• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

Adam S2V Studio Monitor Review

OP
amirm

amirm

Founder/Admin
Staff Member
CFO (Chief Fun Officer)
Joined
Feb 13, 2016
Messages
44,368
Likes
234,395
Location
Seattle Area
Curious if there's a measured difference between analog in and digital in.
It is not going to show up in the context of an acoustic measurement in non-silent room.
 

andreasmaaan

Master Contributor
Forum Donor
Joined
Jun 19, 2018
Messages
6,652
Likes
9,399
Unusual or not, both Andrew Jones and the founder of Hedd, Klaus Heinz, say it's quite awkward, and decide to do the opposite with their current speakers.

IMHO, Klaud Heinz's approach is what's awkward :p Due to his insistence on avoiding DSP in the speaker itself, his systems require the user to run the output from their computer source through an FIR filter before sending it to the speakers so as to achieve the superior phase performance that could have been achieved inside the speaker if Hedd simply adopted the standard practice of using DSP crossovers in the speakers. Not to besmirch Mr Heinz's skill and reputation as a speaker designer, which is formidable - I just strongly disagree with him on this point.

But then again, it it takes digital as you mention via that AES3, what kind of digital signal is that? it then has a DAC inside? the dsp thing is done with that dac, or after it's turned into analog?

From a practical perspective, there's no difference between AES3 and SPDIF. They both transmit digital audio and clock information between components. It's preferred in pro environments for various reasons, some practical and some historical. There's no reason you can't use it at home.

Regarding the location of the DAC in the system, this is in essence how the signal path flows:

Digital source --> AES3 input (digital) --> DSP --> DACs --> amps --> speaker drivers

In other words, the signal remains wholly in the digital domain until after DSP processing. This is the optimal signal path primarily for the following reasons:
  • Only one DA conversion takes place.
  • The DSP enables exponentially more precise control and complexity in the implementation of the crossover than analogue filters would allow.
  • The user does not need to concern themselves with analogue gain staging, since the entire analogue chain is inside the speaker.
PS: a few home audio companies are now also using AES3 rather than SPDIF in their speaker systems. Kii is one that comes to mind.
 
Last edited:
D

Deleted member 12179

Guest
You probably need to consider the target market for these speakers which is professional studio mixing and mastering. The advantage of DSP in the monitors is that the digital mixer audio interface allows for endless options when it comes to adding presets, templates, etc and ease of transfer projects from one studio to another. and that the signal is sent as digital purely prior to DSP that is calibrated to speakers and the studio room. The DSP built-in to the speakers makes sense to me in that scenario.

Sure, but there is not that scenario, there are several scenarios, each scenario is going to give slightly different results, and therefore, that is what I am interested about, how much is that slightly. Because all I mentioned, the issue, is when the line coming is analog. Now if there is digital input, the dsp after is it's absolutely marvelous and could not complain about it at all. Not to mention than the possible issue when using the analog, it's just possible, and would have loved to have the measurements about how much is different.

Thing is, these kind of speakers, or the similar model with 3 ways, or the hedd 3 ways... they are designed and built towards pro use, but. But, but, but, and then but again... Considering what a good passives costs plus a decent amplifier, they are not so expensive, and some people will prefer to have a set of these at their homes, rather than elac actives, or those retro JBL made out of wood. I bet there are a ton of electronic music enthusiasts and amateur dj's, and also plenty of drug dealers, who would love to have a pair of these for listening, rather than a set of white old usa man in their 60-70's speaker model, with tube amp, separate amp for left and right, separate power supply, separate pre-amp, separate, phone amp... LOL!! Even with more hi-fi active speakers coming, many people will like to have pro monitors at their living room instead. Many people already do. Then again, not sure how that AES input is used at home indeed ;-)
 
D

Deleted member 12179

Guest
IMHO, Klaud Heinz's approach is what's awkward :p Due to his insistence on avoiding DSP in the speaker itself, his systems require the user to run the output from their computer source through an FIR filter before sending it to the speakers so as to achieve the superior phase performance that could have been achieved inside the speaker if Hedd simply adopted the standard practice of using DSP crossovers in the speakers. Not to besmirch Mr Heinz's skill and reputation as a speaker designer, which is formidable - I just strongly disagree with him on this point.



From a practical perspective, there's no difference between AES3 and SPDIF. They both transmit digital audio and clock information between components. It's preferred in pro environments for various reasons, some practical and some historical. There's no reason you can't use it at home.

Regarding the location of the DAC in the system, this is in essence how the signal path flows:

Digital source --> AES3 input (digital) --> DSP --> DACs --> amps --> speaker drivers

In other words, the signal remains wholly in the digital domain until after DSP processing. This is the optimal signal path primarily for the following reasons:
  • Only one DA conversion takes place.
  • The DSP enables exponentially more precise control and complexity in the implementation of the crossover than analogue filters would allow.
  • The user does not need to concern themselves with analogue gain staging, since the entire analogue chain is inside the speaker.
PS: a few home audio companies are now also using AES3 rather than SPDIF in their speaker systems. Kii is one that comes to mind.

It will work like that if you use it like that, that is; digital input to the speakers.

If you use it as Amir has tested them, then not. You would have a digital source, then dac, then analog cable, then ADC
, then dsp, then DAC, amps, speaker drivers.

What you describe is exactly how Amir has not tested them.
 
D

Deleted member 12179

Guest
IMHO, Klaud Heinz's approach is what's awkward :p Due to his insistence on avoiding DSP in the speaker itself, his systems require the user to run the output from their computer source through an FIR filter before sending it to the speakers so as to achieve the superior phase performance that could have been achieved inside the speaker if Hedd simply adopted the standard practice of using DSP crossovers in the speakers. Not to besmirch Mr Heinz's skill and reputation as a speaker designer, which is formidable - I just strongly disagree with him on this point.



From a practical perspective, there's no difference between AES3 and SPDIF. They both transmit digital audio and clock information between components. It's preferred in pro environments for various reasons, some practical and some historical. There's no reason you can't use it at home.

Regarding the location of the DAC in the system, this is in essence how the signal path flows:

Digital source --> AES3 input (digital) --> DSP --> DACs --> amps --> speaker drivers

In other words, the signal remains wholly in the digital domain until after DSP processing. This is the optimal signal path primarily for the following reasons:
  • Only one DA conversion takes place.
  • The DSP enables exponentially more precise control and complexity in the implementation of the crossover than analogue filters would allow.
  • The user does not need to concern themselves with analogue gain staging, since the entire analogue chain is inside the speaker.
PS: a few home audio companies are now also using AES3 rather than SPDIF in their speaker systems. Kii is one that comes to mind.

Well, then you have another ackward with andrew jones. I think this is very obvious; if you are sending a digital signal to the speaker, the DSP is awesome idea, otherwise, it is not a good idea. Both andrew jones and hedd have designed their speakers, actieve, but still, the cable going to them contains analog audio, therefore the dsp is better to have it before. In the future, if the cables contain digital, the dsp in the speaker will seem a more apropiate thing. But then again, how is the DAC in the speaker then if you are sending digital to it??? if these speakers can take digital AES, where is the measurement of that DAC? is it good the dac, is it 50$ good, or 200$ good, or 2000$ good? Why 50$ dacs are measured but then these 4000$ speakers have a dac and we do not care about how it performs?

Ton of questions not answered here, but these things still costs 4000$.
 

andreasmaaan

Master Contributor
Forum Donor
Joined
Jun 19, 2018
Messages
6,652
Likes
9,399
Sure, but there is not that scenario, there are several scenarios, each scenario is going to give slightly different results, and therefore, that is what I am interested about, how much is that slightly.

I'd be willing to bet, inaudible and acoustically immeasurable so long as appropriate gain staging is used prior to the analogue input into the speaker.

Then again, not sure how that AES input is used at home indeed ;-)

All you'd need would be a USB-AES3 converter, the same basic type of device as a USB-SPDIF converter. This would do the job, for example.

I think this is very obvious; if you are sending a digital signal to the speaker, the DSP is awesome idea, otherwise, it is not a good idea.

Analogue crossovers are very limited in comparison to DSP crossovers. This limits the options the designer has, and therefore limits the speaker.

In contrast, decent quality ADCs do not introduce any audible noise or distortion. They allow an analogue signal to be converted to digital where it can be processed in much more sophisticated ways inside the speaker, completely free of audible consequences.
 

andreasmaaan

Master Contributor
Forum Donor
Joined
Jun 19, 2018
Messages
6,652
Likes
9,399
But then again, how is the DAC in the speaker then if you are sending digital to it??? if these speakers can take digital AES, where is the measurement of that DAC? is it good the dac, is it 50$ good, or 200$ good, or 2000$ good? Why 50$ dacs are measured but then these 4000$ speakers have a dac and we do not care about how it performs?

TBH, I mostly don't read the DAC measurements on this site (or elsewhere). So many of them perform well in advance of any human auditory system, there's no reason to concern oneself with them, except to rule out the odd lemon.

EDIT: you might also ask, why don't we remove the amps and analogue crossovers from all active speakers that are measured and measure those components? They are almost guaranteed to be producing more noise and distortion than any ADC or DAC.
 

Francis Vaughan

Addicted to Fun and Learning
Forum Donor
Joined
Dec 6, 2018
Messages
933
Likes
4,697
Location
Adelaide Australia
Something worth mentioning, just so there isn't too much confusion. The AES3 and S/PDIF protocols are essentially identical. When you want a converter you are only changing connectors (maybe driving a twisted pair) and changing voltage levels. The bits on the wire are the same thing. There is nothing like the work involved in a USB to S/PDIF conversion. The only difference is in the metadata, most importantly a lack of copy protection in AES3. But other than that, the actual coding of audio is exactly the same. Which is why it is so easy for DACs to be made with AES3 inputs. AES3 suffers from exactly the same ills as S/PDIF. For domestic audio products, adding AES3 is just vanity, making products look "professional". AES3 comes into its own when longer runs are needed, and generally more robust installations are needed. OTOH, the XLR connector was never designed for the bandwidth, and is something of a wart on the standard.
 

KSTR

Major Contributor
Joined
Sep 6, 2018
Messages
2,690
Likes
6,013
Location
Berlin, Germany
I'd say it's more a low tech assembly process, than not a high precision process, it was a simple jig they were using to get each pleat the same.
The jigs may seem low-tech but we must not forget the 20 years of experience of the staff making the pleats, apply damping and assembling the whole things. And note, perfect smoothness and geometry of the pleats to a single, static target doesn't neccessarily give best performance. AMTs are a can of worms, design-wise.
 

thewas

Master Contributor
Forum Donor
Joined
Jan 15, 2020
Messages
6,747
Likes
16,186
AMTs are a can of worms, design-wise.
But on the other hand they are drivers that can be self manufactured with a relatively small amount of investment and tools, that's why its very fashionable for many small high end loudspeaker manufacturers to build their own ones as a "unique selling point" while almost none is able to build their own cone or dome drivers ;)
 

Soniclife

Major Contributor
Forum Donor
Joined
Apr 13, 2017
Messages
4,500
Likes
5,417
Location
UK
The jigs may seem low-tech but we must not forget the 20 years of experience of the staff making the pleats, apply damping and assembling the whole things. And note, perfect smoothness and geometry of the pleats to a single, static target doesn't neccessarily give best performance. AMTs are a can of worms, design-wise.
For clarity, I wasn't implying low tech was bad, I like a relatively simple engineering solution to a problem.
 

Costas EAR

Active Member
Forum Donor
Joined
Jan 15, 2020
Messages
157
Likes
348
Location
Greece
Something worth mentioning, just so there isn't too much confusion. The AES3 and S/PDIF protocols are essentially identical.
I think that there is an AES EBU consumer protocol, exactly identical to spdif, and an AES EBU pro version, with some more bits of info, excluding also the copy protection.

Voltage variation also noted between pro and consumer AES EBU versions.
 

Andreas007

Active Member
Joined
Mar 11, 2019
Messages
138
Likes
353
Location
Germany, Bavaria
IMHO, Klaud Heinz's approach is what's awkward :p Due to his insistence on avoiding DSP in the speaker itself, his systems require the user to run the output from their computer source through an FIR filter before sending it to the speakers so as to achieve the superior phase performance that could have been achieved inside the speaker if Hedd simply adopted the standard practice of using DSP crossovers in the speakers. Not to besmirch Mr Heinz's skill and reputation as a speaker designer, which is formidable - I just strongly disagree with him on this point.



From a practical perspective, there's no difference between AES3 and SPDIF. They both transmit digital audio and clock information between components. It's preferred in pro environments for various reasons, some practical and some historical. There's no reason you can't use it at home.

Regarding the location of the DAC in the system, this is in essence how the signal path flows:

Digital source --> AES3 input (digital) --> DSP --> DACs --> amps --> speaker drivers

In other words, the signal remains wholly in the digital domain until after DSP processing. This is the optimal signal path primarily for the following reasons:
  • Only one DA conversion takes place.
  • The DSP enables exponentially more precise control and complexity in the implementation of the crossover than analogue filters would allow.
  • The user does not need to concern themselves with analogue gain staging, since the entire analogue chain is inside the speaker.
PS: a few home audio companies are now also using AES3 rather than SPDIF in their speaker systems. Kii is one that comes to mind.

I just wonder...
If the future is DAC within the speaker how will amir measure DAC performance then (which we all so care about)? ;)
 

Costas EAR

Active Member
Forum Donor
Joined
Jan 15, 2020
Messages
157
Likes
348
Location
Greece
In real life, the dynamic range of the speaker at the listening distance is the only "sinad" parameter that counts, and that is what should be measured even now. ;)

So, max spl (with low thd) at the whole 3rd octave is needed, this is the limit. Background noise level is the other limit. Between these two, you can find the useful dynamic range of the speaker system.

The first 2 octaves belong to the subs.
 

pozz

Слава Україні
Forum Donor
Editor
Joined
May 21, 2019
Messages
4,036
Likes
6,827
@Lolito

The ADC/DAC in the S2V is audibly transparent. I used both analog input and AES3. I'm not complaining at all about the sound in my setup.

The AES3 in/out I actually daisy-chained for multichannel at one point. The DSP for EQ, levels, delays and channel configuration is useful but the interface, both on the speaker and when using the ADAM software on my computer, is beyond awkward. I got used to it eventually, although everytime I wanted to change something or experiment I had to ask myself many times whether or not that tweak was really necessary. Like Amir said, messing with the knob on the back is maybe good for selecting inputs or the overall profile (Pure, UNR, User 1, 2, 3, etc.). What ends up happening is that you connect USB to the speaker, load up the software, read the config, change it, write it, set the profile, and then rinse and repeat. It takes a long time.

To test all the things you mentioned in the manner you'd like means taking apart the speaker. Even then a direct test might not be possible depending on how the board inside is designed.
 

tuga

Major Contributor
Joined
Feb 5, 2020
Messages
3,984
Likes
4,281
Location
Oxford, England
In real life, the dynamic range of the speaker at the listening distance is the only "sinad" parameter that counts, and that is what should be measured even now. ;)

So, max spl (with low thd) at the whole 3rd octave is needed, this is the limit. Background noise level is the other limit. Between these two, you can find the useful dynamic range of the speaker system.

The first 2 octaves belong to the subs.

How do you measure:

a) resolution - the speaker's ability to reproduce low-level detail

b) self-noise (CSD?)
 

Costas EAR

Active Member
Forum Donor
Joined
Jan 15, 2020
Messages
157
Likes
348
Location
Greece
Ah, big questions. ;)
Maybe i can just give some facts, and then everything else is quite obvious.

First of all, I suppose that these questions can apply to all speaker measurements. :)

By definition, the perceived low level detail (this is the point) is above Fletcher and Munson's 20 phon curve (if we assume that a nice quiet listening room has the threshold of noise down to 20 dB's, which is rather underestimated), so it is frequently dependent - at different spl's.

The perceived usable spl is the following:

Screenshot_20181206-091830.png


And of course above the 100 dB phon curve it is only estimated, it exists just for science books to discover the threshold of pain.

Screenshot_20190928_110507_com.android.chrome.jpg


Which means, that the maximum possible usable dynamic range is between these lines, i suppose that you can calculate the difference in every frequency.

The problem in this diagram is nothing else but the extreme high spl's at low frequencies, which can be tolerated quite easily, but it is quite difficult to achieve, you have to use many really big and powerful subs.

Of course, listening to reference level, doesn't need to cover all the available gray space in this graph spl's, but then go figure the really usable resolution, without forgetting the facts of psychoacoustics, the loudness curves, the need for loudness filter at low levels and the average noise in a typical home.

So, before asking the measurements of the speakers, do you really know what is the absolutely best expected resolution in high fidelity reproduction?

Self generated noise, is quite easy to measure, this is not a problem in high quality active speakers, it is well below the noise floor of the room, or even under the phon 10 curve.

Any thoughts?
 
Last edited:

tuga

Major Contributor
Joined
Feb 5, 2020
Messages
3,984
Likes
4,281
Location
Oxford, England
Ah, big questions. ;)
Maybe i can just give some facts, and then everything else is quite obvious.

First of all, I suppose that these questions can apply to all speaker measurements. :)

By definition, the perceived low level detail (this is the point) is above Fletcher and Munson's 20 phon curve (if we assume that a nice quiet listening room has the threshold of noise down to 20 dB's, which is rather underestimated), so it is frequently dependent - at different spl's.

The perceived usable spl is the following:

View attachment 50282

Which means, that the maximum possible usable dynamic range is between these lines, i suppose that you can calculate the difference in every frequency.

The problem in this diagram is nothing else but the extreme high spl's at low frequencies, which can be tolerated quite easily.

Of course, listening to reference level, doesn't need to cover all the available in this graph spl's, but then go figure the really usable resolution, without forgetting the facts of psychoacoustics, the loudness curves, the need for loudness filter at low levels and the average noise in a typical home.

So, before asking the measurements of the speakers, do you really know what is the absolutely best expected resolution in high fidelity reproduction?

Self generated noise, is quite easy to measure, this is not a problem in high quality active speakers, it is well below the noise floor of the room, or even under the phon 10 curve.

Any thoughts?

You've probably experienced some (flat response) speakers producing more perceived detail at low listening levels than others. You turn down the volume and apart from the change in tonal balance the detail is still there with the former but with the latter it becomes a bit of a mash.
What could be causing this?
Suspension, surround, delayed driver resonances, delayed cabinet resonances, something in the crossover?

The importance of this may be relative with close-mic'ed jazz trios but the problem becomes apparent when listening to orchestral music which not only produces massively wide dynamic swings but sometimes an instrument or group of instruments will be playing at a lower level.
 
Top Bottom