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Cheapest Full Range 20hz - 20khz Speakers?

tuga

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The impedance and frequency response measurements fully characterise the behaviour of the port (within its linear range of operation - to go beyond that, we need compression testing). Assuming that peak in the Soundtsage! measurement is not a measurement artefact, we see the problem in the frequency response, anyway. Why would we (assuming we're not the designer) need to know what was causing it? A peak in the FR is a peak in the FR... right?

Re: your comments about damping, how will this impact the audible quality of the low-bass, other than by affecting its frequency response (which we see in the FR graphs)?
Can you tell if the tuning is over- or under-damped from an Impedance plot?
What if its response is broadband instead of a peak, will it show in a FR plot or an Impedance plot?

What can you tell about transient response from a FR plot?
What can you tell about transient response from a HD plot?
What can you tell about transient response from an Impedance plot?

Correct me if I'm wrong but from my understanding ports have worse transient response than sealed cabinets (visible in a CSD plot or perhaps less clearly in an Step Response plot - see here), and under-damped ports have worse transient response than over-damped ports.

Do you have an example of such an issue that you could share so I better understand what you're talking about? And do you have any evidence that such an issue could be audible even when there is no measurable anomaly in the amplitude response?
Some blips in the frequency response decay normally or as fast as the whole of the spectrum whilst other blips in the frequency response decay a lot slower (ring). This example is exaggerated for illustration purposes (I know that the blip in the FR is very high at ~5dB but it could be a higher-Q and 1dB, I'm more interested in the long tail; the one higher up at ~6.5kHz doesn't sustain hardly any more than the rest of the spectrum).
 
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andreasmaaan

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Can you tell if the tuning is over- or under-dampled from an Impedance plot?
What if it's response is broadband instead of a peak?
My question is, though, why do you need to know these things if you know what the frequency response is? What additional, useful information do they give you?
What can you tell about transient response from a FR plot?
What can you tell about transient response from a HD plot?
What can you tell about transient response from an Impedance plot?
Ported enclosures are, for practical purposes, minimum-phase systems (sure, there will also be some energy storage in the system, but it is not significant in comparison to the overall group delay picture). The so-called transient response and the amplitude response are not independent. The time-domain behaviour can be largely inferred from the amplitude response (and vice versa).

Correct me if I'm wrong but from my understanding ports have worse transient response than sealed cabinets (visible in a CSD plot or perhaps less clearly in an Impulse Response plot - see here), and under-damped ports have worse transient response than over-damped ports.
It's true that ported enclosures produce more group delay than sealed enclosures. But that's primarily because they roll off faster. Both closed and ported systems are essentially minimum phase, i.e. the group delay can (more or less) be inferred simply from the slope and frequency of the roll-off.

To date, there is no evidence (of which I'm aware) that the degree of group delay introduced by a ported enclosure is audible. There are plenty of theories that say it is, but you know my attitude towards such speculation by now, I guess :)

In case you need convincing, though, think about what is happening in a room at those frequencies. Generally speaking, wavelengths around the port tuning frequency are longer than typical room dimensions. We can't even make sense of these frequencies until they've hit room boundaries and bounced back at us. Just thinking about this from a common-sense point of view (which I normally would never advise anyone to do, but humour me), what do you think the chances are of hearing c. 360° (ported) vs c. 180° (sealed) worth of group delay at these frequencies?

Some blips in the frequency response decay normally or as fast as the whole of the spectrum whilst other blips in the frequency response decay a lot slower (ring). This example is exaggerated for illustration purposes (I know that the blip in the FR is very high at ~5dB but it could be a higher-Q and 1dB, I'm more interested in the long tail; the one higher up at ~6.5kHz doesn't sustain hardly any more than the rest of the spectrum).
Interesting CSD. Which speaker is it? My first thought is that we have a sharp crossover at exactly the frequency of the long-decaying peak.

Anyway, I'm sure the peak is audible. Do you have any basis for the belief that the decay is, too?

My view is that the best way to answer audibility questions like this is by looking at psychoacoustics research. This is what Zwicker and Fastl say about post-masking:
A Gaussian-shaped condensation impulse with a duration of only 20 µs produces a spectral shape that corresponds to that of white noise. It is similar to that of a Dirac impulse. Postmasking produced by a white-noise masker can therefore be measured without spectral influences using such a brief Gaussian impulse as a test sound. The peak value of this Gaussian impulse expressed in level is plotted as the ordinate in Fig. 4.22, which shows the level necessary to just reach threshold in postmasking as a function of delay time td from the end of the masker. The parameter in the figure is the overall level of the white-noise masker. The solid lines indicate results that show almost no decay for the first 5 ms after the masker is switched off. There, the values correspond to those measured in simultaneous masking. After about 5ms delay time, the threshold in postmasking decreases and, at about 200 ms delay time, reaches the threshold in quiet...

Postmasking depends on the duration of the masker. Figure 4.23 shows a typical result measured using, as a test sound in this case, a 2-kHz tone burst of 5-ms duration. Again, the time at which the test-tone burst is presented after the end of the masker is plotted as the abscissa. The level of the test-tone burst is the ordinate. For a masker duration of 200 ms, the solid curve indicates postmasking comparable to that displayed in Fig. 4.22. Quite different from that is the postmasking produced by a masker burst, which lasts only 5 ms, as indicated by the dotted line in Fig. 4.23. In this case, the decay is initially much steeper. This means that postmasking depends strongly on the duration of the masker and therefore is a highly nonlinear effect.
1605547806474.png


1605548193994.png


As you can see, even for a masker of only 5ms duration (which is short in comparison with transients you normally encounter in music), we have masking effectiveness at -20dB, even 5ms after the transient has ended.

Looking again at that CSD you posted, even that ugly-looking resonance is down to -20dB after about maybe 2ms. It's not even on the graph by 3ms, let alone 5ms:

1605548588014.png


FWIW, I've attached a WAV file containing a 5ms burst of white noise, just so you can verify that this is indeed a very quick transient (which moreover, unlike musical transients, has absolutely zero decay).

EDIT: I've also just added a file containing 200ms of white noise, as I thought that might also be illustrative.

WARNING: these are loud. Turn your system down before playing them.
 

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LS50's in the bedroom.
AudioEngines in kitchen, second pair in garage.
JBL708P's and 705P's in main listening room
Paradigm Studio 40's in office with the Devialet..
Misc Paradigm and Triad Speakers in downstairs theater.
JBL705P's on desk.
JBL306's in my son's room now as his keyboard speakers, but normally those are in the guest room.
2 pairs of outdoor speakers.

I think that's it.
Nothing set up in any bathrooms or closets ?
 
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richard12511

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See the problem here is that I'm asking you to prove to me what else is more important than what's already being measured. You're the one making the claims. You can't keep trying to flip it over to me to answer for you.
Yes, I don't think the step response will tell you anything you don't already know. Are you able to explain to me how it would? What other distortions do you think are more important than harmonic and possibly intermodulation? Do you know of any audibility studies for them? I'm genuinely curious.
I agree with this completely. The burden of proof that these things that Amir doesn't measure(like step response) are critical for loudspeaker assessment lies with the people trying to beget doubt in the currently established science.

No problem with beliefs that are contrary to established science. As @tuga rightfully pointed out, such beliefs are actually healthy for continued scientific advancement. The established science is not always correct. But, if you want to convince others that your personal beliefs should supersede the that science, then you need to back up those beliefs with evidence, just like Copernicus did. Without real evidence, it's going to be very difficult for you to change my mind, and I'm assuming others here feel the same way.

I think @Duke has done a good job thus far of at least trying to explain the reasons for his beliefs(I enjoyed that video).
 

tuga

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I agree with this completely. The burden of proof that these things that Amir doesn't measure(like step response) are critical for loudspeaker assessment lies with the people trying to beget doubt in the currently established science.

No problem with beliefs that are contrary to established science. As @tuga rightfully pointed out, such beliefs are actually healthy for continued scientific advancement. The established science is not always correct. But, if you want to convince others that your personal beliefs should supersede the that science, then you need to back up those beliefs with evidence, just like Copernicus did. Without real evidence, it's going to be very difficult for you to change my mind, and I'm assuming others here feel the same way.

I think @Duke has done a good job thus far of at least trying to explain the reasons for his beliefs(I enjoyed that video).
I wish I could di as good a job as Duke but alas...
 
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tuga

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To date, there is no evidence (of which I'm aware) that the degree of group delay introduced by a ported enclosure is audible. There are plenty of theories that say it is, but you know my attitude towards such speculation by now, I guess
As you know my knowledge and vocabulary are limited and I am not very good with technical terms but bear with me for a second.

Imagine a series of quick successive notes played on a kick drum.
The step reponse of a sealed cabinet shows a quick return to 0V whilst that of the ported cabinet continues to oscilate (ring) for a bit.
Won't this post-ringing affect the speaker driver's ability to accurately reproduce the ensuing note and the one after?





Transients (attack and decay) are an essential part of the timbre characteristics which make the sound of instruments disctinctive.


1. tabla (3 beats), 2. french horn (3 notes), 3. flute (1 note)


Frequency response does not provide any information regarding the speaker's ability to reproduce transients.
Nor would listening to steady state pink-noise for that matter.

Anyway, I'm sure the peak is audible. Do you have any basis for the belief that the decay is, too?
Only anecdotal experience with the old Monitor Audio Studio series from the early '90s.

Have you never listened to the effects of room resonances in the bass? Do you think it's only the peak that is audible or also the tail?
 
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richard12511

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Stereophile measures the response of individual drivers and port, have you never found it useful?
I guess it depends on what you mean by "useful". Useful for demonstrating problems? No. Useful for identifying the causes of problems? Yes.

For me, as a consumer, individual driver measurements are not very useful, and I tend to skip past them in Stereophile reviews. As a consumer, I don't necessarily care why problems exist, just that they do. I think they're more useful for speaker designers who are wanting to understand the "why".
 

infinitesymphony

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Speaking of transient response and group delay, I'm interested to see the opinions fly when we talk about our first speaker with a passive radiator. Here's a Mackie rep's response about the HR824:

"...Yes the output of the passive radiator is slightly delayed from the front fired wave. It does require some time to get the mass of the radiator moving. But it also requires some time (granted not as much) to get the air moving through a port. And a port that was long enough to do what the passive radiator does would create other issues if it was even possible..."
 

tuga

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Interesting CSD. Which speaker is it? My first thought is that we have a sharp crossover at exactly the frequency of the long-decaying peak.
You are probably right, it would have been my guess too (edit: you wrote sharp and I read shallow so not my guess), but unfortunately JA didn't produce the individual driver response plot.
But the specs can help:

Drive-units: ¾"-dome tweeter, 8"-cone woofer
Crossover frequency: 3.5kHz; first-order crossover to tweeter, second-order crossover to woofer

https://www.stereophile.com/content/spectrum-audio-108cd-loudspeaker-measurementss
 
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richard12511

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If the research were right no one would prefer narrow-dispersion speakers like horns or dipole panels or even D&Ds. Unless they think that they prefer it but they don't...
I agree with you here(and disagree with Toole). I've actually seen Toole list "wide dispersion" as the second most important factor(after flat LW) for predicting listener preference. Personally, my preferences have changed over time to preferring more narrow dispersion designs.

It is also my belief(based on some logic and my own blind listening tests) that mono tests bias the result in favor of wider dispersion designs. I also believe that their stereo tests(and perhaps why they always agreed with the mono tests) were similarly biased towards wider dispersion designs. Their machine puts every speaker in the same spot with equal toe in. I believe this favors designs that are less sensitive to position and toe in(ie wider dispersion designs).

However, these are just my personal beliefs, and I definitely don't expect anyone to agree with my beliefs just because I shared them publicly. If I want to convince others, the burden of proof is on me.
 

richard12511

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I wish I could di as good a job as Duke but alas...
I hadn't read your later replies when I wrote that post. I think your later replies did a much better job than your initial replies.
 

tuga

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Duke just talks a good game.
I've heard he designs good game too. And ultimately that's what matters.
FWIW, I've attached a WAV file containing a 5ms burst of white noise, just so you can verify that this is indeed a very quick transient (which moreover, unlike musical transients, has absolutely zero decay).

EDIT: I've also just added a file containing 200ms of white noise, as I thought that might also be illustrative.
Are they safe to reproduce or will they blow my tweeters?
 

Shazb0t

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Duke just talks a good game.
I think @Duke has done a good job thus far of at least trying to explain the reasons for his beliefs(I enjoyed that video).
I wish I could di as good a job as Duke but alas...
Duke just talks a good game.
So, at the risk of being labelled the resident ASR jerk, I have some thoughts regarding the experiment Duke performed. I agree with @andreasmaaan that from the description of the crossover modifications we were able to glean from Duke used in his experiment that it appears likely that he modified the actual on and off-axis response of his "control" loudspeakers. It's not really a surprise to me that they sounded different from this type of modification. It likely wasn't due to time delay at all. Better, maybe? Who knows? But not a great test for a single variable. Definitely not in the ballpark of evidence that would personally move me closer to reversing the findings of an AES published study. Is that talking a good game? Maybe??

Secondly, it seems obvious that phase was not your only variable. Switching from a 4th-order XO to something like the XO you describe is clearly going to produce side-effects in the amplitude response, both on-axis and off-axis, as well as changes in the profile of the nonlinear distortion (I know you're aware of all that ofc).
 
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Duke

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So, at the risk of being labelled the resident ASR jerk, I have some thoughts regarding the experiment Duke performed. I agree with @andreasmaaan that from the description of the crossover modifications we were able to glean from Duke used in his experiment that it appears likely that he modified the actual on and off-axis response of his "control" loudspeakers. It's not really a surprise to me that they sounded different from this type of modification. It likely wasn't due to time delay at all. Better, maybe? Who knows? But not a great test for a single variable. Definitely not in the ballpark of evidence that would personally move me closer to reversing the findings of an the AES published study. Is that talking a good game? Maybe??
Shazb0t, I am not trying to convince you of anything. Can I not state my experience without you taking it as a challenge to your belief system?

Let's try an experiment. I'm going to ask you to put yourself in my shoes for a minute or two, and then I'm going to ask you an open-ended question.

Suppose you are a loudspeaker designer who does not have the resources of a large company like Harman at his disposal. For years, perhaps decades, you have been convinced that loudspeaker phase response is of no audible consequence. And you have proceeded accordingly.

Then one day as you are researching room interactions and psychoacoustics, you come across a youtube video of a researcher you have learned to respect, and he suggests a completely new (to you) idea of WHY phase coherence matters: Because when the overtones arrive at the same time as the fundamental, the result is a brief spike, which effectively increases the signal-to-noise ratio.

Now you are in a situation where you have "dueling experts": Most say that phase doesn't matter, and one says that it does (actually he's saying that time coherence matters, but he uses the word "phase"). And the argument made by the one who says it does matter seems plausible.

If you were in this situation, and your goal was to make worthwhile improvements to your product but not waste time and resources on things which do not matter...

What would you do?
 
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Shazb0t

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Shazb0t, I am not trying to convince you of anything. Can I not state my experience without you taking it as a challenge to your belief system?

Let's try an experiment. I'm going to ask you to put yourself in my shoes for a minute or two, and then I'm going to ask you an open-ended question.

Suppose you are a loudspeaker designer who does not have the resources of a large company like Harman at his disposal. For years, perhaps decades, you have been convinced that loudspeaker phase response is of no audible consequence. And you have proceeded accordingly.

Then one day as you are researching room interactions and psychoacoustics, you come across a youtube video of a researcher you have learned to respect, and he suggests a completely new (to you) idea of WHY phase coherence matters: Because when the overtones arrive at the same time as the fundamental, the result is a brief spike, which effectively increases the signal-to-noise ratio.

Now you are in a situation where you have "dueling experts": Most say that phase doesn't matter, and one says that it does (actually he's saying that time coherence matters, but he uses the word "phase"). And the argument made by the one who says it does matter seems plausible.

If you were in this situation, and your goal was to make worthwhile improvements to your product but not waste time and resources on things which do not matter...

What would you do?
I would design an experiment to test the single variable of phase response. Do you see how it's likely that is not what you did?
 

Duke

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I would design an experiment to test the single variable of phase response. Do you see how it's likely that is not what you did?
Excellent! You would experiment. That's what I did.

However you would apparently conduct a far more complicated experiment than mine. Exactly how are you going to alter the phase relationship between the two drivers without simultaneously altering the frequency response? That falls into the category of "easier said than done".

And even if you COULD entirely separate the phase response from the frequency response using powerful digital EQ, would that tell you which direction to go with your passive crossover design? Of course not! We could not BEGIN to duplicate that result with a passive circuit.

So, what would be the best way to test whether a passive crossover designed with Griesinger's phase paradigm in mind is a net improvement?
 
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andreasmaaan

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As you know my knowledge and vocabulary are limited and I am not very good with technical terms but bear with me for a second.
You're vocabulary around this is fine :)
Imagine a series of quick successive notes played on a kick drum.
The step reponse of a sealed cabinet shows a quick return to 0V whilst that of the ported cabinet continues to oscilate (ring) for a bit.
Won't this post-ringing affect the speaker driver's ability to accurately reproduce the ensuing note and the one after?
Yes, this will affect the (frequency-dependent) "speed" of everything that the speaker plays. You can think of it as though the entire signal from start to finish gets "convolved" with the speaker's "transfer function". It will most definitely affect the speaker's accuracy, as you say.

The question, though, is whether the effects will be audible. I guess I can do two things to try to convince you. Firstly, let me refer you back to the study I talked about in post #181. Look at the group delay in the low bass for those stimuli (Group A) for which no audible difference was discerned. They are literally (way) off the chart.

Secondly, let me suggest you try this piece of software. It's a little piece of freeware that you can load any WAV file into and then play around with phase distortion, and then ABX test yourself with. If you want to emulate the phase distortion caused by a ported speaker, click "low filter" and "phase 24dB/octave", then choose your enclosure resonance Q factor (values under 0.71 are overdamped, values above 0.71 are underdamped).

1605560466642.png


FWIW, I've analysed the output impulse response and it appears to be doing what it claims to do. NB however, at least with my soundcard, there's a tell-tale lag when switching between stimuli, which totally invalidates the ABX test aspect of the software.

@Duke, would you be interested in having a go with this, too? It's also possible to model the phase distortion caused by e.g. a 24dB/octave mid-tweeter crossover... ;)
Transients (attack and decay) are an essential part of the timbre characteristics which make the sound of instruments disctinctive.
So you insist :) Yet decades of research has attempted to establish that humans can hear the levels of phase distortion that you say are a problem, and again and again humans have failed to...
Frequency response does not provide any information regarding the speaker's ability to reproduce transients.
Well, it provides some but not all the information. For closed and ported systems, the bass and treble roll-offs are largely minimum-phase. The frequency response won't tell you directly about energy storage or excess group delay caused by crossovers (although a fair bit can be inferred).
Only anecdotal experience with the old Monitor Audio Studio series from the early '90s.
I guess you will have anticipated what I might say about uncontrolled variables and sighted bias etc when it comes to that...? :)
Have you never listened to the effects of room resonances in the bass? Do you think it's only the peak that is audible or also the tail?
Well, I must admit I haven't put as much work into this question as into the other questions we've been discussing. Conditions are very different in different rooms and the decay times are very long, even compared to those of ported, low-tuned, low-frequency systems.

My inclination would be to say that, with most music in most rooms, these decays probably won't be audible per se - except of course in the frequency domain, in which respect they will tend to be extremely audible as peaks and dips in the response. Though I'm sure with the right signals they could be quite easily made to be audible (i.e. with very sharp transients with no decay of their own).

However, I wouldn't pretend to be able to give a knowledgeable answer to this question ;) I think @Duke would have looked at this in a lot more detail than I have...
Are they safe to reproduce or will they blow my tweeters?
Yes sorry, should have warned you. They are quite loud, please keep the volume low :)
 
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