• 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 daily reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

Meyer Sound Amie Monitor Review

Rate this speaker:

  • 1. Poor (headless panther)

    Votes: 8 3.0%
  • 2. Not terrible (postman panther)

    Votes: 37 13.8%
  • 3. Fine (happy panther)

    Votes: 162 60.2%
  • 4. Great (golfing panther)

    Votes: 62 23.0%

  • Total voters
    269

changer

Addicted to Fun and Learning
Joined
Dec 4, 2020
Messages
555
Likes
598
Well,
And back to my main and in fact only point which is being ignored. A fixed parameter auto-EQ process "maybe" should not be a trusted sound score attribute for speaker comparison.

you are free to do it by hand. Optimizing algorithms are a thing, and you would also need to decide if you weight-in the room and reflexions into the speaker transfer function (before room EQ), something engineers do since decades. The decision between On, LW, In-Room remains, and a blind test for comparison is still required.

Please note that you are mixing things up: what applies to room correction is not the same as what we are talking. These EQs are for anechoic response. Its perfectly fine to EQ at any frequency within the speaker passband, that is just the same what an analogue crossover does.
Why not include XO phase correction and add it to the score then? Why limit digital correction to IIR filters only?

How could you EQ the crossover of a multi-way speaker? That is simply not possible, unless you removed the crossover and replace it with another.
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
you are mixing things up: what applies to room correction is not the same as what we are talking. These EQs are for anechoic response
Maybe you're mixing up a little bit. I wrote a whole paragraph on exactly that!

How could you EQ the crossover of a multi-way speaker? That is simply not possible
Have you heard of a free tool called rePhase? It's mainly a crossover phase linearization tool and very popular among especially the DIY crowd ;)
 

changer

Addicted to Fun and Learning
Joined
Dec 4, 2020
Messages
555
Likes
598
Does not work without prior replacement of the original crossover with an active crossover (that can do FIR).
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
Does not work without prior replacement of the original crossover with an active crossover (that can do FIR).
So now the "with EQ" score is about the capacity of the DSP engine inside the active speaker in question???

That would have actually made some sense for comparison purposes. But passive speakers do exist, some of us live at homes rather than studios, none of the active speakers tested have PEQ capacity nearly as much as the parameters used in the method requires and it's very possible and indeed way more common to apply convolution filters to a system elsewhere rather than inside the speaker.
 

reficver

Member
Joined
Feb 19, 2023
Messages
14
Likes
7
@OCA @changer @ernestcarl

It seems like pretty interesting stories.
Could you please post a new thread about the possibility of IIR/FIR applicable for EQ score and anechoic measurement?
Then I think a lot of people can participate.
 
  • Like
Reactions: OCA

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
@OCA @changer @ernestcarl

It seems like pretty interesting stories.
Could you please post a new thread about the possibility of IIR/FIR applicable for EQ score and anechoic measurement?
Then I think a lot of people can participate.
Or maybe a new column in the Spinorama named "Tonal score if you could make a headphone from a pair of these speakers" :)
 
Last edited:

changer

Addicted to Fun and Learning
Joined
Dec 4, 2020
Messages
555
Likes
598
@OCA EQ is for passive speakers and active speakers, but with neither can you change the crossover filters. Which is why the Power Response cannot be altered, nor how the filters affect directivity in the crossover region.
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
@OCA EQ is for passive speakers and active speakers, but with neither can you change the crossover filters. Which is why the Power Response cannot be altered, nor how the filters affect directivity in the crossover region.
Ok, I agree with that part but every passive/active crossover will cause phase shifts and time delays between drivers depending on the filter type/order and this can be corrected with time-reversed first/second degree allpass filters quite easily. Same is true for also box/port phase shifts.

What speakers and source do you use for music playback if you don't mind me asking? I can send you a custom phase linearization FIR filter in .wav (or .bin) format and you can hear the difference in sound for yourself.
 
Last edited:

ernestcarl

Major Contributor
Joined
Sep 4, 2019
Messages
3,071
Likes
2,279
Location
Canada
@OCA @changer @ernestcarl

It seems like pretty interesting stories.
Could you please post a new thread about the possibility of IIR/FIR applicable for EQ score and anechoic measurement?
Then I think a lot of people can participate.

I understand the interest... But, to be honest, I don't equalize to maximize the preference scoring algorithm. I equalize to maximize measured linearity across multiple listening positions in a given room space -- without boosting excessively to avoid incurring distortion and headroom loss -- this does involve at least having a very good idea and ability to measure the the speaker's on- and off-axis frequency response.
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
without boosting excessively to avoid incurring distortion and headroom loss
I will add to that "without cutting aggresively above a certain frequency" which will cause audible degradation in the sound even if the frequency amplitude seems to be corrected. John/Room EQ Wizard came up with auto-EQ decades ago before anyone else (or at least made it freely available to public before anyone else) and has been optimizing it ever since and I don't know what kind of checks are being done in the algo but (and correct me if I am wrong) REW will refuse to apply such strong (if any) filters above 15kHz no matter what parameters you choose.

Also, you only correct for anechoic response (the speaker stripped off of room reflections) in the HF anyway so the discussion of anechoic vs in room response is not a valid explanation for the robustness of "with EQ" tonal scores IMO.
 

dominikz

Addicted to Fun and Learning
Forum Donor
Joined
Oct 10, 2020
Messages
779
Likes
2,531
Also, you only correct for anechoic response (the speaker stripped off of room reflections) in the HF anyway so the discussion of anechoic vs in room response is not a valid explanation for the robustness of "with EQ" tonal scores IMO.
Just to clarify the terminology, when you say "anechoic" do you actually mean "on-axis"?

Because in the context of spinorama all of the derived curves are anechoic - not just the on-axis. So when we say we generate EQ based on anechoic data we can (and do) use various polar responses to optimize the radiation in all directions (and therefore consequently we can optimize spectrum of early reflections as well).
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
Just to clarify the terminology, when you say "anechoic" do you actually mean "on-axis"?

Because in the context of spinorama all of the derived curves are anechoic - not just the on-axis. So when we say we generate EQ based on anechoic data we can (and do) use various polar responses to optimize the radiation in all directions (and therefore consequently we can optimize spectrum of early reflections as well).
anechoic response (the speaker stripped off of room reflections)
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
therefore consequently we can optimize spectrum of early reflections as well
I am not so sure having only the off axis angles for the floor, ceiling, side walls and back/front walls but not the distances will lead to a placement-free improvement in the sound of that speaker. I need to study that a bit more.
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
So do you consider spinorama ER or PIR curves anechoic or not? :)
I think they are better then even actual anechoic measurements. I also fully support tonal scores, even "with sub" scores for comparison. Because of this forum, I have started looking into active studio monitors for some time now and checking spinorama regularly lately. I just cannot get comfortable with the "with EQ" score logic.
 
Last edited:

changer

Addicted to Fun and Learning
Joined
Dec 4, 2020
Messages
555
Likes
598
Well, then please study the EQ suggestions that @Maiky76 and @pierre have delivered so far for various speakers, usually to be found within the first couple of pages of a review thread, the PEQs are open to view, and make a step from not being comfortable to a grounded critique, based on concrete examples.

The research that culminated in ANSI/CTA-2034-A found that linear distortion is the single most important factor, as we notice nonlinearities in the frequency response easily, and that a smooth Directivity and Power Response together with linear frequency response create a speaker that is perceived as better in double-blind tests.

What EQ-ing “after the fact,” of a speaker with a given crossover can achieve is taking care of linear distortion and it will greatly improve a speaker if it has deficiencies in this area. Given a nonlinearity wasn't deliberately created to alleviate a Sound Power issue. It is certainly debatable how linear a frequency response must be to sound very good, but if we can achieve +-1.5 dB from 20-20000 Hz, that's certainly a high value.

Also, you only correct for anechoic response (the speaker stripped off of room reflections) in the HF anyway so the discussion of anechoic vs in room response is not a valid explanation for the robustness of "with EQ" tonal scores IMO.

No, you correct the full speaker. Think of the EQ based on the spinorama as if you would make a speaker, but you work in a team and the other department already decided which crossover slopes, drivers and bass tuning must be used. And your team is supposed to implement notches, attenuations and so forth, until the circuit is completed. This is what the EQ does. And only after this follows room equalization that is restricted to the modal region.
 
Last edited:

dominikz

Addicted to Fun and Learning
Forum Donor
Joined
Oct 10, 2020
Messages
779
Likes
2,531
I think they are better then even actual anechoic measurements.
ER and PIR are directly calculated from anechoic responses, so despite what the names of the curves may imply, they are still anechoic data.
So when we say we can calculate loudspeaker EQ corrections based on anechoic data, we can do so while optimizing for the spectrum of direct sound (ON or LW), early reflections (ER) and/or steady-state in-room response (PIR).
Also, you only correct for anechoic response (the speaker stripped off of room reflections) in the HF anyway so the discussion of anechoic vs in room response is not a valid explanation for the robustness of "with EQ" tonal scores IMO.
See this example for an illustration how in a loudspeaker with smooth directivity (Revel M16) to see how we can optimize on-axis as as well as all lateral radiation responses with PEQ:
index.php

Basically by applying appropriate PEQ filters we're playing loudspeaker designer :)
Note that many powered loudspeaker designs already have similar filters built-in and I suppose you accept the validity of the score in those cases - this is really not that different!
Of course, we should note that DSP-capable powered speakers usually have the crossover implemented in DSP as well, which definitely gives more control to the speaker designer and can result in better overall designs.
I am not so sure having only the off axis angles for the floor, ceiling, side walls and back/front walls but not the distances will lead to a placement-free improvement in the sound of that speaker. I need to study that a bit more.
Firstly - spinorama (link: ANSI/CTA-2034-A) is comprised of more than just the few early reflection angles that you mention - there is a total of 70 responses measured over two planes to get an estimate of the full 360° radiation pattern of a loudspeaker. E.g. the Sound Power (SP) curve comprises of responses radiated in all measured angles. I can only suggest to look at the standard to see how each individual curves are calculated, as well as some explanations.

Secondly, the measured in-room response in various rooms (and for various listening distances) seems to closely correlate to the PIR calculated from anechoic data above ~1kHz in basically any example I've encountered so far, including my own measurements. IMHO that gives the method a lot of validity.

Some examples:
Nearfield distance (~80cm), with non-symmetrical placement in a small untreated room:
index.php

Midfield distance (2,3m) in a slightly larger untreated room:
index.php

Measurement at two listening distances by @hardisj:
PIR%20vs%20MIR.png


I am not so sure having only the off axis angles for the floor, ceiling, side walls and back/front walls but not the distances will lead to a placement-free improvement in the sound of that speaker. I need to study that a bit more.
Distance mainly affects high frequencies (above ~10kHz) in the direct sound due to effect of air absorption, see chapter 10.6 of the 3rd edition of "Sound Reproduction: The Acoustics and Psychoacoustics of Loudspeakers and Rooms" for more details.

Furthermore, high frequencies in the measured in-room room response (examples above) will be further affected by:
  1. Calculated PIR assumes certain characteristic of the room, while in real rooms the amount of absorption will vary, which can result in some attenuation in high frequencies compared to PIR calculated from anechoic data.
  2. Measurement microphones become directional at high frequencies, and so can only be calibrated to flat for some incidence angles. This means that some early reflections can be captured with a non-flat spectrum and influence the measured steady-state in-room response. The effect will be less severe if vertical microphone orientation is used with a 90° calibration file - I wrote about this previously here.
I just checked the post and I generally refrain to disagree with others in these forums but anyone with some real life experience in DRC will know that IIR filters like a -7dB cutoff at 15kHz (highlighted below) are very unlikely to produce sound improvements in a room. In practice, you don't want to touch above 1kHz unless you have a very good reason and some very robust IR gating method. The treble will sound throttled at best and with a wavelength of 2.2 cm (shorter than the distance between your ears) for 15kHz, the dependence of the benefit of the filter on speaker directivity is also very questionable IMO.

View attachment 321483
Well for any reasonably-well-performing loudspeaker I'd agree. But this one has such a severely mangled response that severe filters are indeed required to get it to sound remotely close to flat. For me it was a very interesting experiment that would be less educational with loudspeakers that measure well out of the box!
With well-measuring loudspeakers I find EQ is only required to correct the in-room bass response (below the transition frequency); with such loudspeaker the minimal potential gains of correcting the loudspeaker based on anechoic data are IMHO not really worth the effort!
I just cannot get comfortable with the "with EQ" score logic.
Personally I believe that the score should be approached with caution because we anyway know there are limits to its precision - even if overall corelation to preference is robust. It is absolutely a very useful tool for rough sorting and filtering though, but when we distil so much measurement information into just one number we're bound to lose some valuable information.
I typically see the various score variants as follows:
  • If a speaker's score improves significantly with EQ it means it probably has good directivity and is limited by that (and vice-versa)
  • If a speaker's score improves significantly with sub it means it probably doesn't have a lot of LF extension and is limited by that (and vice-versa)
So I find these variants quite useful for quick reference, but I'd always look at the full measurement suite when shortlisting or just analysing loudspeaker performance!
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
Thank you for all that information. I've also watched a couple of Erin's videos yesterday and learned a bit about how ER and PIR are derived. Obviously, there're lots of approximations and average room size/reflection/volume and parameter weight assumptions in the calculations but since the same parameters are used for every speaker, I am convinced it's a robust way of quantifiying a speaker's performance. I have also learned it's real world accuracy is limited to between 1-2Khz and 10kHz.

A couple of comments on your notes:

how we can optimize on-axis as as well as all lateral radiation responses with PEQ
Check that 2.6dB booster at 13kHz. While it's flattening the on axis response, it's pumping up the already boosted signal on the horizontal axis. Center stage is an illusion (hence the name phantom image) created in our brains by merely the side reflections. That frequency will reflect from the side wall boosted and shift the sound of that signal more to that side than it needs to be as it's going to be louder than it should. In the imaginary stage, that signal was not coming from the speaker so correcting its SPL at the speaker didn't help anything and ruined its location at the imaginary sound stage we were hearing. Vertical directivity similarly determines the center stage height and vertical precision of sound locations. I could agree with you if the algorithm only corrected for problems that will be improved on axis, at ER and PIR. Some filters above are of that kind but some are not although I have seen ER/PIR parameters in the code generating these filters?

Lastly, most of the useful filters above are below 1-2kHz but it's also shown in the predicted vs actual comparison graphs that these will ALL need to be completely changed for every room. In fact, these graphs are great examples for why you shouldn't EQ above 1-2kHz.
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
630
Likes
456
Location
Germany
On a different note, I wonder why no phase data is being released from those Klippel measurements although that's already embedded in the stored impulse responses. Most of the deviations in ER and PIR relative to on-axis seem to be at speaker crossover regions and they could really benefit from XO phase linearization filters.
 

dominikz

Addicted to Fun and Learning
Forum Donor
Joined
Oct 10, 2020
Messages
779
Likes
2,531
Thank you for all that information.
No problem, I'm happy if people find some of it useful!
I have also learned it's real world accuracy is limited to between 1-2Khz and 10kHz.
I guess it depends on how accurate you expect it to be. Deviations of in-room measurements from the PIR are the smallest in the 1-10kHz range (i.e. almost perfect match), but even from about 400Hz to 20Hz we're mostly talking about a couple dB difference at most.
Many loudspeakers show unit-to-unit variation of a similar order, and here we're talking about comparing measurements taken in very different room and from quite different listening distances - so I'd dare say the PIR is amazingly accurate!
Check that 2.6dB booster at 13kHz. While it's flattening the on axis response, it's pumping up the already boosted signal on the horizontal axis.
Sure, but that is only because that specific EQ preset optimizes for LW flatness with 100% weight, and that specific loudspeaker model has inconsistent directivity in that region.
However, note that a) the side wall early reflection will cover a significantly larger distance compared to direct sound and therefore >10kHz region in the reflected sound will be more attenuated by air absorption, and b) >10kHz part of the spectrum contains very little energy in most recordings (which is also an argument no to EQ there), so perceptually speaking this might not really be a big issue.

Let me also reiterate that I feel good loudspeakers (such as Revel M16 I took as an example in my previous post) don't need such EQ correction.
Also, note that I actually developed three different EQ profiles for Revel M16 loudspeaker while doing this set of experiments, with different weights assigned to different attributes:
Flat LW target EQ:
Highest preference score target EQ:
70% flat LW and 30% smooth PIR target:
My listening impressions after testing all of these filters (and many more) were that, while all audibly changed the loudspeaker sound to an extent, IMHO none of them made it worse - but also not significantly better. This is because Revel M16 is simply a good speaker as is and IMHO doesn't require correction. But there are loudspeaker with good directivity and poor LW response - those can benefit from such EQ presets significantly! See one such example here.
Center stage is an illusion (hence the name phantom image) created in our brains by merely the side reflections.
Can you provide a reference to back this statement?
In my experience center phantom image exists even if side reflections are largely absorbed, because it is mainly created by the direct sound from both loudspeakers which sum and create the center image illusion. In very extreme cases (listening in anechoic chamber) in-head localization can occur, similar to headphones - but I assume there are not that many such residential spaces.
Side wall reflections can however influence the phantom image and soundstage size.
More side reflections result in a wider and more diffuse phantom images and a soundstage that extends laterally beyond the loudspeakers, while less reflections result in more precise/focused phantom images that are fully contained between the two loudspeakers.
That frequency will reflect from the side wall boosted and shift the sound of that signal more to that side than it needs to be as it's going to be louder than it should. In the imaginary stage, that signal was not coming from the speaker so correcting its SPL at the speaker didn't help anything and ruined its location at the imaginary sound stage we were hearing. Vertical directivity similarly determines the center stage height and vertical precision of sound locations.
Again, I'll have to ask for references. My own experience and the research I've read doesn't seem to align with these views, but I'd of course love to learn more in case such research exists!
I could agree with you if the algorithm only corrected for problems that will be improved on axis, at ER and PIR. Some filters above are of that kind but some are not although I have seen ER/PIR parameters in the code generating these filters?
As I said, that was only one example of the very many approaches that can be taken (some illustrated above) - unfortunately we don't have preference research yet that tell us how best to optimize such filters, so we have to rely on the existing loudspeaker preference research combined with our own experience and some conjecture - IMO it is better than nothing and can be quite valuable with certain loudspeakers!
The filters I show in this and previous post I generated myself with the VituixCAD optimizer and based on my own quasi-anechoic polar measurements.
Lastly, most of the useful filters above are below 1-2kHz but it's also shown in the predicted vs actual comparison graphs that these will ALL need to be completely changed for every room.
I disagree, the filters are definitely meaningful from about 3-500Hz to about 8-10kHz; in-room steady-state measured response doesn't accurately describe what we hear above the transition frequency (around 300Hz in many "small" rooms) - there we perceive more and more of the loudspeaker direct sound, which means that filters derived from anechoic measurements make sense in that region.
In fact, these graphs are great examples for why you shouldn't EQ above 1-2kHz.
I'll have to respectfully disagree, due to reasons stated above and before. :)
 
Last edited:
Top Bottom