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Multi-Channel, Multi-Amplifier Audio System Using Software Crossover and Multichannel-DAC

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dualazmak

dualazmak

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(Grin) No pics of the wiring behind it all?? :rolleyes:;)

Thank you again for your kind encouragements!

You can find my "wiring behind" photo as of May 30 2022 in my post #540 (before the revival of LP player); please allow me showing it again here.:)
WS003808.JPG
 
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DrCWO

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I am really impressed about your setup. Thanks‘ for sharing :)
Live playback for LPs in a digital system is always a challenge.
As I also have a nice solution for LP playback I like to share it here.

I run a 2x2 active system with a pair of full range horns and two woofers. XO and DRC are done with convolution filters in Roon. Playback via the new RME ADI-2/4 SE. It has two stereo DACs so I can drive Horns and Woffers in parallel from Roon.

But how to handle LPs? As I am the developer of the rooExtend-box I programmed a Roon Extension for it I called rooPlay. With rooPlay I can connect my turntable directly to the ADI-2/4 that also offers great RIAA support and stream it via Roon and its DSPs to the DACs of the ADI-2/4.

For me this was the perfect solution. I only need the ADI-2/4, the rooExtend-Box that I connect directly to the active Woofers and my Krell 2250e driving the horns. Wireless remote control is done with a Nuimo and my rooNuimo Ronn Extension also hosted by the rooExtend-Box. To play live CDs I connected an Apple Super Drive That is also supported by rooPlay.

These few devices are the whole technical stuff I need and I like what I hear :cool:
0286029F-23A3-4DF7-8D2E-46A70611CBDC.jpeg


Best DrCWO
 

MCH

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I am really impressed about your setup. Thanks‘ for sharing :)
Live playback for LPs in a digital system is always a challenge.
As I also have a nice solution for LP playback I like to share it here.

I run a 2x2 active system with a pair of full range horns and two woofers. XO and DRC are done with convolution filters in Roon. Playback via the new RME ADI-2/4 SE. It has two stereo DACs so I can drive Horns and Woffers in parallel from Roon.

But how to handle LPs? As I am the developer of the rooExtend-box I programmed a Roon Extension for it I called rooPlay. With rooPlay I can connect my turntable directly to the ADI-2/4 that also offers great RIAA support and stream it via Roon and its DSPs to the DACs of the ADI-2/4.

For me this was the perfect solution. I only need the ADI-2/4, the rooExtend-Box that I connect directly to the active Woofers and my Krell 2250e driving the horns. Wireless remote control is done with a Nuimo and my rooNuimo Ronn Extension also hosted by the rooExtend-Box. To play live CDs I connected an Apple Super Drive That is also supported by rooPlay.

These few devices are the whole technical stuff I need and I like what I hear :cool: View attachment 251066

Best DrCWO
Cool!
Can also be done easily and for free with LMS.
 
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dualazmak

dualazmak

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I am really impressed about your setup. Thanks‘ for sharing :)
Live playback for LPs in a digital system is always a challenge.
As I also have a nice solution for LP playback I like to share it here.

I run a 2x2 active system with a pair of full range horns and two woofers. XO and DRC are done with convolution filters in Roon. Playback via the new RME ADI-2/4 SE. It has two stereo DACs so I can drive Horns and Woffers in parallel from Roon.

But how to handle LPs? As I am the developer of the rooExtend-box I programmed a Roon Extension for it I called rooPlay. With rooPlay I can connect my turntable directly to the ADI-2/4 that also offers great RIAA support and stream it via Roon and its DSPs to the DACs of the ADI-2/4.

For me this was the perfect solution. I only need the ADI-2/4, the rooExtend-Box that I connect directly to the active Woofers and my Krell 2250e driving the horns. Wireless remote control is done with a Nuimo and my rooNuimo Ronn Extension also hosted by the rooExtend-Box. To play live CDs I connected an Apple Super Drive That is also supported by rooPlay.

These few devices are the whole technical stuff I need and I like what I hear :cool: View attachment 251066

Best DrCWO

Hello @DrCWO,

Welcome to my project thread, and thank you for sharing your solution/setup for real time LP playback in your digital audio system.

In my case, I was quite lucky that I already have/had all the components except for one phono preamp for revival of DP-57L + DL-301 II (MC) in my already well established DSP multichannel multi-driver multi-amplifier active stereo audio system.

I believe our communication and info exchange would be "nice and worthwhile reference" for many of ASR friends still love real time on-the-fly listening to LPs even in digital software-DSP-controlled audio system.
 
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DrCWO

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Hello @DrCWO,

Welcome to my project thread, and thank you for sharing your solution/setup for real time LP playback in your digital audio system.

In my case, I was quite lucky that I already have/had all the components except for one phono preamp for revival of DP-57L + DL-301 II (MC) in my already well established DSP multichannel multi-driver multi-amplifier active stereo audio system.

I believe our communication and info exchange would be nice and worthwhile reference for many of ASR friends still love real time on-the-fly listening to LPs even in digital software-DSP-controlled audio system.
Thanks' for responding.

I forgot to mention one thing:
For generating the FIRs for XO and DRC Woofers and the Horns I used the Acourate software hat also delivers phase correction and therefore creates a perfect step-response. I attached the Roon playback path below. As you can see I re-sample all to 96kHz. In the signal path you also can see the 8-channel convolution filter (only four of them are used). The others I set to zero. Not doing this the other channels of the ADI-2/4 will show full volume and I disliked that.

1671447532004.png


And this is how it looked like from the listening position...
IMG_4790.JPG
 
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dualazmak

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Thanks' for responding.

I forgot to mention one thing:
For generating the FIRs for XO and DRC Woofers and the Horns I used the Acourate software hat also delivers phase correction and therefore creates a perfect step-response. I attached the Roon playback path below. As you can see I re-sample all to 96kHz. In the signal path you also can see the 8-channel convolution filter (only four of them are used). The others I set to zero. Not doing this the other channels of the ADI-2/4 will show full volume and I disliked that.

View attachment 251072

And this is how it looked like from the listening position...
View attachment 251074

Thank you for your further info on wonderful beautiful and simple setup including the paint/pictures!

I too resample all the tracks in 88.2 kHz or 96 kHz by JRiver MC to be fed into software DSP EKIO. Please find my "rationales" in my post here.
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532

BTW, have you objectively measured the time domain matching (I like to say "time alignment") all over the SP drivers, especially between your full range horns and sub-woofers?

In my setup, I explored rather intensive "method development", "precision measurements" and "fine tuning" in terms of "time alignment" ("wave matching") all over my SP drivers; i.e. sub-woofers, woofers, squawkers (mid range drivers), tweeters, super-tweeters as summarized in my post #520;
- Perfect (0.1 msec precision) time alignment of all the SP drivers greatly contributes to amazing disappearance of SPs, tightness and cleanliness of the sound, and superior 3D sound stage: #520

If possible, would you please share with us the Fq response (SPL) data measured by measurement microphone at your listening position?
For your reference, here is mine;
WS00005125.JPG
 
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DrCWO

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BTW, have you objectively measured the time domain matching (I like to say "time alignment") all over the SP drivers, especially between your full range horns and sub-woofers?
Yes I have :cool:
See my step-response first 15ms at the listening position 3,5m in front of the speakers (red=left, green=right).
1671458105230.png

I also attached a paper how linear phase crossovers can be built with IIR filters. This approach is called subtraktive delayed which I used.

The horns run without any crossover and cover the complete spectrum. By design frequency response drops here below 130Hz.
The woofers have DSP built in and a corner frequency of 170Hz with a 4. degree LR characteristics.

To achieve the linear phase I measured the rising edge of the woofers generated by their crossovers. From this I calculated an all-pass that has an exactly reversed polarity of this initial woofer slope. I put the calculated all-pass in front of the horns. With exact time and amplitude match of both, horns and woofers, both signals perfectly add resulting in a (nearly) perfect step as described in the attached paper. The rest is a bit nice amplitude and group delay processing by Acourate...
 

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  • A New High Slope, Linear Phase Crossover Using the Subtractive Delayed Approach.pdf
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dualazmak

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Hello again @DrCWO and ASR friends,

It is always great pleasure and very much informative for me (and for our friends visiting this thread, I believe) knowing other people's digital DSP audio setup together with the theoretical background.

I have slightly different approach comparing to yours; I use DSP (XO/EQ/delay) software EKIO as "system wide" DSP center within my PC which can receive digital audio signal through ASIO routing from any other audio-output software including Roon, JRiver MC, Adobe Audition, Audacity, Web browsers, etc.

As I wrote here, EKIO uses IIR filters. The processing is done using a cascade of second order transposed direct form II biquad sections. Every calculation is done using 64 bit floating point numbers.

Furthermore, I use rather "mild" second-order (-12 dB/Oct slope) LR (Linkwitz-Riley) filters in my XO configuration except for the sharp high-cut (low-pass) -48 dB/Oct UHF noise cut-off filter at 25 kHz. The reason for my use of -12 dB/Oct mild filters is that I simulate Yamaha's original passive LCR network in my active configuration; I have been using Yamaha NS-1000 cabinet and its three SP drivers throughout this project. Of course, I have tested higher order filters in XO configuration, and concluded that -12 dB/Oct filters should be the best choices at least for my NS-1000 cabinet and drivers (please refer to my early post here). In case if I would use different main SP (and SP drivers) other than NS-1000, I will/should start a new project thread, I believe.


Let me touch on the "nature" of my "naive/primitive but steady" approaches in measurements and fine tuning throughout this project.

My fundamental policy would be "Always use my own well understandable validated reproducible measurement methods in each of my go-up steps (do not over trust existing black-box type measurement tools/software unless I would fully validate and understand it)."

Just as one typical example, my policy in Fq response measurements; nowadays, I always do not like psychoacoustic smoothing* using very short/rapid sine sweep data over all the SP drivers; it gives too much smoothing with much statistical deviations/fluctuations based on poor (little amount of) air sound data, and hence lack of details of Fq responses of the drivers as well as of the room acoustics.
*In REW user manual: "Psychoacoustic smoothing uses 1/3 octave below 100Hz, 1/6 octave above 1 kHz and varies from 1/3 octave to 1/6 octave between 100 Hz and 1 kHz. It also applies more weighting to peaks by using a cubic mean (cube root of the average of the cubed values) to produce a plot that more closely corresponds to the perceived frequency response."

Any of Fq response measurements includes FFT analysis of the recorded sound data, and FFT Fq analysis is a "statistical procedure/calculation" on given raw data; the "statistical" accuracy/precision/resolution are greatly dependent on the "richness" of the raw sound data; the higher richness in raw data the better accuracy/precision/resolution/reproducibility of FFT Fq response results.

I fully evaluated and validated, therefore, my rather naive/primitive "cumulative white noise averaging method" (please refer to post #392 and #393) for my Fq response measurements; the pros and merits were summarized here in my post #404 where I wrote as follows.

1. the method is universally applicable in the stages of digital out of crossover software (EKIO), DAC's analog out, amplifier SP out, and of course in the actual room SP sound,

2. the method is accurate, sensitive and reproducible, having little or no statistical fluctuation, because of the FFT averaging analysis on the "accumulated rich data" of the recorded sound,

3. the recorded "white noise tracks" can be re-analyzed any way, anytime, afterwards,

4. flexible mix-paste (sound mixing) can be done to virtually simulate any combination of the channels, especially in amplifiers' SP out signals before going into SP drivers,

5. if needed, the environmental "continuous room back ground noise" can be reduced/removed by the Adobe Audition's "noise capture - noise reduction" function,

6. if needed, suitable gain/level adjustment can be applied for "level matched comparison" of Fq response shapes between the different series of the recorded data,


7.
flexible and suitable FFT size (as smoothing intensity) can be selected depending on the frequency zone of interest.

Furthermore, I believe the "cumulative white noise averaging" would be more practically suitable than "pure sine sweep" since our SP drivers always receive mixture of various Fq sound signals in our actual music listening situation, not single Fq sine wave.


I also applied such my "fundamental policy" in the measurements and fine tuning of "time alignment" all over the SP drivers, as I summarized here.
And, the same for my DIY 12-VU-Meter Array, and for revival of analog LP Player in my DSP multichannel multi-amplifier system.

My "fundamental policy" throughout this thread also includes the followings;

- First, I (we) need to establish my reference sound system to which I can roll back any time during my exploration (in my case it has been passive single amp stereo system with Accuphase E-460)

- Go up (climb up) step-by-step carefully, not in a hurry

- Do not change multiple parameters/configurations at once; it may cause inter-cancellation of pros and cons to give confusion

- In each step, I should trust my ears and brain in subjective listening test using my own common/consistent preferable "audio sampler music playlist" consists of excellent-recording-quality tracks selected from various genres/categories (like shared
here and here)



In any way, I would highly appreciate all of your continuing kind attention and further participation(s) in this rather long-lasting still-on-going project thread.
 
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See my step-response first 15ms at the listening position 3,5m in front of the speakers (red=left, green=right).
1671458105230.png

I understand your "step-response" data (diagram) as "transient characteristic" of your whole SP system.

What was the stimulation pulse in your step-response measurement, the time width, wave form, stimulation Fq, etc.?

I am greatly interested in "step-response" (transient characteristics) of your system especially stimulated by 8-wave or 3-wave rectangular tone burst of specific frequency for which your sub-woofer and full-range-horn sing together, I mean at the center of the overlapped crossover Fq area (150 Hz in your setup?). I would highly appreciate if you could show such transition-characteristic (step-response) data for your sub-woofer only, for full-range-horn only, and of course for sub-woofer + full-range-horn, measured at your listening position.

Edit:
I assume such "room air wave shape observations/measurements" would confirm and validate your theoretical calculation for crossover settings.


Just for your reference, I did such kind of "transient characteristics (step-response)" measurements and fine tuning for my woofers and sub-woofers as shared in my posts #495, #503, #507; I applied 8-wave and 3-wave rectangular tone burst stimulation signals consist of 63 Hz, 125 Hz, 250 Hz, 500Hz, 1 kHz for woofer, and 31.5 Hz and 63 Hz for my sub-woofer, for assessment of their transient behaviors and selection of XO Fq (in my case 50 Hz) between sub-woofer and woofer, as well as for the validation of my time alignment setting.

In case if you would be interested, I can share all of my test tone signals which I utilized in my time-alignment and transient-characteristics (step-response) measurements.
 
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Holmz

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I understand your "step-response" data (diagram) as "transient characteristic" of your whole SP system.

What was the stimulation pulse in your step-response measurement, the time width, wave form, stimulation Fq, etc.?

I would be greatly interested in "Step-response" (transient characteristics) of your system especially stimulated by 8-wave or 3-wave rectangular tone burst of specific frequency for which your sub-woofer and full-range-horn sing together, I mean at the center of the overlapped crossover Fq area. I would highly appreciate if you could show such tesp-response data for your sub-woofer only, full-range-horn only, and of course subwoofer+full-range horn, measured at your listening position.

Just for your reference, I did such kind of "transient characteristics" measurements and fine tuning for my woofers and sub-woofers as shared in my posts #495, #503, #507; I applied 8-wave and 3-wave rectangular tone burst stimulation signals consist of 63 Hz, 125 Hz, 250 Hz, 500Hz, 1 kHz for woofer, and 31.5 Hz and 63 Hz for my sub-woofer, for assessment of their transient behaviors and selection of XO Fq (in my case 50 Hz) between sub-woofer and woofer as well as for the validation of my time alignment setting.

Usually there is no 8 tone nor 3 tone deal… it is just 0V gong to 1V...
 
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Usually there is no 8 tone nor 3 tone deal… it is just 0V gong to 1V...

OK, please further educate me. In such stimulation of 0V to1V pulse in very short time window, what is/was the Fq composition(s) of the pulse? Just pure 1 kHz pulse? Or, is it white noise or pink noise?
 
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Holmz

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OK, please further educate me. In such stimulation of 0V to1V pulse in very short time window, what is/was the Fq composition(s) of the pulse? Just pure 1 kHz pulse? Or, is it white noise or pink noise?

Think of it as one step… so zero volts from the big bang until now, and then going to 1V and hanging there till the end of time.

So in frequencies, it is all freqs from DC to daylight.
However in the time domain, it is like running a bicycle into (and up) a curb, and then riding on the sidewalk.

The woofer will accelerate endlessly with the 1V, except for the compliance and running out of throw.

But there is usually some cross over in the box to prevent DC from entering into it, so that is why the SPL ends up sagging down to to DC, or dropping below it as the woofer relaxes and decelerates.
Even if it hits the stops and slams to a stop, then it is decelerating and causing a rarefication of the air, or a negative preseeure or negative pressure part if the S P L.

In reality they usually pulse the square waves in… probably at 25-50/second (dunno), which would give 20 msec of measurement... and then we have gated measurements, and one can average them for a bit of SNR improvement.
 
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Hello again @Holmz,

I really appreciate your above kind educational message; I almost understood it.

I assume, consequently, the data given by such "step-response" measurement would be greatly different from my interests on "transient characteristics" of SP drivers at various frequencies stimulated by pulse tone burst signals consist of various frequencies.

And, I also feel/assume such "step-response" measurement with 1V "push" would be somewhat dangerous (or possibly harmful?) for SP drivers especially if there is no LCR passive crossover (or protection capacitors), right?
 

Holmz

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Hello again @Holmz,

I really appreciate your above kind educational message; I almost understood it.

Here is some better worded stuff.



I assume, consequently, the data given by such "step-response" measurement would be greatly different from my interests on "transient characteristics" of SP drivers at various frequencies stimulated by pulse tone burst signals consist of various frequencies.

Usually a perfect step response is also mirrored with a perfect impulse response.
If some drivers are wired with the polarity flipped then the step function does a lot of up and down.


And, I also feel/assume such "step-response" measurement with 1V "push" would be somewhat dangerous (or possibly harmful?) for SP drivers especially if there is no LCR passive crossover (or protection capacitors), right?

We would have to find the specifics, or put in a protection cap for each driver.
But if you have active XO, then it should be sort of boring compared to a high order passive XO.
(But worth checking, IMO)



 
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DrCWO

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Please see the attached paper from Angelo Farina, pubished by Audio Engineering Society 2007.
It describes how today impulse responses got measured. Integrating the impulse response gives the step response.

Best DrCWO
 

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  • Advancements in impulse response measurements by sine sweeps.pdf
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dualazmak

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Usually a perfect step response is also mirrored with a perfect impulse response.
Thank you for the kind additional educational response. Now I understand this point in terms of;
Integrating the impulse response gives the step response.

So both of "impulse response" and "step response" are reflecting the SP behaviour from different angle/perspective, and essentially they are mathematically/theoretically interchangeable with each other, right?

Please see the attached paper from Angelo Farina, pubished by Audio Engineering Society 2007
The paper is really educational and interesting, thank you for your kind response sharing this article.

Even though a little bit out of the scope of my current learning process on fundamentals of "step response" and "impulse response", I am very much impressed by the Section 3.3. "Pulse noises during the measurement" describing several approaches (algorithms) for elimination of pulsive artifacts which are now widely applied for (even on-the-fly real time) auto Click/Pop elimination in several audio playback devices and software.


BTW, just for my reference and naive curiosity, I would be much interested in total Fq response shape (15 Hz - 22 kHz), if available, of your very unique SP setup with "non-XO" full-range-horns plus sub-woofers, at your listening position in your room acoustics.
 
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DrCWO

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Thank you for the kind additional educational response. Now I understand this point in terms of;


So both of "impulse response" and "step response" are reflecting the SP behaviour from different angle/perspective, and essentially they are mathematically/theoretically interchangeable with each other, right?


The paper is really educational and interesting, thank you for your kind response sharing this article.

Even though a little bit out of the scope of my current learning process on fundamentals of "step response" and "impulse response", I am very much impressed by the Section 3.3. "Pulse noises during the measurement" describing several approaches (algorithms) for elimination of pulsive artifacts which are now widely applied for (even on-the-fly real time) auto Click/Pop elimination in several audio playback devices and software.


BTW, just for my reference and naive curiosity, I would be much interested in total Fq response shape (15 Hz - 22 kHz), if available, of your very unique SP setup with "non-XO" full-range-horns plus sub-woofers, at your listening position in your room acoustics.
Here it is:
Red and Green the raw FFT 64k long, Black and Blue smoothed with a frequency dependent window.
The smoothed response is within a rangeof 2dB.
All measured at the listening position approx. 3,5m from the horns.

Step and impulse response are interchangeable. Step response in the integrated impulse response.

Best DrCWO
1671727347385.png
 

Holmz

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So both of "impulse response" and "step response" are reflecting the SP behaviour from different angle/perspective, and essentially they are mathematically/theoretically interchangeable with each other, right?

I am not sure… but they certainly seem to be somewhat related.
In general a step response that is negative, should also have an the impulse response that is negative.
  • step response is the pressure field “over time” (of a step)
  • impulse response is wide band response “with respect to time”… (and a narrow Dirac delta function.
maybe a step up response, and a step down response… fully capture the impulse response in a mathamatical sense?


The impulse response is in amplitude versus time, for a wide band signal.

The step response however, shows the pressure field versus time… so one assumes that the left side is higher frequencies and as we go to the right, we get MR and woofer kicking in.

So the step response is pressure happening “over time”, whereas the impulse response is signal “with respect to time”.


The paper is really educational and interesting, thank you for your kind response sharing this article.

My pleasure sir.


Even though a little bit out of the scope of my current learning process on fundamentals of "step response" and "impulse response", I am very much impressed by the Section 3.3. "Pulse noises during the measurement" describing several approaches (algorithms) for elimination of pulsive artifacts which are now widely applied for (even on-the-fly real time) auto Click/Pop elimination in several audio playback devices and software.

Step response gives a better picture of whether the phase (polarity) is 180 degrees out.
(especially with many drivers.)
 
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Here it is:
Red and Green the raw FFT 64k long, Black and Blue smoothed with a frequency dependent window.
The smoothed response is within a rangeof 2dB.
All measured at the listening position approx. 3,5m from the horns.

Step and impulse response are interchangeable. Step response in the integrated impulse response.

Best DrCWO
View attachment 251731

Hello again @DrCWO,

I thank you so much for your sharing the total Fq response data of your unique full-range-horn + sub-woofer system. As I thought and expected, the shape of Fq response looks very nice and impressive.

You wrote;
>smoothed with a frequency dependent window
Is this a kind of psychoacoustic smoothing?

In my Fq response measurement attached below, the recorded "cumulative flat white noise" (for 60 sec) was analyzed by Adobe Audition 3.0.1's frequency analyzer by scanning (averaging) the 60 sec recorded band with FFT size = 4096 for all over the 20 Hz - 20 kHz (Blackmann-Harris window), therefore no Fq dependent smoothing applied. It means that the Fq response curve (in very good reproducibility) well reflects the details of room acoustics including reflections, resonances, etc. by my listening room and furnitures. Please refer to my posts #404, #405-#409 for the details.

(If I use smaller FFT size like 1024 for stronger smoothing, I can easily smooth my Fq response curve in 500 Hz - 20 kHz area to make it within a rage of 2 dB just like yours.)

It is also a little bit of my surprize and interest that the slightly downward "tendency" of your Fq response is quite similar to that of my system except for the slight upward slope over 6 kHz in my case compensating my (and wife's) age dependent slight hearing decline in high Fq (please note that the smoothing is different for the two systems/measurements);
WS00005185.JPG


Consequently, I can rather easily "imagine" the wonderful sound of your system.

As for my "flexible" (even on-the-fly real-time) upward Fq response control in high Fq zone (over 7 kHz), please refer to my post #643 for the details.
 
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3ll3d00d

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Is this a kind of psychoacoustic smoothing?
FDW as used by acourate is described in https://www.audiovero.de/acourate-w...ionen:td-functions:frequency_dependent_window
It just means that the length of the window applied to signal varies with frequency, acourate further allows the actual length at low (16Hz) and high frequencies (fs/2) to be specified as different values (no of cycles) and it then interpolates between those two points to find the actual no of cycles to use at a particular frequency.
REW has a similar function but with a fixed no of cycles across the range.
Acourate typically applies FDW in conjunction with a psychoacoustic smoothing https://www.audiovero.de/acourate-wiki/doku.php?id=en:wiki:funktionen:td-functions:psychoacoustics which has the effect of trimming the dips before the FDW is applied.
 
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