After listening for some time with this frequency response: 

 

 

I started to realize that although the system sounds neutral, linear, and has amazing clarity, it does not have the same punch in the low end that I have experienced in some other systems. I have tried different target curves, e.g. a bass shelf from 20dB - 80dB - this gives some low end impact, but it also makes the bass sound bloated. 

 

That is when someone in another forum suggested that I google the "Pultec Punch". Watch this video for an explanation: 

 

 

Or you can read this article. 

 

This is what it does: 

 

 

It boosts bass up to a certain frequency, followed by an undershoot. The boost and attenuate functions as well as the corner frequency can be adjusted. In professional settings, it might be used for drums, etc. where the signal still exists in individual channels before mixing. At home, all I have access to is the finished product where all instruments that produce bass have already been mixed together, so it selective for frequency only, and not for individual instruments - a major weakness. As you can see, Pultec is not a "simple" EQ that simply raises the target band, it is a "dirty" EQ in that there is a Q and there are sideband side effects. It is basically a sound effect. 

 

There are other Pultec EQ's that can be used to dial up / down the mid and top end. I have only played with these briefly, they have the effect of fattening up vocals and making them sound richer, or thinning them down until they have no body. 

 

Nonetheless, it is free to download and try, so I downloaded it. It comes as a VST, and needs to be hosted on a VST capable player or convolver. I used JRiver. A few companies make Pultec EQ VST's. Apparently the best and most faithful reproduction is UAD so that's the one I used. 

 

Outcome: HOLY CRAP this thing is amazing! I can wind it up to maximum settings and the power and snap of the bass is overwhelming. In fact it is far from realistic and it is almost comical. I have dialled it all the way down and it gives me a nice low end. It blends in nicely with the rest of the frequency response. Previously, only the mids and top end sounded "physical"; while the bass was there, it didn't have the same impact as the rest of the FR. Now it does. I have paid for my license and I will keep it. 

 

Would be great to see measurement with the curve in place. Do you actually get to see the adaptation?

Unfortunately there is no display to show you what is happening. As you know, "someone" has my interface at the moment and I can't take measurements. I suppose I could play white noise and do a screen capture using JRiver's real time analyzer. 

You could probably implement this as a series of curves in Acourate. Maybe I should read the article before mouthing off. 

Edited December 13, 20232 yr by aris

5 minutes ago, aris said:

You could probably implement this as a series of curves in Acourate. Maybe I should read the article before mouthing off. 

 

I probably could. But I prefer effects to be defeatable at the push of a button. And easily adjustable in real time, given that I am still tuning it. 

You have been to my place. I am not sure if I showed you the effect of turning off the VST's one by one. As you know, I have a few VST's in the pipeline. 

On 13/12/2023 at 7:22 PM, Keith_W said:

After listening for some time with this frequency response: 

 

 

I started to realize that although the system sounds neutral, linear, and has amazing clarity, it does not have the same punch in the low end that I have experienced in some other systems. I have tried different target curves, e.g. a bass shelf from 20dB - 80dB - this gives some low end impact, but it also makes the bass sound bloated. 

 

That is when someone in another forum suggested that I google the "Pultec Punch". Watch this video for an explanation: 

 

 

Or you can read this article. 

 

This is what it does: 

 

 

It boosts bass up to a certain frequency, followed by an undershoot. The boost and attenuate functions as well as the corner frequency can be adjusted. In professional settings, it might be used for drums, etc. where the signal still exists in individual channels before mixing. At home, all I have access to is the finished product where all instruments that produce bass have already been mixed together, so it selective for frequency only, and not for individual instruments - a major weakness. As you can see, Pultec is not a "simple" EQ that simply raises the target band, it is a "dirty" EQ in that there is a Q and there are sideband side effects. It is basically a sound effect. 

 

There are other Pultec EQ's that can be used to dial up / down the mid and top end. I have only played with these briefly, they have the effect of fattening up vocals and making them sound richer, or thinning them down until they have no body. 

 

Nonetheless, it is free to download and try, so I downloaded it. It comes as a VST, and needs to be hosted on a VST capable player or convolver. I used JRiver. A few companies make Pultec EQ VST's. Apparently the best and most faithful reproduction is UAD so that's the one I used. 

 

Outcome: HOLY CRAP this thing is amazing! I can wind it up to maximum settings and the power and snap of the bass is overwhelming. In fact it is far from realistic and it is almost comical. I have dialled it all the way down and it gives me a nice low end. It blends in nicely with the rest of the frequency response. Previously, only the mids and top end sounded "physical"; while the bass was there, it didn't have the same impact as the rest of the FR. Now it does. I have paid for my license and I will keep it. 

 

Interesting, I might try it. But I can't really see the point when my Dirac Live plugin can be used to create up to eight different curves with configurable curtains.

On 13/12/2023 at 8:52 PM, Keith_W said:

This is what it does

To understand in theory what is best (you can never account for individual subjectivity, obviously) in a system, we need to know the "directivity" of the speaker.

In the ~300 to ~3000 window, the directivity often changes quite dramatically, depending on the cabinet size/shape (sometimes called "baffle step") or the coverage pattern of horns, or whatever.

Different coverage patterns will give rise to different requirements for the flatness/tilt of the axial/direct/windowed response, or the "power" response measured in the listening position.

 

The point being that recommendations (like this one, and others) are created (or verified by people using) typical speakers, which may not match yours.

 

Your (previous) response looks about as loud as I'd want at 1khz and about as flat as I'd want from 100 to 400Hz, which might sound a bit clear and cold .... but as said, it's hard to generalise too much without seeing more data.

4 hours ago, Satanica said:

Interesting, I might try it. But I can't really see the point when my Dirac Live plugin can be used to create up to eight different curves with configurable curtains.

 

I had some misgivings about using it. This is a professional tool that is used to provide snap and kick to individual tracks before mixing, e.g. adding snap to drums whilst leaving other instruments alone. Obviously, with music that has already been mixed, you can not separate out individual tracks for this treatment. You would be applying the correction to a frequency band, which is a rather crude way of doing it. 

 

However, another way of looking at it is as a tool to help find a better target curve. I have dialled it all the way back to boost/cut +2dB/-2dB with a corner frequency of 60Hz. The effect is more subtle, but still enough to provide some punch. Push it all the way up and the bass hits you like a sledgehammer, there is actual physical force in the sound. However this physical force also rattles windows and makes my subwoofers "walk" which is quite impressive given how heavy those subs are. 

 

I still don't know exactly what it is doing in my system with these settings, because at the moment I have no ability to measure. I lent my interface to a friend. You are right, it can be baked into the target curve, but then you don't know how much boost/cut you need without a tool like this to help you get there. 

 

3 hours ago, davewantsmoore said:

To understand in theory what is best (you can never account for individual subjectivity, obviously) in a system, we need to know the "directivity" of the speaker.

In the ~300 to ~3000 window, the directivity often changes quite dramatically, depending on the cabinet size/shape (sometimes called "baffle step") or the coverage pattern of horns, or whatever.

 

The horns are crossed over at 500Hz so unless there are residual effects from the Pultec at those frequencies, it probably won't matter. Having said that I am totally talking out of my arse here, because at the moment I don't know where it cuts off because I haven't measured it. It might still be in the omni portion of the FR in which case the directivity is moot. All speculation at this point! 

 

For some time I have wanted to measure the directivity of this speaker. The ONLY way I can do it is to take the speaker to my backyard, hoist it up at least a meter off the ground, construct a turntable sturdy enough to support a 110kg speaker, and then figure out a way to place a mic that high because my tripod maxes out at maybe 1.8m (same height as me). Then there are the logistics - move the speaker, move the amps, cable the whole thing up, and try to do it on a weekend with reliably good weather (far from guaranteed in Melbourne!!). Then comes hours of measuring, then I can generate a polar plot using software like VituixCAD. 

 

I spoke to the Klippel representative for Australia and apparently NOBODY owns a Klippel capable of measuring whole loudspeakers. The only Klippels in Australia are for measuring drivers. And given that I would encounter the same logistical problems transporting my speakers elsewhere, the Klippel would have to come here, and those things are not compact either. Not to mention, I have no idea how to drive one of those things given I have never seen one, let alone used one, so the expense of renting a Klippel + hiring the guy is really going to add up. 

 

And I would go through all this effort to obtain information that is not actionable. I can't do anything about the design of my loudspeaker, short of designing another loudspeaker myself. 

 

I think it already sounds amazing, so i'll just continue. I will put that lingering curiosity away and ignore it. I will remind myself of the trouble and expense of measuring the directivity every time that question comes up. 

 

I think I saw in another thread that you have horns. How did you measure them? 

 

3 hours ago, davewantsmoore said:

 

 

Hi Keith_W, just wondering if you have thought that in all your manipulations of the musical signal you may be getting farther from rather than closer to the original intent of the artist (or recorder, producer etc.)

 

Cheers, fox

7 minutes ago, foxontherun said:

Hi Keith_W, just wondering if you have thought that in all your manipulations of the musical signal you may be getting farther from rather than closer to the original intent of the artist (or recorder, producer etc.)

 

I suggest you can't conflate 'artist' / 'recorder' (I presume you mean 'recording engineer'?) / producer.

 

I think it's the recording engineer's preferences that rule - bugger what the artist thought s/he wanted!  

 

Hi andyr, you may have missed the point a little (thats why i added etc), because it is a question more along the lines of missing the forest for the trees, but without any of the negativity usually associated with that saying.

 

fox

10 minutes ago, foxontherun said:

Hi andyr, you may have missed the point a little (that's why i added etc), because it is a question more along the lines of missing the forest for the trees, but without any of the negativity usually associated with that saying.

 

Whereas, I would say ... it is you who have missed the point!

 

My view is that we have NFI what the intent of the artist(e) was ... all we have is the recording.

 

So we do what we can, to make the recording sound its best on our playback system.

 

I surrender to your pedantry.

 

fox

1 minute ago, foxontherun said:

I surrender to your pedantry.

 

fox

 

Que???  😝

 

3 hours ago, foxontherun said:

Hi Keith_W, just wondering if you have thought that in all your manipulations of the musical signal you may be getting farther from rather than closer to the original intent of the artist (or recorder, producer etc.)

 

Cheers, fox

 

Great question! 

 

My answer goes something like this: I think most SNA'ers would agree that the most weak link in any system is the speaker and the room. The performance of all electronic components is orders of magnitude superior to the distortion of speakers, and the alterations of frequency response imposed by the speakers and the room. ALL the interventions like speaker correction, room correction, and even uBACCH get closer to the "ideal" speaker in the "ideal" room. 

However, at some point I am moving beyond that. If I think there is not enough bass, I give it some more. If I don't like the target curve, I can change it. If I like more "warmth", I can add it. This system is chameleon-like and can go anywhere from sounding as thick as a pair of old Tannoys to bright like Missions to punchy like Wilsons, or any combination thereof. I have already tuned it to neutral, now I am only playing and having fun. Such is the power of DSP 🙂

 

I think there is room for personal preference in this hobby, and I know that many people do not agree with what I am doing. That's okay, disagreement and differences in philosophy and approach is NORMAL in any hobby. I add salt and pepper to my hi-fi to taste. In fact, I think that everybody adds salt and pepper whether unwittingly or not. 

 

The important feature of this system is that all the interventions can be turned off with a few button pushes and the system is restored back to standard. Maybe i'll post another thread and ask SNA the question

22 hours ago, Keith_W said:

The horns are crossed over at 500Hz so unless there are residual effects from the Pultec at those frequencies, it probably won't matter. Having said that I am totally talking out of my arse here, because at the moment I don't know where it cuts off because I haven't measured it. It might still be in the omni portion of the FR in which case the directivity is moot. All speculation at this point! 

 

If you're using it at 100Hz ..... then between 100Hz and 500Hz, it has quite a significant effect.

If you're horn "cuts off" at 500Hz, then you will have a (potentially relatively sudden) change in directivity/power at 500Hz, and somewhat below, depending on how sharply the horn is cut off.    With a 24dB/octave slope (ie. just he horns natural response) then 100dB at 500Hz, means 75dB at 250Hz ..... so still somewhat relevant.

 

You might be using it lower in frequency, then yes, it will be out of the way of the horn.

 

22 hours ago, Keith_W said:

For some time I have wanted to measure the directivity of this speaker. The ONLY way I can do it is to take the speaker to my backyard, hoist it up at least a meter off the ground

, construct a turntable sturdy enough to support a 110kg speaker, and then figure out a way to place a mic that high because my tripod maxes out at maybe 1.8m (same height as me). Then there are the logistics - move the speaker, move the amps, cable the whole thing up, and try to do it on a weekend with reliably good weather (far from guaranteed in Melbourne!!). Then comes hours of measuring, then I can generate a polar plot using software like VituixCAD. 

It isn't at all easy.  The ideal way is if the speaker manufacturer provides this data.

 

22 hours ago, Keith_W said:

I spoke to the Klippel representative for Australia and apparently NOBODY owns a Klippel capable of measuring whole loudspeakers. The only Klippels in Australia are for measuring drivers.

Yep.

 

But I don't think you need to go anywhere this far.   The discussion I raise about directivity shouldn't be seen so much as "you need to measure it" .... rather than just "be mindful of it".   Be mindful that if your speaker doesn't radiate like a "typical box speaker", then the typical "house curves", or recommendations for EQ, or anything else, might not apply so much.

 

The other thing(s) to be mindful of.... while I've grabbed the microphone.... is to consider that putting EQ "to preference" on top of a speaker where the underlying levels of summation is not great, will be fraught with issues.   I guess said another way, there are two sides to this "optimisation coin" when you're doing what you're doing (your own levels, tilts, crossover summing, etc.) .... there is your ultimate target SPL (referenced to some measurement axis) .... and then there is the driver levels, band-limiters, tilts, delays, etc.   Spend a lot of time getting this side of the coin right (eg a "dead flat", well summing response) ......  and it will pay a lot of dividends on the other side (the EQ to "preference") side of the coin.    Obviously those people who are not wholly//partly designing their own speaker, only have to deal with one side of that coin.

 

22 hours ago, Keith_W said:

And I would go through all this effort to obtain information that is not actionable. I can't do anything about the design of my loudspeaker, short of designing another loudspeaker myself. 

Of course, but will help you in understanding why certain target curves sound the way they do.... and help you predict what is a good thing to do.

You can also alter the crossover point of the bass to horn, or the driver slopes, to optimise the directivity (eg. if bass is narrow and horn is wide, etc.)

 

22 hours ago, Keith_W said:

I think I saw in another thread that you have horns. How did you measure them? 

Yes.  I have the PSE144 horn, which is an elliptical horn of  1m x 0.6m and so operates from about 400Hz up.

 

I hung the horn from my balcony in an extremely janky way, and got pretty rough data.... but the manufacturer provided directivity plots that told me what I roughly needed to know.

The crossover for the coaxial arranged drivers is ~1khz, so I can tinker with that using windowed measurements done inside.

I was also designing multi-entry horns (like "danley synergy horn") before Paul the manufacturer upped his horn game to produce the PSE144, and so I give up on my project, as I simply don't have the manufacturing capability (these sort so horns are difficult to make for ordinary people).

 

I have designed some much smaller horns (10cm to 30cm in diameter) for more ordinary 2 ways and 3 way speakers, that I've had moulded from plastic, or made with CNC wood ..... and for those I measurement outside in a suitably janky setup that is 4m off the ground, but it's windy and inaccurate.

I rely a lot on simulation, and use measurements to confirm alignment with the simulation.   That being said, coaxial horns, or multiway horns, are extra tricky to simulate.

 

It is not easy.

 

As I said above... my recommendation would lean towards "don't bother"....  but don't "ignore" it (be mindful of it).   The "baffle step" in a speaker (which still happens in a horn speaker) is the trickiest part, because it both represents the largest amount of diffraction / distortion in the speaker.... and it is also the most difficult to measure (due to close by surfaces).   To me this is one of the biggest advantages of EQ (and/or active speakers) is that the critical range from 100 to 1000Hz can be optimised and tilted to the right overall SPL for the speaker position in a room and the directivity of the driver(s).

14 minutes ago, davewantsmoore said:

 

If you're using it at 100Hz ..... then between 100Hz and 500Hz, it has quite a significant effect.

If you're horn "cuts off" at 500Hz, then you will have a (potentially relatively sudden) change in directivity/power at 500Hz, and somewhat below, depending on how sharply the horn is cut off.    With a 24dB/octave slope (ie. just he horns natural response) then 100dB at 500Hz, means 75dB at 250Hz ..... so still somewhat relevant.

 

That is a very good point! I did not think of that!! I wonder if I should consider a lower order high pass for the horn. 

 

This is the frequency response of the horn, with the factory crossover in place. 

 

 

And this is a nearfield measurement of the horn, with the passive crossover bypassed, but with no driver correction or digital crossover. You can see that it naturally starts to roll off at 500Hz, and by 400Hz it is -6dB and about to transition to the steep part of the slope: 

 

 

The curve below is with full driver correction and bandpass applied. After I took the above measurement, I realized that I have a LOT of headroom for the horns since they are so efficient. They are 98dB/W/m, compared to the woofers below it which are about 85dB/W/m. If I wanted to, I could cut the volume of the horn and use 400Hz as the corner frequency. But I decided to keep it at 500Hz: 

 

 

I am crossing over the horns much steeper than factory but at the same crossover point. 

 

Hmmmmm. Now I really want to go and measure the effect and think about redoing the XO. Thanks for pointing that out! 

 

14 minutes ago, davewantsmoore said:

It isn't at all easy.  The ideal way is if the speaker manufacturer provides this data.

 

Yeah. I wrote to them a few years back and they never replied. 

 

14 minutes ago, davewantsmoore said:

Yes.  I have the PSE144 horn, which is an elliptical horn of  1m x 0.6m and so operates from about 400Hz up.

 

I know that Paul Spencer has measured the directivity of his horn. He would have to, since the entire purpose of that design is to provide the smoothest directivity possible from a horn! I think his measurement setup was as janky as yours, I have seen pictures of the measurement process. At least Paul has a massive backyard but he lives not far from open water so he would have to be patient to wait for the wind to die down

 

I know that Uli (author of Acourate) also uses horns. He sent me a pic of his system: 

 

 

(And BTW, for those who are concerned that I might be violating his privacy by posting a picture from a private email - he has posted this image on some other public forums) 

 

 

14 minutes ago, davewantsmoore said:

 

 

5 hours ago, Keith_W said:

That is a very good point! I did not think of that!! I wonder if I should consider a lower order high pass for the horn. 

It depends on the blend between the woofer the horn, at different angles (ie. getting them to sum across their "directivities").    It is not something you would/should do, then then see what "sounds good".... it is something you should "get right" (so it sums properly across angles), and then EQ the total/summed response to whatever target "sounds good".

 

5 hours ago, Keith_W said:

And this is a nearfield measurement of the horn

Are you using this data to determine your correction(s)?  If so, then that is mistake, generally.

When you measure a speaker at different distances, it changes how the diffraction sums with the direct sound.

 

5 hours ago, Keith_W said:

If I wanted to, I could cut the volume of the horn and use 400Hz as the corner frequency. But I decided to keep it at 500Hz: 

I'd make the decision by look at the frequency response of the woofer, and of the horn, vs angle.     (and also based on not using a driver where the non-linear distortion was too high, but I doubt that is a problem here).

Basically trying to get it to if you look at the crossover between the horn and woofer, from many different angles, that the sum is as optimised as possible through the listening window (and much wider, if possible).

Yep, not possible to do any of that. Not without 3 strong men and good Melbourne weather, anyway. Sigh.

I found a whole bunch of old photographs of systems I have owned over the years! Unlike many people here, I am slow to change equipment. I tend to buy things and enjoy them for many years before upgrading. So here we go ... 

 

 

1991: my first system. Mission 760i speakers, NAD 3020i amp, Sony Discman CD and Sony Walkman (cassette) as source. About a year later I added a Marantz CD52 CD player. 

 

 

1995: The Missions went out and I bought a pair of Audio Definition Signature speakers. The amp was changed to Pioneer A400. The turntable was a Sota Sapphire with Sumiko FT-4 arm. I used a Sumiko Blue Point Special which my dad destroyed while he was cleaning the turntable (snagged the cloth on the needle and ripped it out). It was replaced with a Denon DL304 cartridge. 

 

 

2004: I bought an entire new system. ProAc D38 speakers, Cary CAD-805AE SET monoblocks, Cary CD-306 SACD player, Cary SLP-05 preamp. The Audio Definitions were relegated to surround speakers. 

 

 

2008: enter my current speakers. When I heard them, I fell in love with the top end, but I knew that the bass had problems. It sounded like two speakers in one, one speaker for the top, and another for the bottom. I thought that it would be an easy fix, after all I was driving them with a pair of SET's. Maybe they needed more power? So I bought a pair of Cary CAD-211AE monoblocks and vertically bi-amped them. It helped a little, but not enough. 

 

 

2012: Over the next few years I kept adding to the system to try to fix the bass. By 2012 I had a pair of JL Audio F110 subs, I sold the Cary SET amps, and bought an SGR amp. By this time, every driver in the system had its own amp channel but I was still not satisfied with the bass. 

 

 

2016: In 2016 I decided to take the plunge and convert the speakers to active. The passive crossovers were bypassed, and I bought an 8 channel Merging NADAC MC-8 DAC. The woofer drivers were also swapped to Lorantz, and now the SGR amp didn't have enough power. So out it went, now replaced with Accuphase power amp which was on kind loan from a friend. The JL Audio subwoofers were replaced with a pair of custom subs which I designed with the help of @Red Spade Audio, each sub has two Rythmik drivers for a total of four sub drivers in total. The system was now controlled by the PC which made the crossovers. 

 

I knew close to nothing about DSP and DRC back then, I was only aware of its potential. It has taken me years of learning to get to this point. And for a couple of years after switching to this system, the sound was close to unlistenable. It really was a downgrade. I eventually made some filters which were "good enough" and kept the system as it was until late last year, when I decided to redo all the filters. This resulted in a massive improvement in quality without having to spend any money. 

 

And that takes us to now 🙂 Nothing more needs to be done. 

Awesome Keith - great to see your journey 

That's a great little journey through your adult life in audio.

In 1992 we were about on par. Your 2004 system already screams endgame! at me. You have really invested so much labour, finesse and funds into your current set up...and really, nothing more needs be done. That, or buy a house with a customised listening space. Until such time, enjoy Schubert's Lieder, Bach's cantatas and Shostakovich's orchestral explosions.

I gave considerable thought to posting the following, because I don't want to go off on a tangent on this thread which I (and many others) follow and enjoy so much. 

However, in answer to @foxontherun's post of 16/12/2023, @andyr (whose knowledge of audio exceeds mine by a huge amount in most areas/aspects) posted some views which I think may be questioned:

 

On 16/12/2023 at 6:35 PM, andyr said:

I suggest you can't conflate 'artist' / 'recorder' (I presume you mean 'recording engineer'?) / producer.

 

I think it's the recording engineer's preferences that rule - bugger what the artist thought s/he wanted!  

 

I have spoken to several people who have considerable experience and working knowledge in the various roles of tracking engineer/recording engineer/mastering engineer/ producer within Australia, working with musicians across a variety of genres.

I have also had the pleasure and privilege of enjoying ongoing contact with a number of musicians whose recordings are widely available and well known, both in Australia and overseas.

 

Each of these people spoke about the importance of the collaboration which occurs between the artist and the people who work for the recording label.

Artists generally have input (to varying degrees) in recordings which are released in their name.

Established artists especially, have considerable input into recordings of their work, from the initial stages right through to the final product which is released to the public - invariably, at least as much as anyone who works for the label.

In classical music, where great artists not only have an extraordinary talent, but will also have put tens of thousands of hours of hard work into developing their craft (often over many decades), nothing is going to be released by their label which is not approved by the artist.

 

On 16/12/2023 at 7:16 PM, andyr said:

My view is that we have NFI what the intent of the artist(e) was ... all we have is the recording.

 

So we do what we can, to make the recording sound its best on our playback system.

 

Great classical musicians quite often contribute a detailed input into the liner notes which accompany their recordings, in which they often make it very clear, what they wish to convey in their rendition of the piece, i.e. their "intent".

Rather than having "NFI what the intent of the artist(e) was", often we are privy to their very clearly stated intent, and as you say "....we have .. the recording" - a recording which they have approved as being a record of their effort to pass on to us their intent , and their skills, to us the appreciative listener.

 

Your conclusion - "So we do what we can, to make the recording sound it's best on our playback system"?

Absolutely!!!!  

 

I admit I was being a bit flippant when I made that post @parrasaw ... but it was based on this book I'd read:

 

 

An hilarious read!  👍

 

 

 

On 24/12/2023 at 1:10 PM, Keith_W said:

I found a whole bunch of old photographs of systems I have owned over the years! Unlike many people here, I am slow to change equipment. I tend to buy things and enjoy them for many years before upgrading. So here we go ... 

 

 

1991: my first system. Mission 760i speakers, NAD 3020i amp, Sony Discman CD and Sony Walkman (cassette) as source. About a year later I added a Marantz CD52 CD player. 

 

 

1995: The Missions went out and I bought a pair of Audio Definition Signature speakers. The amp was changed to Pioneer A400. The turntable was a Sota Sapphire with Sumiko FT-4 arm. I used a Sumiko Blue Point Special which my dad destroyed while he was cleaning the turntable (snagged the cloth on the needle and ripped it out). It was replaced with a Denon DL304 cartridge. 

 

 

2004: I bought an entire new system. ProAc D38 speakers, Cary CAD-805AE SET monoblocks, Cary CD-306 SACD player, Cary SLP-05 preamp. The Audio Definitions were relegated to surround speakers. 

 

 

2008: enter my current speakers. When I heard them, I fell in love with the top end, but I knew that the bass had problems. It sounded like two speakers in one, one speaker for the top, and another for the bottom. I thought that it would be an easy fix, after all I was driving them with a pair of SET's. Maybe they needed more power? So I bought a pair of Cary CAD-211AE monoblocks and vertically bi-amped them. It helped a little, but not enough. 

 

 

2012: Over the next few years I kept adding to the system to try to fix the bass. By 2012 I had a pair of JL Audio F110 subs, I sold the Cary SET amps, and bought an SGR amp. By this time, every driver in the system had its own amp channel but I was still not satisfied with the bass. 

 

 

2016: In 2016 I decided to take the plunge and convert the speakers to active. The passive crossovers were bypassed, and I bought an 8 channel Merging NADAC MC-8 DAC. The woofer drivers were also swapped to Lorantz, and now the SGR amp didn't have enough power. So out it went, now replaced with Accuphase power amp which was on kind loan from a friend. The JL Audio subwoofers were replaced with a pair of custom subs which I designed with the help of @Red Spade Audio, each sub has two Rythmik drivers for a total of four sub drivers in total. The system was now controlled by the PC which made the crossovers. 

 

I knew close to nothing about DSP and DRC back then, I was only aware of its potential. It has taken me years of learning to get to this point. And for a couple of years after switching to this system, the sound was close to unlistenable. It really was a downgrade. I eventually made some filters which were "good enough" and kept the system as it was until late last year, when I decided to redo all the filters. This resulted in a massive improvement in quality without having to spend any money. 

 

And that takes us to now 🙂 Nothing more needs to be done. 

 

Great post sharing the journey !
 

Can you remember the model Sony Discman?