Pushing the limits of small speakers - The Reference Mini build thread

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Would still love to see some measurements and hear your thoughts on the sound after using it for a while.
It is a very cool concept, and with good execution.

Thanks for your interest. You'll see measurements soon. Still playing with the speaker. Have been really really busy lately and speakers just had to be put on the sidelines :eek:. Lately I did use rePhase and made linear phase filters for all of the high pass, bass boost shelf, and multi-band compressor filters. This noticeably tightened the bass response. More improvements will come, when I get a breather.
 
I don't know if i'm off-topic (off-contest?) or not :eek:

but here i have some prototypes of floorstanding F.A.S.T. speakers that are only 0,42 ft² (or 0.04m²) each of footprint (base included).

My goal was to get maximum quality-performance for a minimum footprint.

(1) 8W1v3 per side in 12 liters sealed
(1) FR151 per side in OB

Will post pictures later.. :cool:
 
Another update. After learning about measurements for some time, I think I'm getting a hang of how to really properly measure speakers. I don't have anything to show yet, because I ran into problems doing accurate polar measurements. I will show measurements when I have a full set of measurements ready.

A few things I noticed. Turns out bipoles are tricky little things. Maybe this is why there are hardly any bipole speakers. I have to use these vertically. The midrange response is rippled and has a dip around 200Hz in horizontal orientation because the baffle is much wider, and the woofer is off center. This causes various interferences due to the path length causing the rear woofer to not integrate with the front woofer properly. The issue is mitigated in vertical orientation since the baffle is much narrower and the woofers are centered. One unintended bonus from the 200Hz dip in horizontal orientation is that when placed vertically, the 200Hz dip appears in the vertical plane, and it is actually pretty close to the typical ceiling bounce cancellation frequency in a normal height room. So the problem of ceiling bounce is unexpectedly reduced.

I cleaned up the phase of the speaker and used FIR filters to do the bass boost, baffle step, high pass, and clean up the phase from the 3-band compressor. The bass is significantly tighter now, which isn't surprising because I had 6 major sources of phase shift (8th order HPF, 26dB bass boost, 2 LR4 band splitting for 3 band compressor) and it really screwed with the time domain of the bass.

I did a comparison between doing the DSP processing at 44.1KHz and 176.4KHz. The comparison isn't ideal because it isn't blind, and it wasn't instantaneous because JRiver has to be restarted to change sampling rate. I *think* the 176.4KHz had a little more clarity, but the difference is very small. I went back to 44.1KHz because the delay for 176.4KHz is really long at nearly 2 seconds. I need to find a way to process the bass at a lower sampling rate so I can have 176.4KHz for mid and treble without the long delay.
 
Little update. I just moved last week for a new job, so I didn't get a chance to really work on the speakers much to deal with moving. But for something fun, I improved the limiter design a bit to get a bit more maximum clean bass output from the speaker before the limiter clamps the output. I've wrote a lot about my speakers so far, but this time here's a video of the Reference Mini's doing ~107dB of bass and 109dB peaks. The listening position is about 8 feet from the speakers.

Link to the video
 
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I did a comparison between doing the DSP processing at 44.1KHz and 176.4KHz. The comparison isn't ideal because it isn't blind, and it wasn't instantaneous because JRiver has to be restarted to change sampling rate. I *think* the 176.4KHz had a little more clarity, but the difference is very small. I went back to 44.1KHz because the delay for 176.4KHz is really long at nearly 2 seconds. I need to find a way to process the bass at a lower sampling rate so I can have 176.4KHz for mid and treble without the long delay.

Jriver doesn't need to be restarted to change the sample rate of convolution filters. You can set it up with a text file so it chooses the file based on the rate of the track.

To use a filter generated at higher sample rates will need to have the audio matched to that rate, either natively or via upsampling otherwise the coefficients won't be correct.

I think you have the delay time backwards it should be lower the higher the sampling rate if you use the same number of taps in the filter.

Delay due to convolution is (Filter taps /2) / Sample Rate

Eg Filter taps 16000 / 2 = 8000

8000 / 44100 = 0.181 sec

8000 / 88200 = 0.091 sec

8000 / 176400 = 0.045 sec
 
Jriver doesn't need to be restarted to change the sample rate of convolution filters. You can set it up with a text file so it chooses the file based on the rate of the track.

To use a filter generated at higher sample rates will need to have the audio matched to that rate, either natively or via upsampling otherwise the coefficients won't be correct.

I think you have the delay time backwards it should be lower the higher the sampling rate if you use the same number of taps in the filter.

Delay due to convolution is (Filter taps /2) / Sample Rate

Eg Filter taps 16000 / 2 = 8000

8000 / 44100 = 0.181 sec

8000 / 88200 = 0.091 sec

8000 / 176400 = 0.045 sec

While that's correct, you also need correspondingly more taps to achieve the same resolution (and low frequency cut off) for the higher sampling rate. For example, if you double the sampling rate, the number of taps need to be doubled as well to have the FIR filter to do the same thing. However, that should not increase delay, just a higher CPU usage, so I'm not sure why there is more delay in JRiver.
 
Yes to keep the filter identical the taps would need to double but at half the delay you come out with no change. My point was that the sample rate being higher in of itself doesn't add latency.

The way you wrote it before seemed like that is what you thought when you said you wanted to find a way to use a different sample rate on the bass.

That makes sense to be able to use less taps and decrease the latency but I'm not aware of any player software that can do decimation.

Maybe it is the way you have Jriver configured, if you have play from memory set and the track is at a different rate to the filter that could cause Jriver to upsample before starting playback. It seems more like a configuration problem rather than the filter itself being responsible if the filter was identical apart from quadrupling the taps to get to 176.4.
 
one of the best sounding speakers today at BAF2017.
Congrats Brain.

I am going to read this thread in detail.

Thank you! I should give an update to the thread. There are parts that are factually incorrect. It is also outdated as there are lots of improvements since the last update.

I was wondering what you did with the amplifiers. Do you use internal amps now or run passive with amplifier connected to pc with jRiver software. Do you use a mini pc or laptop In other words, how is the setup?

The speakers have internal amps. I use my laptop as the source and DSP processing with JRiver.
 
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Thanks. An update with some latest measurements would be great.
Any details on the custom AT driver, isn't that an expensive affair?

I didn't end up using the AT driver. I ended up using the Wavecor woofers. The AT drivers were expensive, but not nearly as expensive as what I plan next :eek:

Really very interesting. Congrats
one question: you run DSP processing with JRiver on Mac, what multichannel sound card is good for the job?

I used the Asus Xonar U7. It's quite nice, and cheap too.
 
I'm long overdue for an update. I've actually lately been too busy simply enjoying these speakers or giving people demos of these! But there were actually a number of big improvements that made night and day differences to the sound quality of these. I will talk about them over a number of posts. This post will be about my discoveries in sound signature preferences.

The Universal Sound Signature Preference

I'll start with a story. I brought this to an audio enthusiasts get together a few weeks ago as a bunch of people want to hear this speaker I've built. I set the speakers up in my friend's room, but I was having strange setup problems that I've never encountered before that took some time to fix. Since there was about a dozen people waiting and eager to hear this, for the sake of time I only did a rough setup that resulted in poor placements (speakers placed right against the side wall and above ear level) and did only a rough room correction to compensate. However, once it was set up, nobody wanted to leave. This is a room where at least half the people owned 5 figure sound systems at home, many had traditional speakers, some had tubes, some had huge horn speakers, and they all sat there, continuously adding songs to the queue, and listened to the speakers for over 4 hours besides a little break here and there to talk and discuss. The fact that they listened for over 4 hours tells me that everyone truly loved the speaker. Otherwise they would have simply left after a few songs and went to talk to other people outside.

Unlike the traditional thought that people have different preferences in how a speaker sounds, where some people like their speakers sounding bright, dark, warm, etc. I believe there is a universal preference (with one exception), and now I have strong anecdotal evidence that supports this. This is going to be difficult to believe, but once you hear this, I think most of you will agree. I believe the universally preferred sound signature is one that is subjectively flat. I've tried this with over 50 people at this point, and it is clear that this is an appealing sound signature regardless what their original preference is. I've had people actually tell me this is very different from what he's used to hearing, and it changed his view on what is "good sounding".

What is a "subjectively flat" sound?

Now this is tricky. This is not flat, like a speaker with a flat frequency response, but subjectively flat. Our ears hear differently at different volumes. At normal volumes, our ears are less sensitive to bass, and to treble, but to a lesser extent. As the volume goes up, our ear becomes increasingly sensitive to bass and treble. This means that a speaker that measures flat will sound thin at normal volumes since the subjective frequency response that our ear hears will be a "semi-circle" shape where the bass and treble is rolled off due to the lower sensitivity of bass and treble. This is why many speakers sound better at louder volumes. This is because as the volume goes up, our ear's sensitivity to bass and treble gradually increases. This means that the speaker is sounding more and more subjectively flat as the volume goes up. We like a flat sound, which is why we like the speaker played louder because it is closer to flat. The same reasoning can be applied to why bright speakers sound nice at normal volumes, but becomes annoying at high volumes. At normal volumes, the bright sound compensates for our lack of sensitivity to treble, so the top end sounds subjectively flatter than a neutral speaker. But when you turn the volume up, the ear's sensitivity to treble increases, and now the ear hears a sound with too much treble, and we don't like it because we like a flat sound.

What does this mean? What we perceive as sounding "flat" varies dramatically depending on the volume. In order to achieve the universal preference of subjectively flat, we need a speaker that changes its frequency response depending on the volume it is played at. This is not possible to achieve this with any passive speaker.

So for a speaker to sound subjectively flat, there must be a bass boost and treble boost. It is not a straight boost either, but a continuous slow rising response starting from low midrange (around 400Hz) and into the very deepest of bass. A similar, but a much smaller rising response is needed for treble starting around 5000Hz. Isn't that just a V shaped frequency response? You must be shaking your head in disgust! V shaped??? Blasphemy!!! However, the response I describe here is almost impossible to do on a passive speaker since the boost required on the bass would require the speaker to lose over 10dB of sensitivity, and the inductor needed to start cutting at 30Hz is just impractical. This is why most "V" shaped speakers don't sound great. They are not getting the right target curve needed to sound correct. Even if they are through the use of an external equalizer, the amount of bass boost and treble boost needs to be different depending on the volume so it always results in a flat subjective response to our ears. The equalizer provides a static change in frequency response and it doesn't changing with volume, so it'll sound bad at higher volumes and create fatigue (too much treble) and boominess (too much bass). With my speakers, after a calibration it knows exactly what the SPL is at the listening position, so it can automatically adjust the bass and treble depending on the listening volume. This is the key to get the speaker to sound subjectively flat to our ears, and if done right, it sounds downright amazing, and just sounds right.
The Exception

(*) What is the exception? I've found that this is not true for people with substantial hearing loss, i.e. a lot of old people. This is the group that heavily favours a very rolled off treble sound. For some reason I don't yet know, these people seem to think treble is the devil. I would think with hearing loss, you would want MORE treble to compensate for their reduced high frequency hearing. However, it seems like people with hearing loss genuinely hate treble because for some reason it greatly irritates them. I brought these to the Burning Amp, where most of the attendees are well over 50. I ran a long 20 second frequency response sweep, which meant there were 10 seconds or so where the sweep is in the treble region. I noticed several people covering their ears during the sweep, and some looked like they're in pain. I got much less positive reception there, which is understandable because most of the speakers that were presented had, in my opinion, essentially no treble. And of course, these "treble-less" speakers got huge positive receptions, which is not surprising at all if hearing high frequency causes these people to contort their facial expressions.
 
This is all pretty interesting, and I won't take issue with your loudness contour analysis as I have not done the research. I do however find that in your analysis of old people hearing and the lack of praise for your speakers at Burning Amp you jump to a lot of conclusions and make a lot of assumptions.
 
An old friend used to run a mobile disco in the 1970's.
One day he had a problem with a speaker box.
He took the front off and I was shocked to see about a dozen small speakers.
I was expecting to see one 12 incher.

The speakers sounded fine, although he was a bit paranoid about not playing them too loud. I guess he was afraid of breaking them with too much power.
 
Equalizer APO has a loudness correction filter similar to what you mentioned. You set reference level with system volume and then it adjusts the curve based on that. What it lacks is the ability to design the curve yourself. The knees in the curve are actually pretty close to the numbers you used, 400hz/6000hz on program but I doubt the slopes are the same.
 
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