matching the original instrument's SPL. However, for various reasons the realistic SPL of musical instruments is generally impractical to achieve within a domestic environment. Nor would we want it even if it could be attained--at least on a consistent listening basis. The example I presented to Peter Aczel was simple: take a well known John Coltrane recording. Four instruments: Coltrane on soprano or tenor saxophone; McCoy Tyner on acoustic piano; Steve Davis on acoustic bass; Elvin Jones on drums. And let us imagine (with our unlimited imagination) that the recording engineer is able to realistically (in all aspects) record the session.

With that in mind, can anyone expect to play the said recording at a 'realistic' level in one's living room? Move the plants, coffee table and chairs, and imagine these four musicians in your living room (even if it is large enough), playing at moderate volume. Possibly you could deal with it for a short time; at least until John brings in Pharaoh Sanders, and instructs him to wail away. That sort of SPL would quite frankly drive you out of your house. The live music would, however, possess essentially unlimited dynamic range (instrument delimited, of course) plus zero distortion.

Now, imagine this being reproduced by any normal sound system. First, loudspeakers would never sound as natural--neither as loud, dynamic nor as distortion-free. Certainly not Peter Walker's Quad--at least in the dynamic SPL area. But neither would any box speaker, regardless of how Klippelized it be. You'd need something like Drew Daniels homemade monster (with thousands of watts) set-up. Commercial horn loudspeakers and fewer watts would approach the SPL and do it with a certain low distortion, but the chances of a horn speaker actually sounding as natural as the instruments is a big question.

We move on. Imagine a symphony orchestra. Or a live Grateful Dead concert. At that point and at that level this entire mind experiment breaks down to the point of ridiculousness. Which means we have to substitute 'the reproduction of the original sound' with simply some measure of a reasonable facsimile, one necessarily having numerous inherent limitations.

Every recording gets mastered at one volume level. That is the level the engineer listened at when he made all the final decisions about the recording. If you play it back at a different volume level, it will sound different because of the changes in the human ear's sensitivity across frequency at different spl levels. Now you would also have to add in having different speakers, a much different listening environment, and different ears, so it's not like you can ever make what you hear on a recording exactly the same as what the engineer heard. And no recording is ever going to sound exactly like a live performance, because a live performance doesn't sound exactly the same to all of the different people in the room it is being recorded in. What distance from a trumpet is the reference distance, etc?

 

[DonH56]

Every recording gets mastered at one volume level. That is the level the engineer listened at when he made all the final decisions about the recording. If you play it back at a different volume level, it will sound different because of the changes in the human ear's sensitivity across frequency at different spl levels. Now you would also have to add in having different speakers, a much different listening environment, and different ears, so it's not like you can ever make what you hear on a recording exactly the same as what the engineer heard. And no recording is ever going to sound exactly like a live performance, because a live performance doesn't sound exactly the same to all of the different people in the room it is being recorded in. What distance from a trumpet is the reference distance, etc?

Ooh, ooh, I know this one!

For me, ummm (holds up horn), about 18"...

 

[jaynewt]

I took delivery just last week of two ABH2 to run as mono-blocks with MBL 121s and mola-mola makua pre-amp + dac. Previous amps were mola-mola Kalugas. I just want to report that the combination is absolutely stunning. I've been an audiophile for 50 years and I've never had so good a sound as I have now. Previous amps have included Spectral and MBL 9008A none of which were a patch on this little powerhouse. I was a little worried as the MBLs are 4ohm models but I haven't had any problems at all. I just wish I'd found these years ago.

 

[GXAlan]

Y

I took delivery just last week of two ABH2 to run as mono-blocks with MBL 121s.

You should send those MBL’s in for testing!

 

[Laserjock]

I took delivery just last week of two ABH2 to run as mono-blocks with MBL 121s and mola-mola makua pre-amp + dac. Previous amps were mola-mola Kalugas. I just want to report that the combination is absolutely stunning. I've been an audiophile for 50 years and I've never had so good a sound as I have now. Previous amps have included Spectral and MBL 9008A none of which were a patch on this little powerhouse. I was a little worried as the MBLs are 4ohm models but I haven't had any problems at all. I just wish I'd found these years ago.

Says a lot..

 

[anmpr1]

...so it's not like you can ever make what you hear on a recording exactly the same as what the engineer heard.

I have read (here on ASR) how the goal (or at least some listener's goal) is to have a sound system that let's them 'hear' what the recording engineer heard. If that is the goal (it's not my goal, but it could be someone's) then they had better find out what loudspeakers were used on mixdown, where they were positioned, and the loudness level used. But I really think what they are saying is that they want a loudspeaker that is 'neutral', i.e., reasonably flat across the board, and exhibits fairly low distortion.

The argument about the goal of hi-fi being to accurately reproduce the sound of acoustic (unamplified) instruments in a given space more or less took off with Harry Pearson. His 'absolute sound', as it were. The irony is that Pearson used some of the most colored and frankly downright weird audio gear for his reference. But if you told Harry that he'd just call you a rude name.

 

[RichB]

I have read (here on ASR) how the goal (or at least some listener's goal) is to have a sound system that let's them 'hear' what the recording engineer heard. If that is the goal (it's not my goal, but it could be someone's).

I'd love to hear what the recording engineer heard before he or she compressed it to death and slammed it to 0 DBFS.
Accurate speakers and distortion free amps are not going to help.
It's a sin that vinyl recordings can sound better simply because its limitations apply to the mixing engineer as well.

- Rich

 

[Pjetrof]

I took delivery just last week of two ABH2 to run as mono-blocks with MBL 121s and mola-mola makua pre-amp + dac. Previous amps were mola-mola Kalugas. I just want to report that the combination is absolutely stunning. I've been an audiophile for 50 years and I've never had so good a sound as I have now. Previous amps have included Spectral and MBL 9008A none of which were a patch on this little powerhouse. I was a little worried as the MBLs are 4ohm models but I haven't had any problems at all. I just wish I'd found these years ago.

where the previous amps broken? There should be no audible diffrence!