2 hours ago, rand129678 said:

 

Great post sharing the journey !
 

Can you remember the model Sony Discman?
 

 

Not at all. Sony model names (numbers!) are very difficult to remember! 

I came across another method of generating a target curve. You can watch this video, but the guy isn't the clearest communicator so I will post my own argument / understanding of what he is saying below. 

 

 

Background knowledge (These points are not mentioned in the video, I think he assumes that the viewer already knows these)

 

- The Harman target curve and all target curves represent the predicted in-room response of a speaker that measures flat under anechoic conditions. A speaker that measures flat under anechoic conditions will have a falling frequency response at the main listening position (henceforth abbreviated to MLP). For more background, watch this video. 

- A different approach is required for correction below the Schroeder frequency (Fs) and above Schroeder. To learn more, watch this video. 

- A Moving Microphone Measurement (MMM) is a good way to obtain the average frequency spectrum over an area, in this case he uses an MMM to take a nearfield measurement of the speaker. Read more about MMM here and watch how it is done here. 

 

Argument for a new target curve method 

Floyd Toole has said that the Harman curve should be the result and not the target. This might seem like a semantic distinction, but what it means is that a speaker that measures flat anechoically will product a Harman-like curve as a result. Taking a speaker that does not measure flat anechoically and imposing an artificial target from an in-room measurement at the MLP is wrong. This is because the off axis radiation / directivity and early reflections play a major role in perception. Ignoring this and attempting to correct our speaker to a target will introduce coloration to the sound. 

 

Ideally, you buy speakers that measure flat under anechoic conditions with an ideal radiation pattern. If you already own the speakers, you have to ignore the radiation pattern (since that is already baked into the speaker design and there is nothing you can do about it), and accept that any correction you make will be compromised by your lack of control over the speaker's directivity. You will have to do the best that you can.

 

Then you hire an anechoic chamber or a Klippel, or take a measurement outside to generate a speaker response which you can then equalize so that it is flat. If you have no choice but to measure in situ, a nearfield MMM, taking care to move your microphone far enough so that it encompasses the off-axis response, is the best compromise. 

 

You then take this measurement, generate an inversion, and now you have a speaker with a flat nearfield response. 

 

What this means is that you will no longer have a measurement at the MLP that shows the classic ruler flat falling response (like the ones I have been showing off in this thread). In fact it might look all over the place. He makes the point (in other videos) that this is normal and we should not be aiming for a classic flat falling FR in the MLP in the first place. He gives this example: take two different speakers and equalize both of them so that they show exactly the same FR at the MLP. They don't sound the same. The reason why is because a free field single point measurement does not account for the directivity of both speakers. You can get a lot closer if you correct the nearfield response of both speakers so that they are the same. 

 

In fact he expands on this point in another video. He pointed out that if you hear my voice and your voice in a room, they sound different. If we both go to another room, the room modifies the sound of our voices, but what you sound like is still different to what my voice sounds like. The modification of the sound of my voice from one room to another is the "room transfer function", and it will vary from room to room, and from different points in the room. But if we want my voice to sound like yours, we have to measure your voice and my voice, and equalize it exactly so that it sounds the same. Once we do that, we can apply the "room transfer function" if we wanted to. 

 

However, if you do this nearfield speaker correction, you will not have bass correction at MLP. That video I linked above (about Schroder) points out that the room behaves in two ways - resonator (below Schroeder) or reflector (above Schroeder). So now you have to take another measurement at MLP and perform bass correction for anything < Fs or at the transition frequency, which is < 4Fs. 

 

So now you have two corrections. You will have to split and join them into a single correction, and load that into your DSP software. You can do this, but it is not elegant and it's time consuming. 

 

Explanation of the new target curve method

The method involves: 

 

1. Take a nearfield MMM of your speakers. This will be your poor man's substitute for the anechoic response, and we will use this as our "pure speaker response". 

2. Take a MMM at the MLP. The measurement at the MLP corresponds to "speaker + room". 

3. Subtract the nearfield MMM from the MLP MMM. This theoretically removes the influence of the speaker, and only leaves room response. He calls this the "room transfer function". 

4. Use the room transfer function as your target curve. This will at one swoop correct for frequencies above and below Schroder, i.e. correct your speakers to be flat under anechoic conditions and correct for room induced bass peaks and dips. 

 

My personal opinion 

What I have typed so far is a description of what he proposes. I by no means agree with him. There are several points which I find contentious, for example: 

 

- is a nearfield MMM an acceptable substitute for an anechoic response? 

- he is proposing we can partially correct directivity errors with a nearfield MMM. I am not sure if we can. But it is better than not correcting directivity errors at all. 

- MMM's do not give you phase and timing information. You will still have to take a single point measurement to work this out if you wish to correct it. 

 

I have used this thread to gather my thoughts and my understanding of the theory. Now I have to take some measurements and put it in practice. 

Edited December 28, 20232 yr by Keith_W

I had some time to experiment this evening, so I spent a few hours redoing the target curve. The purpose of this exercise was to chase better accuracy above the Schroder frequency. 

 

Method (for Acourate). 

1. Copy all previously created and linearized crossovers into a new directory. Create a multiway filter. 

2. Do 2-3 MMM's of the same speaker with the multiway filter loaded to verify that your technique is correct and the result is repeatable. If all is OK we move on to actual measurement. 

3. Perform a nearfield MMM of the left and right speakers. I saved these as NF-MMLeft/Right. 

4. Perform a MLP MMM of the left and right speakers. I saved these as MLP-MMLeft/Right. 

5. Go to FD-Functions - Magnitude Difference and calculate the difference between MLP-MMLeft/Right and NF-MMLeft/Right. The purpose of doing this is to obtain the "room transfer function". We will set this as our target curve. I saved these as Target-Left/Right

6. Perform a single point sweep at the MLP. Go through Macro 1 ("Amplitude Preparation") as normal. 

7. Skip Macro 2 ("Target Curve Design"). You have already designed the target curve. 

8. Run Macro 3 with these settings. Remember to UNCHECK "use default target" and enter your own target into the dialog box. Make it do a MONO correction. Repeat this step for the right channel (i.e. Macro 3 is run twice).

 

 

9. Run Macro 4 ("Filter Generation") and the other Macros as per normal. From here we can proceed to verification measurements. 

10. Create a Multiway filter for verification, then run a nearfield MMM: 

 

 

I was pleased when I saw this result. The correction is working as it should, it is supposed to flatten the speaker's nearfield response, and it does that pretty well. Except for >15kHz. I am not sure what happened there. I did see the massive rise in bass, but this was below the Schroder frequency so I ignored it. It should correct itself in the MLP. 

 

 

... except that it doesn't! And there is no Harman-like falling frequency response above 1kHz that you would hope for! In fact there is a dip in the midrange freqs of maybe 2-4dB and it rises again so that the result curve is almost flat above Schroder. 

 

Listening. Well it's obvious I made a mistake somewhere. The new filters sound thick and muddy, with no high freqs and no "air" at all. Sounding thick and muddy is to be expected, given the massive amount of bass boost. But the lack of HF extension came as a surprise, I was expecting it to sound shrill and sibilant. Not the case, in fact it sounds the opposite. 

A small comment if I may?

Audio Definition - Yay!

14 hours ago, Keith_W said:

Listening. Well it's obvious I made a mistake somewhere. The new filters sound thick and muddy, with no high freqs and no "air" at all. 

 

C'mon, Keith - you're too far down the DSP rabbit hole to realise where you are!  

 

As any "audiophile" knows ... the best sound comes from all analogue - with minimal "processing".  With tube amplification.

 

So ideally ... full range spkrs having no XOs; if you must have a multi-way spkr - then it needs to have passive XOs.  😎  And the source needs to be vinyl - not streaming digital files!

 

I doubt anybody understood my post above. But anyway, I decided to do some investigation of how the loudspeaker behaves in my listening room. The purpose of this might become evident later. 

 

Because of the size of the speaker and amount of electronics connected to it, these measurements were made in situ. 

 

 

First, a series of Moving Microphone Measurement (MMM) readings were taken of the loudspeaker at various distances from the speaker as shown above. The microphone was moved to cover on-axis as well as +/- 30 degrees off axis to the speaker. This should allow some of the directivity of the speaker to be captured. Result as below: 

 

 

The reason the frequency response looks so horrible is because these are unadjusted "raw" crossovers of all the drivers with no driver corrections, and no overall room correction. 

 

The volume of the speaker was deliberately not adjusted as the mic went further from the speaker. What you would expect to see: above the transition frequency (in this case about 450Hz), different frequency bands of the speaker will change in proportion to each other, until a critical distance where the shape of the curve remains exactly the same, the only difference being volume. 

 

What you see in this speaker: anything below the 500Hz crossover line is the woofer, which is in a sealed box. Everything above 500Hz is horn - midrange horn (500Hz - 5kHz); and tweeter horn (> 5kHz). You can see that as you get further away, the volume of the woofer drops as expected. However, above 1kHz, the rate of change is relatively small. This implies that the directivity of the horns is very different to the woofers, which would make sense given that they are ... horns. 

 

So let's look at the directivity of the speaker. Ideally the speakers should be taken outside, or better still an anechoic chamber or a Klippel should be used. But nobody has a Klippel, the logistics of hiring an anechoic chamber are horrendous and expensive, and I am too lazy to lug this speaker outside. So once again, I did in situ measurements. 

 

 

The measurements were taken by moving the microphone in a straight line up and down the axis pictured. 

 

 

Result is as above. What we see is that the lowest frequencies (bass) do not change in volume at all. Bass is omnidirectional, and should be relatively immune to microphone movement over a small area. The woofers (80Hz - 500Hz) demonstrate a small drop-off in volume, indicating the directivity is relatively constant. However, > 5kHz, we see a huge drop-off in volume, indicating that the midrange horn and tweeter horn are highly directional. There is a 4dB drop just by going 23 deg off axis, and by 45 deg we are 12dB down. 

 

This method doesn't allow me to get more precise than this. It is difficult to move the mic up and down in a straight arc. In the absence of a robot or some kind of jig to keep the mic aligned while it is moving, this is the best I can do. 

The advantage of the method I described in the previous post is that it can be done with the speakers in situ, with some furniture pushed out of the way. It requires minimal equipment, all you need is your existing measurement equipment and appropriate software. It is also pretty quick and easy to do, it took me just less than an hour to perform all the measurements you see here. Maybe a little longer to come up with the pretty diagrams 

The disadvantages ... well, I can see a few:

- room reflections means that the LF is meaningless below Schroder, and maybe below the transition frequency.
- precision and repeatability is poor, but you may be able to improve on this by constructing a jig (e.g. a vertical slide where you can slide your mic up and down) for the first method
- resolution is poor, and it can only identify large differences in directivity.

For those of us without Klippels, and too lazy to move speakers outside, I believe that these methods give you valuable information. It is not the best or most accurate information, but it is the best we can do within our means. It should provide sufficient information in the higher frequencies for us to make some types of decisions.

 

So let's process the information we have. Using the above measurements, we can generate curves for listening window, directivity index, etc. 

 

 

1. My measurements were done in Acourate, so the first step is to export the measurements into REW (in Acourate, save Mono .WAV; in REW import Audio File, choose .WAV)
2. For all of these measurements, click on "All SPL" and in the control knob in the top right, use RMS Average
3. For Listening Window: Average of 0 deg and 23 deg
4. For Sound Power: Average of all of-axis measurements (I also did 180deg and 360 MMM's - these were taken by moving the mic all around the speaker)
5. For On Axis Directivity index, Trace Arithmetic 0-deg Axis / Sound Power
6. For Listening Window Directivity index, Trace Arithmetic Listening Window / Sound Power

 

The directivity plot is simply a mirrored directivity index. Using an image editor like Affinity, we trace out the directivity index and mirror it vertically. We get this result: 

 

 

And if we use that program to convert to a polar plot, we get this result: 

 

 

This is where it gets interesting. If you look at the polar plot, you will see horizontal dispersion increasing from 500Hz to 13kHz to almost 90 degrees, and then narrows down to about 30 degrees. Below 500Hz the speaker becomes progressively omnidirectional. 

 

The midrange horns are bandpassed at 500Hz - 5kHz, and the tweeter horns are highpassed from 5kHz up. 

 

Most horns support a lower wavelength of twice the diameter of the mouth of the horn. My midrange horns have a diameter of 45cm, meaning the longest wavelength it will support is 90cm. This corresponds to 381Hz. The effective range of the horn is 3 octaves, meaning it maxes out at 3kHz. For the tweeter, the diameter is 12cm. Meaning the lowest wavelength is 24cm, or 1400Hz. The upper range should be 11kHz. Any wavelength longer than the lower limit will not be horn coupled and will lose efficiency. Any wavelength shorter than the upper limit will beam. 

 

This is what we see in the polar plot. The horizontal dispersion progressively widens to 90deg, maybe because the wavelengths are "creeping" along the flare of the horn, then suddenly drops to 30 deg where it is no longer horn coupled and starts to beam. 

 

In other words, predictions from geometry of the horn + some basic mathematics mirrors what I have measured. To me at least, this is really cool and really exciting. 

 

One application of this data, particularly the polar plot, is to see what problem frequencies are and what can be done to make the speaker more "constant directivity". In this case, between 5kHz - 13kHz we see a massive widening of dispersion, up to 90 degrees. I might be able to alleviate that with some thin foam. The foam only needs to be 1/4 wavelength of the freqs that I wish to attenuate, meaning the maximum thickness should be 2cm (5000Hz = 6.9cm wavelength; 1/4 of that is 1.75cm). 

 

We have to be careful not to infer too much from this data since it was collected with poor accuracy and poor resolution though! 

 

But ... Dr. Toole's premise is that a good speaker should have flat on-axis AND constant directivity. This speaker can be equalized so that it is flat on axis, but nothing can be done about the directivity. So now I have to ask myself ... is it time for new speakers? I have evolved this system as far as it will go. I am pleased with the results. There may be more water to be squeezed out of this stone, but I am not too sure. 

Edited January 4, 20242 yr by Keith_W

1 hour ago, Keith_W said:

So now I have to ask myself ... is it time for new speakers? I have evolved this system as far as it will go. I am pleased with the results. There may be more water to be squeezed out of this stone, but I am not too sure. 

I love following this thread, even though most of the tech talk is way above my pay grade.

"Is it time for new speakers? I have evolved this system as far as it will go. I am pleased with the results......"

Isn't that job done?

27 minutes ago, parrasaw said:

I love following this thread, even though most of the tech talk is way above my pay grade.

"Is it time for new speakers? I have evolved this system as far as it will go. I am pleased with the results......"

Isn't that job done?

 

When I bought the speakers, I knew a lot less than I know now. A recurring theme in this thread is my bass problems. Back then, I thought I could solve it with a new amplifier. Then bi-amping. Then bi-amping with more powerful amplifiers. Then subwoofers. Then active (but with passives in place). Then passives bypassed, and went to analog active. Then DEQX. And finally, this iteration. 

Every step along the way, I have not encountered a physical limitation, I was only limited by my budget and learning curve. Now this is a physical limitation I can do nothing about. One solution is to change the bass driver to a horn, so that the directivity mismatch isn't so bad and it would be more "constant directivity". Or I could reconfigure the crossover between the horn/woofer to a gentler slope to avoid the sudden change in directivity. Or stick foam around the woofer box. Or if I can't be bothered ... then it's either stay with it or it's time for new speakers. 

 

I mentioned to a friend that every time I hear large horns, I feel like I am "inside the speaker". He replied that it is because at "normal" listening distances, the speaker hasn't "come together yet". Now that I have done all those measurements at different distances, I now know why I get that "inside the speaker" impression. 

 

People who casually read this thread might think it is nothing more than a tedious bunch of measurements which mean nothing, but what I hope is that people realize how powerful this tool is when it comes to understanding your own system. The cost of entry is very low - all you need is a mic worth a couple of hundred $$$ and free software. What you see will really open your eyes to what your system is doing, and if you know how to interpret what you are looking at, it will point you to what you need to do. 

I am always interested in Keith's new directions and happen to share his belief in the power of DSP - though my own tinkering is at a very basic level.

 

Anyway, I am pleased to report that I finally heard Keith's plasma tweeters!  They are really lovely and, although I won't be upgrading (@$20k per pair) I am very happy to have heard them👍

On 04/01/2024 at 8:33 PM, Keith_W said:

 

When I bought the speakers, I knew a lot less than I know now. A recurring theme in this thread is my bass problems. Back then, I thought I could solve it with a new amplifier. Then bi-amping. Then bi-amping with more powerful amplifiers. Then subwoofers. Then active (but with passives in place). Then passives bypassed, and went to analog active. Then DEQX. And finally, this iteration. 

Every step along the way, I have not encountered a physical limitation, I was only limited by my budget and learning curve. Now this is a physical limitation I can do nothing about. One solution is to change the bass driver to a horn, so that the directivity mismatch isn't so bad and it would be more "constant directivity". Or I could reconfigure the crossover between the horn/woofer to a gentler slope to avoid the sudden change in directivity. Or stick foam around the woofer box. Or if I can't be bothered ... then it's either stay with it or it's time for new speakers. 

 

I mentioned to a friend that every time I hear large horns, I feel like I am "inside the speaker". He replied that it is because at "normal" listening distances, the speaker hasn't "come together yet". Now that I have done all those measurements at different distances, I now know why I get that "inside the speaker" impression. 

 

People who casually read this thread might think it is nothing more than a tedious bunch of measurements which mean nothing, but what I hope is that people realize how powerful this tool is when it comes to understanding your own system. The cost of entry is very low - all you need is a mic worth a couple of hundred $$$ and free software. What you see will really open your eyes to what your system is doing, and if you know how to interpret what you are looking at, it will point you to what you need to do. 

 

Keith, I wonder if you have tried any of your suggestions (such as tinkering with the horn/woofer crossover) since your post?

 

On 04/01/2024 at 8:33 PM, Keith_W said:

I mentioned to a friend that every time I hear large horns, I feel like I am "inside the speaker". He replied that it is because at "normal" listening distances, the speaker hasn't "come together yet". Now that I have done all those measurements at different distances, I now know why I get that "inside the speaker" impression. 

 

Keith - if you haven't already heard them can I suggest you have a listen to @WorkinClassAudio's "ultimate" horn system (in a lovely big space!).    Avantgarde 'Trios' + their 'Bass Horns' + subs.

 

Simply amazing!  

 

 

2 hours ago, tripitaka said:

 

Keith, I wonder if you have tried any of your suggestions (such as tinkering with the horn/woofer crossover) since your post?

 

 

I have. Not quite CEA2034 standard but enough to see a difference: 

 

 

Directivity plot: 

 

 

And the new polar plot: 

 

 

1 hour ago, andyr said:

Keith - if you haven't already heard them can I suggest you have a listen to @WorkinClassAudio's "ultimate" horn system (in a lovely big space!).    Avantgarde 'Trios' + their 'Bass Horns' + subs.

 

I haven't heard his particular horns but I have heard a couple of Avantgarde Trio systems. Yes, they are pretty impressive. With horns, you sacrifice "point source" to get the dynamics, but with big horns you sacrifice even more. I have smaller horns, so the "point source" aspect isn't quite as bad, but then I have to put up with other problems, such as the different directivity of the bass drivers (built into cabinets) and the horns. 

 

I am starting to think that as far as horns go, the PSE-144 is the ultimate design. You get the dynamics, and you also get the point source. It has been a long time since I heard a PSE-144, so I need to find someone who has a pair and go have another listen. 

3 hours ago, andyr said:

 

Keith - if you haven't already heard them can I suggest you have a listen to @WorkinClassAudio's "ultimate" horn system (in a lovely big space!).    Avantgarde 'Trios' + their 'Bass Horns' + subs.

 

Simply amazing!  

 

 

Thanks but not quite amazing at the moment. The preamp is in repair and the volume control on the DAC is a level below the preamp.  The sound is currently a level above what you heard. 

Edited January 28, 20242 yr by WorkinClassAudio

Some of you might know OCA from Youtube. I have been chatting to him and he suggested that I rotate the system in my room 90 degrees against the long wall to improve symmetry and to increase the reflection time to about 5ms. I gave him the measurements of my room (6m x 7m) and he came up with a diagram: 

 

 

I was completely opposed to the idea given the amount of work that would be required. There are 4000 CD's there, a lot of furniture, a lot of heavy equipment, and not to mention the DSP has been dialled in to perfection (something that took me months to do). But then I realized the benefits, besides what he mentioned: because the speakers are further away from the side walls and toed in, the BACCH crosstalk cancellation would work better. The subs can be truly moved into the corners, where there will be further bass reinforcement. I have also been reading Griesinger's work on stereo bass and "bassiousness" (the idea that widely separated subwoofers can widen soundstage). And besides, my room is wider than most rooms are long. 

 

So I enlisted the help of a friend and we started moving stuff around. 

 

Day 1: 

 

 

Day 2: 

 

Day 2 was spent moving extra stuff out of the way and sorting out the cables. I have: 

- 14x Power cables (2 subs, 2 valve monos, 2 transistor monos, 2 tweeters, Vitus, PC, 2 DACs, preamp, monitor, lights) 

- 8x 3.5 TRS to XLR cables

- 9x XLR cables (2 to subs, 2 to pre, 2 from pre to valve monos, 2 to Vitus, 1x long microphone cable) 

- 4x RCA cables

- 4x Speakon cables from sub amp to subs

- 6x speaker cables for the other drivers

- Network cables, USB cables, HDMI cables

 

Day 3:

 

 

Not quite finished yet. I have to move the electronics into the entertainment cabinet, which will require me to modify it by cutting bigger holes into the back of the cabinet. Then I have to redo the DSP. So far I have done a channel check and listened to some music to confirm that it works. It sounds absolutely horrible, given that the DSP is still for the old listening position. 

 

That SGR equipment rack will come up on sale at some point. 

The other thing you may want to look at is the direct vs reflected energy. Linkwitz suggests 6db difference or more. Achieving a time difference between direct and reflected is important and roughly a 1m distance to wall will give you this but then the reflectivity and size of room and type of speaker will dictate how much the difference in direct vs reflected will be. more info here https://www.linkwitzlab.com/listening_room.htm

you can see that for a monopole at 450ms for the example room above max listening distance is about 1.6m and for a dipole its about 2.8m. 

 

There is also a calculator so you can find out exact numbers for your room. You can get it here

 

If I input your room dimension at 6m x 7m x 3m then I get this

 

and this

 

 

which says that max listening distance for a monopole is 1.8m in your room at rt60 of 450ms. If Linkwitz is right I guess the big question is do your speakers act like a monopole or something else. If they do act like a monopole than you will need to drastically dampen the room. Or get some dipoles😁.

Oh, forgot option 3, just sit closer

But…. Where did the TV go?

 

and that Vitus looks rather lonely!

Edited February 7, 20242 yr by aris

I just arrived home from my bus service duties ferrying intoxicated SNA members to their homes 

 

Thanks @frednork that seems to tie in with the concept of "critical distance" which I have been reading about recently. I always thought that it referred to concert halls and it was irrelevant for listening rooms, but OCA suggested that I should consider it. (Critical distance = where the direct sound is the same SPL as reflected sound). As you suggest, one solution is to sit closer to the speaker. I will have a think about that. 

14 hours ago, Keith_W said:

the concept of "critical distance" which I have been reading about recently. I always thought that it referred to concert halls and it was irrelevant for listening rooms

 

That also seems to be Floyd Toole's contention (reading the book at the moment following our discussion on the weekend).

 

Quote

The sound field is not diffuse, and there is no critical distance, as classically defined. If we were to speculate at this early
stage about loudspeaker performance in these rooms, it would seem that a combination of direct and early-reflected sounds would figure prominently in their potential sound quality and that sound power would not be the dominant factor (section 4.3.2)

 

Sitting "too" close to speakers will present a number of non-acoustic psychological "stressors". Disappearance of visual harmony, claustrophobic-like conditions, etc. That might work for eyes-closed listener types but for others it will work against presumed gains in acoustics/psycho-acoustics.

2 hours ago, Steff said:

 

That also seems to be Floyd Toole's contention (reading the book at the moment following our discussion on the weekend).

 

 

Sitting "too" close to speakers will present a number of non-acoustic psychological "stressors". Disappearance of visual harmony, claustrophobic-like conditions, etc. That might work for eyes-closed listener types but for others it will work against presumed gains in acoustics/psycho-acoustics.

 

Sure, you can sit further away and have real acoustic problems but less potential psychological stressors, or you can sit closer , have less acoustic problems and have  more potential psychological stressors.  Best solution is to try and see what works better for you. You dont need to buy anything to try this so a free possible upgrade. however, If your room is highly reverberant > 1000ms then it will sound like a mess from anywhere but up close.

35 minutes ago, frednork said:

 

Sure, you can sit further away and have real acoustic problems but less potential psychological stressors, or you can sit closer , have less acoustic problems and have  more potential psychological stressors.  Best solution is to try and see what works better for you. You dont need to buy anything to try this so a free possible upgrade. however, If your room is highly reverberant > 1000ms then it will sound like a mess from anywhere but up close.

 

Nah it isn't too reverberant. T60 is about 300-350ms depending on the frequency. Not too bad considering there is hardly any room treatment  

 

I have decided to redo the cabling. Just got back from the shops with more cable. 

No probs Keith,  was replying to the comment from Steff moreso than to you. Last rt60 I saw of your setup was at around 450ms which is also perfectly fine.

4 minutes ago, Keith_W said:

I have decided to redo the cabling. Just got back from the shops with more cable.

Bell wire from Bunnings was it?

9 minutes ago, frednork said:

No probs Keith,  was replying to the comment from Steff moreso than to you. Last rt60 I saw of your setup was at around 450ms which is also perfectly fine.

Bell wire from Bunnings was it?

 

High end network cable from Jaycar  $20 for 10m. 

14 minutes ago, Keith_W said:

$20 for 10m.

 

You are just a reckless spendthrift. 😂