Which bookshelf crossover (non-DSP) provides highest performance under $2,500? Acoustic Energy AE1?

[mel]

Some kind of new amplifier design.
I am not familiar with it, but it seems to do away with high frequency filtering of Class-D, but still be more efficient than Class AB.

I would not put much weight into amplifier “types”.
A well designed good amplifier is a good amplifier, it does not matter what ”Class” of design in particular.
What is most certain is the difference in efficiency (power consumption).
Modern Class-D is the most efficient followed by Class AB, and then Class A which is the oldest design together with Tube amplifier design which can typically consume 200w+ at all times (if you forget to switch it off).

• bit annoyed with is that there isn't a +4 db reference level setting
  • which is fairly standard in a lot of audio equipment and is what my
  • BM6A's expect over XLR.

https://en.wikipedia.org/wiki/Alignment_level

The alignment level in an audio signal chain or on an audio recording is a defined anchor point that represents a reasonable or typical level. It does not represent a particular sound level or signal level or digital representation, but it can be defined as corresponding to particular levels in each of these domains.

For example, alignment level is commonly 0 dBu (Equal to 0.775 Volts RMS) in broadcast chains and in professional audio what is commonly known as 0 VU, which is +4 dBu (equal to 1.227 Volts RMS) in places where the signal exists as analogue voltage. Under normal situations, the "0VU" reference allowed for a headroom of 18 dB or more above the reference level without significant distortion. This is largely due to the use of slow responding VU meters in almost all analogue professional audio equipment which, by their design, and by specification responded to an average level, not peak levels. It most commonly is at −18 dB FS (18 dB below full scale digital) on digital recordings for programme exchange, in accordance with EBU recommendations. Digital equipment must use peak reading metering systems to avoid severe digital distortion caused by the signal going beyond full scale or maximum digital levels. 24-bit original or master recordings commonly have alignment level at −24 dBFS to allow extra headroom, which can then be reduced to match the available headroom of the final medium by audio level compression. FM broadcasts usually have only 9 dB of headroom as recommended by the EBU, but digital broadcasts, which could operate with 18 dB of headroom, given their low noise floor even in difficult reception areas, currently operate in a state of confusion, with some transmitting at maximum level while others operate at a much lower level even though they carry material that has been compressed for compatibility with the lower dynamic range of FM transmissions.

 

[mel]

https://en-de.neumann.com/kh-310-a#description

Its closed design enables a very fast transient response.
The closed cabinet design means a lower group delay. This results in almost the entire frequency range being reproduced at the same time; the transient response is precisely controlled at all times.

The KH 310 is a three-way

• tri-amplified monitor with thermally protected
• Class AB amplifiers (210 W and two times 90 W).

XLR electronically balanced analogue input

• Standard interfacing to professional equipment
• Wide range input gain and output level controls
  • Easier interfacing with signal sources

Controller design Analog; active
Crossover frequency 650 Hz and 2.0 kHz
Crossover slope 24 dB/oct; 4th order

Equalization: Bass0; -2.5; -5; -7.5 dB
Equalization: Low-Mid 0; -1.5; -3; -4.5 dB
Equalization: Treble+1; 0; -1; -2 dB

 

[gene_stl]

You're getting warmer.

 

[mel]

https://www.forum.rme-audio.de/viewtopic.php?id=28649
Re: How big of a problem is buying a speaker with its own internal DSP?
Hi,
I have Adi-2 pro and Adam s3v. I have tried analog (XLR) and digital (AES) connections.

• Digital does not use RME D/A converter and Adam A/D converter.
  • It plays better, not just me.
• I use RME as a DSP only.
  • Adam is set to -10 dB to take advantage of the
    • dynamic range of RME's digital volume control.
    • In this case, RME cannot play DSD, but only PCM. I perform DSD to PCM conversion on the player (Foobar2000).
  • I also had Adam A7X and the S3V sound much better for me.

Guys after some research and weighing the options, I've decided to go with the Hedd Type 20 over the S2V.

Advantages are:

-3 way speaker vs. 2 way (S2V) for only $250 more per speaker - I've wanted to move to a 3-way speaker.
-Analog only input with no conversion - Klaus Heinz, originally of Adam Audio, designed many of Adam's best monitors, and started Hedd. He states clearly that a big reason they stayed with an analog design is to avoid having to do reconversion on an analog signal from a high end dac.
-Hedd Type 20 gets some pretty strong reviews from multiple users/publications, including some users I talked to

 

[mel]

https://www.forum.rme-audio.de/viewtopic.php?id=28649

This is a good, lengthy thread about RME ADI and different speakers. This is going to take a long time.

Re: How big of a problem is buying a speaker with its own internal DSP?
Hi,
I have Adi-2 pro and Adam s3v. I have tried analog (XLR) and digital (AES) connections.

• Digital does not use RME D/A converter and Adam A/D converter.
  • It plays better, not just me.
• I use RME as a DSP only.
  • Adam is set to -10 dB to take advantage of the
    • dynamic range of RME's digital volume control.
    • In this case, RME cannot play DSD, but only PCM. I perform DSD to PCM conversion on the player (Foobar2000).
  • I also had Adam A7X and the S3V sound much better for me.

Guys after some research and weighing the options, I've decided to go with the Hedd Type 20 over the S2V.

Advantages are:

-3 way speaker vs. 2 way (S2V) for only $250 more per speaker - I've wanted to move to a 3-way speaker.
-Analog only input with no conversion - Klaus Heinz, originally of Adam Audio, designed many of Adam's best monitors, and started Hedd. He states clearly that a big reason they stayed with an analog design is to avoid having to do reconversion on an analog signal from a high end dac.
-Hedd Type 20 gets some pretty strong reviews from multiple users/publications, including some users I talked to

 

[mel]

They are expensive due to all this swiss manufacturing and you'd better listen them first. At least read opinions about their sound: they are very clear and slightly emotionless, so with RME result might be ... not exciting at all.

OK. I went to the trouble of calibrating my AE3 speakers to the Do-Re-Mi scale using Ut queant laxis recordings. I configured the RME PEQ using a handheld Radio Shack sound meter. My intention was to gain some degree of control to deliberately color a neutral speaker.

https://en.wikipedia.org/wiki/File:Johannes.Hymnus.ogg

Would this technique work with neutral studio speakers, or am I missing something?

Does a five band parametric equalizer have the same effect on a two way and three way crossover?

https://en.wikipedia.org/wiki/Equalization_(audio)

 

[FeddyLost]

Would this technique work with neutral studio speakers, or am I missing something?

You are missing at least listening space boundaries, or simply: room.
Typically the best possible speaker is calibrated to be sufficiently flat in anechoic room on-axis at least. In real room there are a lot of reasons not to have flat responce at listening point, and equalization without understanding what is really going on is a bad idea.
I think you should read more about speaker qualities and their intraction with environment because with such unstable position you can easily make suboptimal decision and disappoint in this hi-fi woodoo while mentioned amount of money is enough to get really good speakers and live with them happily.
PS "Neutrality" also can be different. Anything expensive should be auditioned personally IMO.

 

[mel]

You are missing at least listening space boundaries, or simply: room.
Typically the best possible speaker is calibrated to be sufficiently flat in anechoic room on-axis at least. In real room there are a lot of reasons not to have flat responce at listening point, and equalization without understanding what is really going on is a bad idea.
I think you should read more about speaker qualities and their intraction with environment because with such unstable position you can easily make suboptimal decision and disappoint in this hi-fi woodoo while mentioned amount of money is enough to get really good speakers and live with them happily.
PS "Neutrality" also can be different. Anything expensive should be auditioned personally IMO.

I can program Fourier transforms, when the situation warrants. I have the education and training. I am a software developer with some audio physics classes. I can be as technical as the situation calls for.

I am just looking for some confirmation to prove the concept that a neutral speaker can be colored from my RME DAC PEQ without too much pain. All the PEQ does is transform some input. I want to apply a musical scale transform. Transform seems like a more appropriate term than color. I think a "smoothing function" is more descriptive than transformation function. Just smooth the input according to the Do-Re-Mi scale. You could consider a smoothing function as automated mastering process editing. All of the scales can be mathematically calculated, which would be a true DSP function. The goal is a more analog, than linear curve. Emotionless is a synonym for a flat or linear frequency response curve. The human voice is analog. Analog is difficult to edit manually.

This is quite funny, because I thought I was doing something wrong, when the Ut queant laxis music pushed me towards a similar curve.

I am looking for a speaker that is good at generating "Vocal EQ". My guess is a speaker that lends itself for speech therapy.

https://ledgernote.com/columns/mixing-mastering/how-to-eq-vocals/

The one, indispensable tool you'll want for EQing vocals is a parametric equalizer. Since you found this article, I'm assuming you know your way around a DAW and know what that is (your recording software). The best DAWs will have a parametric EQ plugin. Find that and place it on your vocal track in the mixer.

To re-cap, the general steps and round about values look like this:

1. Roll off the low-end starting around 90 Hz.
2. Reduce the mud around 250 Hz.
3. Add a high shelf around 9 kHz & a high roll off around 18 kHz.
4. Add a presence boost around 5 kHz.
5. Boost the core around 1 kHz to 2 kHz.
6. Reduce sibilance around 5 kHz to 8 kHz.

That's a roundabout tutorial you can always follow once you understand the basics of using the Q and how much or little to boost or cut. Even if you did exactly that on every vocal track you'd still end up with pretty good results. The specifics will vary on each track though.

EQ Frequency Range Characteristics
When vocal EQ, you can think of the frequency spectrum as six different segments with their own characteristics. While there is some blur between them, you'll find most problems fall right into the ranges provided below:

• Low-End Noise - 20 Hz to 80 Hz
• Boominess - 80 Hz to 300 Hz
• Muddiness - 250 Hz to 500 Hz
• Nasal Honk - 800 Hz to 1.5 kHz
• Presence - 4.5 kHz to 9 kHz
• Breathiness - 10 kHz to 15 kHz

You'll find other words that describe these characteristics too and can adopt whichever you like. But you want to be familiar with them all because other professionals may use differing terminology. So "Breathiness" may be called "Air" and "Presence" may be called "Clarity." Mud is often called "Boxiness."

https://producerhive.com/music-production-recording-tips/how-to-use-a-vocal-eq-chart/

When sophisticated software and hardware is necessary, then it can be brought to bear. Conceptually, a musical scale transform seems very straight forward, but I don't really know more than I can reason about.

I am actually more interested in how the brain encodes sound, than how machines do. I think the answer is straight forward. Regular expressions are the natural function of neural networks. Regular expressions are the essence of perceived sound. Music is the pleasant version. For example, when I rode my bike past the train crossing this morning, the bell makes a "ding, ding, ding" sound. That pattern is a regular expression. Bird express regular sound patterns as "chirp, chirp, chirp".

 

[mel]

I adopt a "declarative" (computer science sense, vs. procedural) approach to specifying a speaker. The goal is to abstract a studio monitor as a functional unit that best lends itself to music generation (in the sense of Do-Re-Mi scales). The basic voice reproduction principles might be traceable to LS3/5A. I feel the choices are simplified and naturally fall out after the parameters are "declared". Because I lack this product knowledge, I have to hunt down each brand and compare them.

I get the impression that I want to invert the features that accommodate editing functionality for music producers.

https://en.wikipedia.org/wiki/Fundamental_frequency

The fundamental frequency, often referred to simply as the fundamental, is defined as the lowest frequency of a periodic waveform. In music, the fundamental is the musical pitch of a note that is perceived as the lowest partial present. In terms of a superposition of sinusoids, the fundamental frequency is the lowest frequency sinusoidal in the sum of harmonically related frequencies, or the frequency of the difference between adjacent frequencies. In some contexts, the fundamental is usually abbreviated as f0, indicating the lowest frequency counting from zero.[1][2][3] In other contexts, it is more common to abbreviate it as f1, the first harmonic.[4][5][6][7][8] (The second harmonic is then f2 = 2⋅f1, etc. In this context, the zeroth harmonic would be 0 Hz.)

According to Benward's and Saker's Music: In Theory and Practice:[9]

Since the fundamental is the lowest frequency and is also perceived as the loudest, the ear identifies it as the specific pitch of the musical tone [harmonic spectrum].... The individual partials are not heard separately but are blended together by the ear into a single tone.​

For example:

• RME ADI-2 DAC compatibility
  • Built in filters
    • Find rolloff most similar to human voice graph in previous post
  • Five band PEQ
    • Make it roughly correspond to six regions on vocal EQ graph in previous post.
      • Approximately a sine wave
        • the "Box" region is exaggerated, but my purposes are largely inapproximate.
      • Roll off the low-end starting around 90 Hz.
      • Reduce the mud around 250 Hz.
      • Add a high shelf around 9 kHz & a high roll off around 18 kHz.
      • Add a presence boost around 5 kHz.
      • Boost the core around 1 kHz to 2 kHz.
      • Reduce sibilance around 5 kHz to 8 kHz.
• Digital (DSP) vs. analog based (already have DSP in DAC)
• Crossover
  • 2-way vs. 3-way
    • Three is the closer multiple of six voice regions (2 x 3, vs. 3 x 2)
    • I going to guess 3-way based on human voice curve
      • https://en.wikipedia.org/wiki/Equal-loudness_contour#Relevance_to_sound_level_and_noise_measurements
      • https://en.wikipedia.org/wiki/Vocal_range
  • active vs. passive
  • frequencies
• Input/Output Connections and protocols
  • Pre-amp functionality
  • XLR vs RCA line input
  • Digital vs. analog
  • Open subwoofer compatibility or walled garden
• Driver characteristics
  • magnet type (e.g., neodymium)
  • paper vs. metal
  • size
  • traditional vs. accordion
  • responsiveness
• Enclosure
  • Size
  • Sealed
  • Passive radiator
  • Ported
    • Slotted, vs. round
    • Front, vs. rear

All of a sudden, I understand the structure and nature of my Tidal album collection and listening habits.

https://en.wikipedia.org/wiki/Chord-scale_system

Miles Davis's Lydian Chromatic Concept-influenced first modal jazz album Kind of Blue, is often given as an example of chord-scale relationships in practice.[11]

Essential considerations of a style such as Charlie Parker's, including "rhythm, phrase shape and length, dynamics, and tone color," as well as "passing tones, appoggiatura, and 'blue notes'" are unaddressed.[13] This appears to have led educators to emphasize a specific repertoire of pieces most appropriate to the chord-scale system, such as John Coltrane's "Giant Steps", while excluding others, such as Coltrane's later styles of composition, and producing generations of "pattern" players among college-educated musicians.[13]

 

[RoA]

I adopt a "declarative" (computer science sense) approach to specifying a speaker. The goal is to abstract a studio monitor as a functional unit that best lends itself to music generation (in the sense of Do-Re-Mi scales). The basic principles might be traceable to LS3/5A. I feel the choices are simplified and naturally fall out after the parameters are "declared". Because I lack this knowledge, I have to hunt down each brand and compare them.

For example:

• Enclosure
  • Sealed
  • Passive radiator
  • Ported
    • Slotted, vs. round
    • Front, vs. rear
• Crossover
  • 2-way vs. 3-way
  • active vs. passive
  • frequencies
• Input/Output Connections
  • XLR vs RCA line input
  • Digital vs. analog
• Driver characteristics
  • magnet type (e.g., neodymium)
  • paper vs. metal
  • size
  • traditional vs. accordion
  • responsiveness
• Digital (DSP) vs. analog based
• RME ADI-2 DAC compatibility

I know it seems unfeasible and far fetched ... but what about listening to a few speakers?

 

[voodooless]

I adopt a "declarative" (computer science sense) approach to specifying a speaker. The goal is to abstract a studio monitor as a functional unit that best lends itself to music generation (in the sense of Do-Re-Mi scales). The basic principles might be traceable to LS3/5A. I feel the choices are simplified and naturally fall out after the parameters are "declared". Because I lack this knowledge, I have to hunt down each brand and compare them.

For example:

• Enclosure
  • Sealed
  • Passive radiator
  • Ported
    • Slotted, vs. round
    • Front, vs. rear
• Crossover
  • 2-way vs. 3-way
  • active vs. passive
  • frequencies
• Input/Output Connections
  • XLR vs RCA line input
  • Digital vs. analog
• Driver characteristics
  • magnet type (e.g., neodymium)
  • paper vs. metal
  • size
  • traditional vs. accordion
  • responsiveness
• Digital (DSP) vs. analog based
• RME ADI-2 DAC compatibility

I'd say, start a level higher. Almost all the things you specify are really contingent on the specific implementation to either make it work or produce a horrible speaker. None of these specs by themselves make a good speaker. In the end, what counts is frequency response off- and on-axis, coupled to that directivity, distortion characteristics and resonances and possible maximum SPL (but you don't seem to care that much about that). Those are the things you should be looking for.

 

[mononoaware]

https://en-de.neumann.com/kh-310-a#description

Its closed design enables a very fast transient response.
The closed cabinet design means a lower group delay. This results in almost the entire frequency range being reproduced at the same time; the transient response is precisely controlled at all times.

The KH 310 is a three-way

• tri-amplified monitor with thermally protected
• Class AB amplifiers (210 W and two times 90 W).

XLR electronically balanced analogue input

• Standard interfacing to professional equipment
• Wide range input gain and output level controls
  • Easier interfacing with signal sources

Controller design Analog; active
Crossover frequency 650 Hz and 2.0 kHz
Crossover slope 24 dB/oct; 4th order

Equalization: Bass0; -2.5; -5; -7.5 dB
Equalization: Low-Mid 0; -1.5; -3; -4.5 dB
Equalization: Treble+1; 0; -1; -2 dB

Not sure where you are going with this. . .
KH310 is a great option, but I thought you were limiting your options by the maximum dimensions?

If your 2x REL T-Zero are compatible with the rest of the setup (monitors/DAC), you basically already have the bass drivers, and depending on roll-off/cross-over point could get by with the Genelec 8020d/8030c + 2x REL T-Zero.

Of course if your budget allows, by all means go with something like HEDD Type 20 and maybe you can find a way to use the REL’s to cancel out some possible low-frequency room excitement caused by the Type 20.

Important: HEDD Type 20 (Mk1) is analog through and through, but HEDD Type 20 Mk2 uses DSP, as it has the built-in HEDD Lineariser.

Although a professional customers opinion would say there is no audible difference between both monitors when using the Analog XLR inputs (Analog and DSP models will be equally transparent using XLR inputs).

 

[mel]

HEDD has some attractive features. This brings me back to declarative statement.

https://hedd.audio/studio-monitors-mk2/
https://hedd.audio/wp-content/uploads/HEDD-Type-05-07-Manual.pdf

2-Way-System with 5” Woofer and High-Resolution HEDD AMT Tweeter

+ Integrated Amplifiers with 2x100W for up to 112 dB SPL/pair

+ Frequency Response: 45Hz–40kHz

+ Closed-or-Ported Functionality (I really like this feature!)

• In closed mode the HEDD MK2 speaker functions as an infinite baffle speaker.
• This results in a cleaner sound and increased resolution.
• However, a 6-10 dB reduction in maximum sound pressure level (depending on the audio material) for low frequencies is the consequence.

+ Integrated DSP with AD/DA: 96 kHz, 32 Bit (why this combination?)

+ HEDD Lineariser® Phase-Correction
Due to the physically given relationship between time and frequency, expressed in the groundtaking harmonic analysis or Fourier transform, different tones have slightly different travelling times through loudspeakers. This is true for both active or passive speakers. Broadly speaking, lower frequencies need more time to pass through than midrange frequencies, and these are slower than the high tones.
Our HEDD Lineariser® offers a complex phase-correction of your HEDD MK2 monitors. It solves one of the most challenging problems in loudspeaker designs, e.g. the natural occurance of phase difference during multi-component sound reproduction. With the on-board Lineariser®, our monitors are always “right in time”.
The phase-linearisation will improve transients and imaging. To be noted: the correction necessarily will introduce a latency of around 12 ms.

To avoid the delay that might affect live recordings or instrument playing the HEDD Lineariser® can be switched off.

Desk Filter
Monitors sitting on meter bridges or desks may suffer by early reflections from the desk surface that potentially blur the sound. The “Desk Filter” is a 3-position PEQ (Parametric Equaliser) that minimizes the effect.

Please note that we cannot give an exact recommendation because the acoustical environments play a vital role. Please find out in how far these filters may improve the sound.

+ Adjustable LFE, Shelf and Desk Filters

+ MSRP: 1,300€ / pair incl. 19% VAT ($1,500)

 

[mel]

Not sure where you are going with this. . .
KH310 is a great option, but I thought you were limiting your options by the maximum dimensions?

If your 2x REL T-Zero are compatible with the rest of the setup (monitors/DAC), you basically already have the bass drivers, and depending on roll-off/cross-over point could get by with the Genelec 8020d/8030c + 2x REL T-Zero.

Of course if your budget allows, by all means go with something like HEDD Type 20 and maybe you can find a way to use the REL’s to cancel out some possible low-frequency room excitement caused by the Type 20.

Important: HEDD Type 20 (Mk1) is analog through and through, but HEDD Type 20 Mk2 uses DSP, as it has the built-in HEDD Lineariser.

Although a professional customers opinion would say there is no audible difference between both monitors when using the Analog XLR inputs (Analog and DSP models will be equally transparent using XLR inputs).

Thanks. I appreciate the help.

 

[mel]

I'd say, start a level higher. Almost all the things you specify are really contingent on the specific implementation to either make it work or produce a horrible speaker. None of these specs by themselves make a good speaker. In the end, what counts is frequency response off- and on-axis, coupled to that directivity, distortion characteristics and resonances and possible maximum SPL (but you don't seem to care that much about that). Those are the things you should be looking for.

To me, audio is all about compromise. One feature offsets another. My attitude is to lay out the spectrum of possibilities. Detail the properties. Step back and look at the product from a holistic standpoint.

 

[mel]

You are missing at least listening space boundaries, or simply: room.
Typically the best possible speaker is calibrated to be sufficiently flat in anechoic room on-axis at least. In real room there are a lot of reasons not to have flat responce at listening point, and equalization without understanding what is really going on is a bad idea.
I think you should read more about speaker qualities and their intraction with environment because with such unstable position you can easily make suboptimal decision and disappoint in this hi-fi woodoo while mentioned amount of money is enough to get really good speakers and live with them happily.
PS "Neutrality" also can be different. Anything expensive should be auditioned personally IMO.

OK. I agree with your point from a conceptual standpoint. However, I tweaked the RME ADI DAC PEQ and improved my the AE3 sound quality in my room. I must have roughly approximated what room correction does through code.

I really get how powerful sound reflection is. I knew about DSP theory over a decade ago from my AV receiver. I mucked with the microphone, then. I am a proponent of DSP room correction, don't get me wrong.

I live in Denver close to Union Station, which is a transportation hub. Trains and light rail are prominent features of our environment, because they are a couple of blocks away. The train horn warning, that seems so loud, comes from behind my building and reflects off the building across the street. Everything in this area is oriented around trains and water.

 

[mel]

Not sure where you are going with this. . .
KH310 is a great option, but I thought you were limiting your options by the maximum dimensions?

If your 2x REL T-Zero are compatible with the rest of the setup (monitors/DAC), you basically already have the bass drivers, and depending on roll-off/cross-over point could get by with the Genelec 8020d/8030c + 2x REL T-Zero.

Of course if your budget allows, by all means go with something like HEDD Type 20 and maybe you can find a way to use the REL’s to cancel out some possible low-frequency room excitement caused by the Type 20.

Important: HEDD Type 20 (Mk1) is analog through and through, but HEDD Type 20 Mk2 uses DSP, as it has the built-in HEDD Lineariser.

Although a professional customers opinion would say there is no audible difference between both monitors when using the Analog XLR inputs (Analog and DSP models will be equally transparent using XLR inputs).

I calibrated my AE3s to work at a longer distance using my RME ADI-2. The trouble I had with the AE3s is they previously only generated euphonic sound at three feet. At six feet, the sound quality dropped substantially. Now, I could simply move my AE3s into my bedroom, due to calibration.

Previously, I was looking for bedroom speakers. Now, I am looking for speakers adjacent to my desk. The room is approximately 13m x 5m.

My priority is small speakers, nonetheless. I might move a little further from my desk, but prefer to remain at my desk. The smaller the speaker with the same sound quality, the better, in my opinion.

 

[mel]

Not sure where you are going with this. . .
KH310 is a great option, but I thought you were limiting your options by the maximum dimensions?

If your 2x REL T-Zero are compatible with the rest of the setup (monitors/DAC), you basically already have the bass drivers, and depending on roll-off/cross-over point could get by with the Genelec 8020d/8030c + 2x REL T-Zero.

Of course if your budget allows, by all means go with something like HEDD Type 20 and maybe you can find a way to use the REL’s to cancel out some possible low-frequency room excitement caused by the Type 20.

Important: HEDD Type 20 (Mk1) is analog through and through, but HEDD Type 20 Mk2 uses DSP, as it has the built-in HEDD Lineariser.

Although a professional customers opinion would say there is no audible difference between both monitors when using the Analog XLR inputs (Analog and DSP models will be equally transparent using XLR inputs).

I am thinking of keeping the RELs with the AE3s. I would like to buy new speakers with enough bass response that do not require a subwoofer.

I am generally opposed to DSP inside a speaker, but I am keeping an open mind. I don't like the idea of technology-dependent components inside a speaker. I prefer for electronics to be external. I don't know enough to conclusively exclude them. My guess is DSP inside a speaker will eventually eliminate it.

I was more impressed by the HEDD from the speaker as a functional unit perspective.

I need to understand how AMT and traditional tweeter performance compares.

• Do AMT reproduce sound that more closely resembles human voice?
  • My guess is yes, because they function like the human voice tract or an accordion, but I do not know, yet.
  • Are AMT better at reproducing the human sounds listed in the following quotes?
    • The seven sounds listed below are easier to judge.
    • The seven sounds are also "regular expressions" (in computer science sense)
  • Reproducing human or animal sounds is a criteria that I prefer to use.
    • Specific instruments like piano or drums, that have a full frequency spectrum are also high on my list.

The human voice consists of sound made by a human being using the vocal tract, including

• talking, ("ouch" expresses pain)
• singing, ("Pa rum pum pum pum" from Little Drummer Boy, articulates drum sound)
• laughing, ("ha, ha, ha" expresses enjoyment)
• crying, ("wah, wah" expresses hurt)
• screaming, ("ahhhh" expresses fear)
• shouting, or
• yelling.

The human voice frequency is specifically a part of human sound production in which the vocal folds(vocal cords) are the primary sound source. (Other sound production mechanisms produced from the same general area of the body involve the production of unvoiced consonants, clicks, whistling and whispering.)

https://en.wikipedia.org/wiki/Human_voice

Regular expressions (computer science sense) in parenthesis

https://en.wikipedia.org/wiki/Phonation#Voicing

The sound that the larynx produces is a harmonic series. In other words, it consists of a fundamental tone (called the fundamental frequency, the main acoustic cue for the percept pitch) accompanied by harmonic overtones, which are multiples of the fundamental frequency.[3] According to the source–filter theory, the resulting sound excites the resonance chamber that is the vocal tractto produce the individual speech sounds.

https://en.wikipedia.org/wiki/Sonorant

In phonetics and phonology, a sonorant or resonant is a speech sound that is produced with continuous, non-turbulent airflow in the vocal tract; these are the manners of articulation that are most often voiced in the world's languages. Vowels are sonorants, as are nasals like [m] and [n], liquids like [l] and [r], and semivowels like [j] and [w]. This set of sounds contrasts with the obstruents (stops, affricates and fricatives).[1]

For some authors only the term resonant is used with this broader meaning, while sonorant is restricted to consonants, referring to nasals and liquids but not vocoids (vowels and semivowels).[2]

 

[mel]

I know it seems unfeasible and far fetched ... but what about listening to a few speakers?

Because I want to know what the speaker design is intended to accomplish. I want to be clear about my own goals.

When I contacted the dealer, who I bought my audio equipment from, he told me Elac Navis are special order only. No demos.

I am reluctant to spend more than 15 in a store, due to covid. I am vaccinated, but the Delta Plus variant is a threat.

I need to keep the auditions down to 15 minutes, maximum. So, I really need to be focused and organized.

I need to understand how to use my RME DAC. I plan to bring my DAC to the audition.

 

[mononoaware]

The room is approximately 13m x 5m.

That is big enough for longer distance listening (mid-field).
Powerful 3-way box (HEDD/Neumann) will be excellent for their extra headroom (SPL) capabilities.
HEDD may be the safer choice due to option of being ported design.
Also I like to recommend AMT, but I may be biased here since I am currently listening to Monitors with AMT.
Basic understanding of AMT is larger radiating surface, therefore more efficient than conventional dome.
(I think less thermal/kinetic stress therefore better reliability and transients)