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

[mel]

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.





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".

I like to operate from the standpoint of models. Here is a model that might have general applicability, including rooms.

https://en.wikipedia.org/wiki/Source–filter_model

The source–filter model represents speech as a combination of a sound source, such as the vocal cords, and a linear acoustic filter, the vocal tract. While only an approximation, the model is widely used in a number of applications such as speech synthesis and speech analysis because of its relative simplicity. It is also related to linear prediction. The development of the model is due, in large part, to the early work of Gunnar Fant, although others, notably Ken Stevens, have also contributed substantially to the models underlying acoustic analysis of speech and speech synthesis.[1] Fant built off the work of Tsutomu Chiba and Masato Kajiyama, who first showed the relationship between a vowel's acoustic properties and the shape of the vocal tract.[1]

An important assumption that is often made in the use of the source–filter model is the independence of source and filter.[1] In such cases, the model should more accurately be referred to as the "independent source–filter model".[citation needed]

 

[mononoaware]

To me, audio is all about compromise. One feature offsets another.

This is correct.
Every design has its compromise.
There is no “perfect” speaker.
You must decide which compromise you wish to accept.

 

[voodooless]

I plan to bring my DAC to the audition.

Then make sure to call ahead. If you really want to stay only 15 minutes, you'll need a significant part of that time to set up. If you then also have to bargain first to actually make it happen, no time will be left.

 

[mel]

That is big enough for longer distance listening (mid-field).
Powerful 3-way box (HEDD/Neumann) will be excellent or 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)

I might be able to take advantage of mid-field. I am a music butterfly, rather than a stationary audiophile. I tend to listen in the background. Occasionally, when I hear something remarkable, I briefly focus on it intently. Afterwards, I resort to background listening.

Near-field is my preferred listening style. I like to listen to acoustic (jazz, classical), at voice level volume, in smaller spaces.

 

[mel]

This is correct.
Every design has its compromise.
There is no “perfect” speaker.
You must decide which compromise you wish to accept.

Exactly! I am wiling to forego a short audition, because it has little benefit and more risk.

I am willing to make a sight-unseen, sound-unheard purchase, if I can very carefully distinguish between speakers.

 

[mononoaware]

The smaller the speaker with the same sound quality, the better, in my opinion.

Just to clarify here, in terms of small room with smaller speakers + 2x REL T-Zero you will need to achieve precise cross-over between speakers and subwoofers.
e.g. high-pass to monitors via XLR, and low-pass to subwoofers

Not sure if this can be achieved for XLR and RCA outputs separately with RME ADI-2 DAC.

For large room with larger listening distances, typically you will need larger speakers.
I do all my listening near-field in small room so I do not have much experience with larger rooms.

 

[mel]

This is correct.
Every design has its compromise.
There is no “perfect” speaker.
You must decide which compromise you wish to accept.

I want to compromise towards "human", rather than "studio editing" features.

I am OK with a studio monitor if I can use the ADI DAC to "smooth" or "voice transform" emotionless speaker qualities.

 

[mononoaware]

I am wiling to forego a short audition, because it has little benefit and more risk.

The main benefit of home audition, is it is exponentially superior to store audition.
I still think there is high value in home audition.

But it also helps to narrow your options down to 2 or 3 models first.

And maybe going forward you should clarify which setup you are talking about:

1. Near-field, small room

2. Mid-field, large room

 

[mel]

Just to clarify here, in terms of small room with smaller speakers + 2x REL T-Zero you will need to achieve precise cross-over between speakers and subwoofers.
e.g. high-pass to monitors via XLR, and low-pass to subwoofers

Not sure if this can be achieved for XLR and RCA outputs separately with RME ADI-2 DAC.

For large room with larger listening distances, typically you will need larger speakers.
I do all my listening near-field in small room so I do not have much experience with larger rooms.

You allude to an important pre-amp requirement.

The AE3 has a pre-amp built inside the speaker with RCA inputs and outputs. I connect the DAC to the inputs. I connect the subwoofers from the RCA outputs.

 

[mel]

The main benefit of home audition, is it is exponentially superior to store audition.
I still think there is high value in home audition.

But it also helps to narrow your options down to 2 or 3 models first.

And maybe going forward you should clarify which setup you are talking about:

1. Near-field, small room

2. Mid-field, large room

I can reposition things in my room to accommodate two pairs of speakers. My habit is to listen in near field. My preference is to leave my AE3s on my desk. Place the new speakers about four feet from my desk, towards the window. Listen in mid field from the other side of my desk. It just means that I need to stand up and move away from my desk when something remarkable plays.

 

[mononoaware]

You allude to an important pre-amp requirement.

The AE3 has a pre-map built inside the speaker with RCA inputs and outputs. I connect the DAC to the inputs. I connect the subwoofers from the RCA outputs.

So the AE3 has fixed cross-over/low-pass.
I am wondering if the RME DAC has adjustable high & low-pass filters for its XLR and RCA outputs, which maybe possible with RME’s advanced features.
(Set RCA low-pass to 80hz and below, set XLR high-pass 80hz and above)

 

[raindance]

So the AE3 has fixed cross-over/low-pass.
I am wondering if the RME DAC has adjustable high & low-pass filters for its XLR and RCA outputs, which maybe possible with RME’s advanced features.
(Set RCA low-pass to 80hz and below, set XLR high-pass 80hz and above)

I decided not to purchase an RME because it cannot do this.

 

[mononoaware]

Near-field is my preferred listening style. I like to listen to acoustic (jazz, classical), at voice level volume, in smaller spaces.

In my experience with near-field in small-room with soft floor.
And even moving in this small room with soft floor is non problematic as your mind gets used to the sound of the monitors + room position and reduces your changes in perception.

I imagine though in a large-room listening mid-field, the room effects may become powerful enough your perception cannot ignore, and you will start thinking about room treatment etc.
This is especially horrible if you have a large room with hard floor.

 

[mononoaware]

I decided not to purchase an RME because it cannot do this.

Good information raindance thanks.

 

[raindance]

Good information raindance thanks.

Right now I'm experimenting with a Dayton DSP408 with encouraging results.

 

[mel]

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)

I don't know any of these products, so I take a philosophical approach. I care more about aligning the speakers along these criteria. The closer I can come, the better.

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/

 

[mel]

Right now I'm experimenting with a Dayton DSP408 with encouraging results.

To get "voice smoothing or transformation", I might need more than the five PEWQ bands on my RME DAC.

• 10-band parametric equalizer per channel
• High and low shelf EQ per channel
• Variable crossovers: High/Low/Band-Pass slopes 0-24dB, Linkwitz Riley/Butterworth/Bessel

 

[mel]

Right now I'm experimenting with a Dayton DSP408 with encouraging results.

Do you have a guiding principle you use to set the PEQ, or do you configure by ear or microphone?

I used the following chant (Do-Re-Mi scale or Ut queant laxis), but a different Tidal recording, to calibrate my AE3 speakers from my RME DAC. The PEQ curve somewhat resembles the human voice frequency response curve.

• I wonder how a flat PEQ curve playing this chant sounds on your system.
• Does the chant sound any different if:
  • you very approximately set the crossover frequencies to correspond to the human voice frequency response curve in the previous post? A sine wave is close enough for my "smoothing" purposes.
    • More natural?
    • More euphonic?
    • More bright, dark, etc...?
  • If not, what PEQ settings made the sound appear a certain way?

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

To judge music reproduction, I like MTT. He is preeminent in classical recording. I held season tickets for years at SFS, so the hall acoustics are very familiar to me. I can tell how a speaker performs, by the reverberation from the music hall.

Some sounds, that I think are easier to judge by than music follow. I am interested in comparing AMT and traditional tweeters by these criteria.

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

 

[voodooless]

I don't know any of these products, so I take a philosophical approach. I care more about aligning the speakers along these criteria. The closer I can come, the better.

That stuff you quoted is about audio production, not re-production. The mastering engineers already did all that for you. You should not need to do it again.

 

[mel]

That stuff you quoted is about audio production, not re-production. The mastering engineers already did all that for you. You should not need to do it again.

They can do a much better job than me. I am trying to internalize the concepts.

This is how I learn. Gordon suggests that "audiation is to music what thought is to language."[

https://en.wikipedia.org/wiki/Gordon_music_learning_theory#Audiation