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

tpaxadpom

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thanks for sharing your latest findings.
1) Have you consider rearranging your speaker placement along the long wall? According to the pictures posted you will get better symmetry that way. You have a system in the living room so placing acoustic panels/diffusors could be a challenge but considering how much you have invested (time/money) room acoustics would be my next area to concentrate on. The room is the most important component of your stereo setup.
2) If speaker rearrangement is not on the table, I would move the subwoofers and place them on the side of your couch. It would flatten the bass response. I would also create a convolution filter for low frequencies <300Hz and experiment with that. If anything I would at least apply room correction for the subwoofers. Thankfully you have perfect setup to make it all happen.
3) Another area to experiment with would be raising the crossover frequency between your subs and main speakers.
 
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dualazmak

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Hello @tpaxadpom,

Thank you for your nice suggestions, as always!

As you could find in my above posts # #493 and #494, I set 16 ms delay in ST+TW+SQ+WO against SW. This corresponds to that I pushed my SWs (sob-woofers) 5.5 m ( 345 m/s x 0.016 s) ahead towards my listening position.

At present, since I like the listening feeling as if I am sitting on the best S-class center seat in Concertgebouw Amsterdam, I do not like to have my SWs just beside me at my listening position; even with the "complete and perfect" time alignment of SPs I achieved, the orchestral big drum sound should be coming from the stage direction in front of my eyes and ears!

Of course, I know well that in real Concertgebouw Amsterdam (I have been there several times), I hear the entire hall tone (including the reflections, resonances, standing-waves) from all the directions surrounding me, from behind, above (ceilings), side (side walls) and floor. This is the main reasons that I insists we need suitable and preferable reverberations also in our home audio listening room which somewhat "simulates" the real hall tone, but never never to be perfect.

In my listening environment, I have fairly nice and big open spaces/rooms behind the SPs and also behind my listening position, and these (I designed so when I built my present house) are very nice for the room acoustics I fully enjoy now. Fortunately, many of the semi-professional audio enthu friends well agree me on the nice acoustics even in my present setup.

As you kindly suggested, I may try some fine tuning adjustments of physical alignment of SPs and/or furnitures, as well as putting acoustic treatment material(s) near to the portion of "the wall"; I mean the wall which I identified causing very subtle amount of standing waves in certain frequency, but only very faintly audible with our ears with unusually high volume/gain sounds. (Measurement microphone ECM8000 is really sensitive enough to hear and see the standing waves in 35 - 45 dB below the main sound, though.)

I feel/assume that I and my wife,as well as my audio enthu friends, even unconsciously "love" these kind of faint standing waves in our listening environment, audible or inaudible, while enjoying music.
 
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dualazmak

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Hello again @tpaxadpom,

As for your kind suggestion of;
3) Another area to experiment with would be raising the crossover frequency between your subs and main speakers.

I also have thought about it, but decided not to do so, since I measured and observed the data in this diagram which I also shared in my post #495;
WS003153.JPG


In 31.5 Hz, SW well responds to give rather pure and tight 31.5 Hz sound. In 63 Hz, however, together with the expected 63 Hz sound, it gives prolonged and delayed lower frequency "Helmholtz resonance" sound/noise, even in fairly low gain though, but may affect somewhat deteriorating the "tightness" of the 63 kHz sound.

Consequently, I decided to continue using SWs as in my present XO configurations crossing over to WOs at about 50 Hz; down to 50 Hz, my Yamaha 30 cm WO, directly and dedicatedly driven by powerful amplifier A-S3000, works quite nicely, as I shared in my previous posts.
 
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dualazmak

dualazmak

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Precision measurement and adjustment of time alignment for speaker (SP) units
Part-3: Precision single sine wave matching method in 0.1 msec accuracy


Abbreviations
in this post;
SW: sub-woofer, Yamaha YST-SW1000 (L & R)
WO: woofer, Yamaha NS-1000’s 30 cm JA-3058
SQ: squawker, Yamaha NS-1000’s 8.8 cm Beryllium dome JA-0801
TW: tweeter, Yamaha NS-1000’s 3 cm Beryllium dome JA-0513
ST: super-tweeter, Fostex metal horn T925A

Hello friends,

This post is a follow-up of my recent post #493 "Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1: Precision pulse wave matching method", and post #494 " Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2: Energy peak matching method"

After the successful "pulse wave matching" mainly for SW-to-WO in 1 msec (ms) precision shared in the above posts, now I would like to apply the same approach in 0.1 ms accuracy/precision for WO-to-SQ time alignment adjustment.

For this purpose, instead of the 8-wave tone bursts I used in the Part-1 post #493, I prepared a new series of single sine wave pulses in various frequencies for easier and simpler precision measurement, analysis and time alignment adjustment in 0.1 ms accuracy;
WS003303.JPG


The single sine wave of 500 Hz is included for the detailed measurement and analysis at the crossover Fq 500 Hz for WO to SQ in my present multichannel multi-driver multi-amplifier setup. I am really interested in this issue since the WO is a 30 cm cone driver and the SQ is 8.8 cm very light-weight Beryllium dome; there is a very large difference in "inertial mass" (mass of moving parts) between the two crossover drivers.

I first recorded the sound of SQ-only and WO-only at 500 Hz by stimulating with the 500 Hz single sine wave, where the initial kick-down (input of the sine wave) time point was set exactly identical at 3.340 sec position;
WS003304.JPG


As shown in above diagram, even with the single wave stimulation, both of SQ and WO give two and one aftershocks, respectively, which are reasonable and understandable.

My main interest is that, due to the very large difference in "inertial mass" (mass of moving parts) between the two drivers, the bottoms and peaks given by WO sound delay in 0.3 ms to 0.7 ms against SQ.

If I would like to have complete/perfect time alignment between WO and SQ, therefore, I need to "relatively delay" the SQ sound against WO in 0.1 ms precision/accuracy which I can control by digital crossover software EKIO's group delay controller.

I actually did it by 0.1 ms step relative delays in SQ sound from zero to 0.9 ms as shown here;
WS003309.JPG


As you can clearly see in the above diagram, the "0.3 ms relative delay in SQ sound against WO" gives the best match, i.e. the best time alignment, both in the overlapped (mixed) gain scale and in the time domain.

I also applied this "single wave matching" in measuring and analyzing the SW-to-WO crossover at 50 Hz for which I already have found in my posts #493 and #494 that "16.0 ms delay in WO sound against SW" gives the best match, best time alignment;
WS003310.JPG


Again, I found and confirmed, this time with single wave stimulation, that the "16.0 ms delay in WO sound against SW" gives the best time alignment.

Here, I will not share my detailed measurements and analyses well confirming that there is no delay at all, in 0.1 ms precision, between SQ and TW, as well as TW and ST, recorded at 1.5 m from the surface of the SP drivers.

Let me also inform you that all the results in this post were obtained at 1.5 m from my left SPs, and I also found exactly the same results at 1.5 m from my right SPs.

Consequently, as the results of my precision time alignment measurements and analyses shared in the three posts #493, #494, #504 (this post), I could conclude that the best (perfect) time alignment (delay) setting in EKIO's group delay controllers in my present audio setup must be;
SW: zero delay
WO: 16.0 ms delay
SQ+TW+ST: 16.3 ms delay


WS003317.JPG


The frequency response of the total sound, of course, remains unchanged;
WS003316.JPG


and,
WS003315.JPG
 
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mikessi

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Great work in examining all of these details in your system & making corrections! Such precise time alignment is probably audibly beneficial at the higher frequencies.

One question: The wavelengths of low bass frequencies (say 60Hz and lower) are long -- ie, A 20 Hz wave has a cycle time (360 degrees) of 50ms and is about 56 ft long; a 60 Hz wave has a cycle time of 16.6ms & is ~18' long. While I appreciate that 1ms time alignment between SW and WO is theoretically preferable to a longer delay, given the wavelengths and the size of your room/distance from the speakers, I wonder whether it is possible to perceive any improvement after the adjustment?
 
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dualazmak

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.....
One question: The wavelengths of low bass frequencies (say 60Hz and lower) are long -- ie, A 20 Hz wave has a cycle time (360 degrees) of 50ms and is about 56 ft long; a 60 Hz wave has a cycle time of 16.6ms & is ~18' long. While I appreciate that 1ms time alignment between SW and WO is theoretically preferable to a longer delay, given the wavelengths and the size of your room/distance from the speakers, I wonder whether it is possible to perceive any improvement after the adjustment?

Hello @mikessi,

Thank you as always for your nice comments.

I understand and essentially agree with your point that the low Fq area, less than ca. 80 Hz, the nature of room acoustics greatly affects our real listening experiences and sensations. In my environment, the open spaces (rooms) behind the SPs and also behind the listening position are nicely contributing to the fairly enjoyable music listening, I believe.

You would please note that my delay (time alignment) adjustment on WO to be matched with SW is 16 ms (not 1 ms) which is identical to that I would physically move the SWs 5.5 m ( 345 m/s x 0.016 s = 5.5 m) ahead towards listening position!

As I shared in the bottom portion of my previous post #493, my subjective impressions before-and-after the SW-to-WO time alignment would be somewhat placebo effect, especially in orchestral music recorded in a large hall.

As for the "measurement and fine tuning" point of view, which I feel nice sharing with other people, the "Fq-amplitude-time three dimensional sound energy distribution spectrum" would be also useful, especially in SW-to-WO time alignment. I really like the "spectral view" of Adobe Audition 3.01 which has flexible color spectrum controls including gamma setting, saturation, FFT size, etc.

Let me share here, therefore, one diagram comparing the "tightness" of the 63 Hz sound energy distribution before and after the adjustment;
WS003329.JPG


Whether we can subjectively differentiate the difference in "energy distribution tightness", or not, would be greatly depending on our room environment, as you kindly pointed.
 
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dualazmak

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Total system configuration and the best tuned Fq response; as of January 25, 2022

Hello friends,

Having the recent overhaul maintenance of super-tweeters T925A with the fine tuning of the signal into them (post #485), and also achieving the 0.1 msec precision perfect time alignment between all of the SP drivers (posts #493, #494, #504, #507), now I can subjectively hear and believe the further improved "total sound quality" of my multichannel multi-way multi-driver multi-amplifier stereo system.

I was also very much impressed by the measured still-amazingly-wonderful transient characteristic of the woofer Yamaha 30 cm JA-3058 in sealed NS-1000 cabinet directly and dedicatedly driven by powerful amplifier Yamaha A-S3000 (post #495).

Consequently, I tentatively suspended my possible "new woofer implementation plan" since the completely time-align sub-woofer plus woofer now gives really nice rigid solid tight low Fq sound.

I shall not change any further parameter and room acoustics at least for the coming a few months, and I will carefully listen to the sound and music with the significantly improved "total sound quality".

Now, therefore, I feel it would be worthwhile sharing my present total system setup as of January 25 2022, as follows;

WS003411.JPG


WS003406.JPG


WS003408.JPG

As for the high-cut (low-pass) -48 dB/Oct filters at 25 kHz, please refer to my posts #362-#386 on "near ultrasound - ultrasound ultra-high frequency (UHF) noises in improperly engineered/processed HiRes music tracks".

WS003404.JPG

Please refer to posts #248 and #251 for the fine tuning 22 Ohm resistors in SQ, TW and ST speaker level signals.

WS003410.JPG


WS003402.JPG


WS003401.JPG


WS003399.JPG


WS003400.JPG

As for the unique super tweeter physical alignment (positioning), please refer to #027.

WS003398.JPG


WS003397.JPG


For the details of my listening environment, you would please refer to the latter half of my post #311.

And, please find here (on this thread) and here (remote independent thread post) the Hyperlink Index of this thread as well as some of my related posts in remote threads.
 
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gene_stl

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Probably the most accurate stereo system on the planet , and undoubtedly one of the very few with its' accuracy PROVEN with microphone traces.

If more audiophiles, went to the amount of effort, that dualazmak has, there would be less opinion argument, and more discussion of facts.
 
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bennybbbx

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Precision measurement and adjustment of time alignment for speaker (SP) units
Part-1: Precision pulse wave matching method


Hello friends,

Abbreviations in this post;

SW: sub-woofer, Yamaha YST-SW1000 (L & R)
WO: woofer, Yamaha NS-1000’s 30 cm JA-3058
SQ: squawker, Yamaha NS-1000’s 8.8 cm Beryllium dome JA-0801
TW: tweeter, Yamaha NS-1000’s 3 cm Beryllium dome JA-0513
ST: super-tweeter, Fostex metal horn T925A

First of all, please note that this "time alignment discussion" here is limited to pure audio-only system, and excluding audio-visual system where you need "time alignment adjustment" not only for the SPs but also for visual images/movies.

You well know that, throughout this project thread, I have been using digital music players, such as JRiver, in PC, and feeding the digital signal in digital XO/EQ "EKIO" for crossover, and then sending the divided digital signals into DAC8PRO for multi-channel multi-driver multi-amplifier stereo music listening.

In the digital signal processing, we have so many buffers or latencies; JRiver output buffer, ASIO4ALL's I/O buffers, EKIO's processing buffer, DIYINHK USB ASIO driver's buffer, and so on. Consequently, it is not straightforward to exactly measure the "absolute delay" between the JRiver's "shout" and the final air sound kick-up by SP.

I usually set all the buffers in the digital domain in rather large size, so that I should not have any latency or delay problems; in our audio setup, we have no problem at all if all the bunch of the digital and analog signal (15Hz - 30 kHz) have identical amount of delay time from the signal origin at JRiver, and this is always the case in our digital (PC based) audio system.

The relative delay between the SP units, or "time alignment" in multiple SPs, however, is always one of the critical issues in audio system, especially the multichannel multi-driver multi-amplifier system, as you may agree.

I have been always take my attention and care on this issue, and in my very early posts #18 through #21, by using REW's wavelet analysis, I briefly checked that all of my SP units, super-tweeter (ST), Be-tweeter (TW), Be-squawker (SQ) and woofer (WO) have essentially no delay with each other, while my sub-woofer (SW) has 10 - 20 ms delay against the other SP units.

Now, I became really would like to establish my own simple, reliable and reproducible precision method for "time alignment" or "relative delay" measurement, and fine adjustment(s) if needed.

For this purpose, I prepared one 6-second signal consists of multiple rectangular tone-burst (8 waves) signals of various frequencies in exact timing series with time-zero 15 kHz marker signal at 3.000 s time position;
View attachment 178981

Each of the start-up (kick-up) time positions was set exactly 200 ms intervals, except for the lowest 31. 5 Hz pulse set after 400 ms from the preceding 63 Hz pulse.
View attachment 178982

I may record the air sound of this signal by using a measurement microphone, BEHRINGER ECM-8000, and an audio interface TASCAM US-1x2HR in my second PC for the air sound recording and analyses. Again, the buffers and/or latencies of the recording system should have no problem, since the time alignment measurement would be done on relative time distance from the zero-time-marker, for relative delay assessments.

If the recorded sound of this signal has exactly the identical tone kick-up timings after the 15 kHz zero-marker, then all the SPs should have no relative delay; I can read/find the tone kick-up time position in sub-ms precision by enlarging the specific time area of the recorded sound by Adobe Audition 3.01 (or Audacity).

I should be careful enough, however, the positioning of the measurement microphone. The sound velocity in 20 degree-C temperature (my listening room now) is 344 m/s, and this means sound travels 34 cm/ms (milli-second). I fixed the microphone at 1.5 m from the surface of my SPs so that the sound traveling differences from the SPs to microphone is less than 5 cm or less than 0.2 ms, securing ms level accuracy/precision in my SP time alignment measurements.

All the data shared in this post were recorded at 1.5 m from my left SPs, and please note that the right SPs gave exactly the same results.


I first applied this method for precision measurement of sound delay with my sub-woofer, Yamaha YST-SW1000, as I already knew it has 10 ms - 20 ms delay. I played the prepared signal by JRiver, together with using the flexible "solo" buttons of digital crossover software EKIO; the highest frequency (Fq) L-panel was in solo for 15 kHz zero-time marker sound to be sung by L-super-tweeter, and the lowest Fq L-panel was in solo for 31.5 Hz and 63 Hz to be sung by L-sub-woofer.

The recorded sound track as a whole was easily time-shifted to adjust the kick-up timing of the zero-time marker at exactly 3.000 sec so that the time sequence of the recorded track would be identical to the original test track.

Incase if SW sound has no delay against the zero-marker, then the 63 Hz burst should start at 4.800 sec, and the 31.5 Hz burst at 5.200 sec. The precisely measured time points, however, were at 4.815 sec and 5.216 sec showing the SW sound delays in 15 - 16 ms;
View attachment 178983

As I use SW for 15 Hz - 55 Hz Fq zone, now I could precisely measure and confirm that the SW sound delays in 16 ms.

Then, as shown in above diagram, I delayed ST sound (and TW, SQ WO) in 16 ms by EKIO's numeric group delay function to be fully matched with the SW sound. As you can see, after the 16 ms delay of ST, and hence 16 ms delay of the zero-marker, the SW sounds appeared at exact 4.800 sec and 5.200 sec, means perfect time alignment was achieved for SW.

Having this 16 ms delay set in ST+TW+SQ+WO sound against SW, I measured the 6 sec test signal with all the SPs singing together, and again recorded the air sound for precise analysis. It was quite easy precisely identifying each of the kick-up timing of the tone bust signals in ms (millisecond) accuracy using time-enlarged view of Adobe Audition 3.01;
View attachment 178984

As shown in above diagram, the kick-up positions of the tone bursts were exactly identical to those of the input test signal means there is no relative delay at all in millisecond accuracy between the five SP units, ST, TW, SQ, WO and SW; perfect time alignment for all the SPs was established.

Furthermore, since now I have the 16 ms delay adjustment between WO and SW, I am very much curious about the tone burst wave shapes before and after the delay adjustment at the Fq area where WO and SW are singing together, which is given by the 63 Hz tone burst signal.

I carefully recorded the WO-only sound, SW-only sound, and WO+SW sound of the 63 Hz 8-wave tone burst signal, after and before the 16 ms delay adjustment for WO;
View attachment 178985

We should note that the "63 Hz, -10 dB gain, 8-wave excitation" for WO is a rather strong one, and as seen in the above diagram, WO gives one additional "inertia aftershock" followed by the proper 8-wave response.

On the other hand, the transient response of SW for 63 Hz tone burst signal is rather nice with 8 proper wave peaks as in the input signal, but followed by slower low-gain sound of inner air movements which can be easily understood in consideration of the unique "Helmholtz resonance" mechanism for bass boost with the heavy (48 kg) and rigid Yamaha YST-SW1000.

I will further share and discuss the "measured" transient characteristics of my WO and SW in my separate post(s) coming hopefully within a few weeks.

Here, it is just an accidental coincidence that the delay of 16 ms given to WO sound is identical to peak-to-peak interval of the input 8-wave 63 Hz excitation which is also 16 ms (8 waves in 128 ms period).

We can see and understand the shapes of WO+SW sound after and before the 16 ms delay in WO sound in the above diagram. In this case, the "before delay adjustment sound" has considerably longer singing time than the "after delay adjustment sound". In real music listening circumstances, much more complicated intermodulation may happen if there is significant delay (asynchronization) between WO sound and SW sound in overlapped Fq zone.

In any way, it should be better to have as perfect time alignment as possible throughout all the SP drivers. I decided, therefore, to have 16 ms delay settings in EKIO's output panels for WO, SQ, and TW+ST.

I am again very much impressed by Yamaha's original design and physical alignment of SP drivers in NS-1000's rigid cabinet with almost no relative delay between the Yamaha TW, SQ and WO.

Then, what would be the subjective difference in music listening before and after the fine "time alignment" tuning?

I should say the difference is minimal in my usual classical music listening, at least a few days after the adjustment, but today I felt tighter and rigid kick-up of large bass drum sound in the orchestra music, in Rachmaninoff Piano Concerto No.2 at the beginning and later-on in the third movement, even though it may be a placebo effect;

I feel the positive effects are more clearly heard in jazz music; especially the sharp kick-drums and low bass string sound which contains wide range of Fq spectrum including the finger and/or nail touching on the lowest (thickest) string;
and,


In my next post in a few days, I will share and discuss;
Precision measurement and adjustment of time alignment for speaker (SP) units
Part-2: Energy peak matching method

measure with burst signals i find very usefull. I have done this first. https://www.audiosciencereview.com/...ct-canton-look-much-better.17854/#post-580592 . but not with my newer speakers.people here tell i should better use measure software.

but i still think measure with measure software is not so real world and most speakers look nice but sound not good.. because they use continuos sinus and so there can not measure good what happen when come a burst(transients).

in your example with the short bursts better use a also a 1 sec burst to see the room influence. After the 5. wave the following waves should be exact the same. i think this can every speaker do. the error rate you can see on the following cycles how much they are diffrent. but you can see that the speakers i test have for the 1. cycle much more less level as 10% less to full
 
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dualazmak

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Hello @bennybbbx

Thank you for your visit on this thread and comments on time alignment measurements of SP drivers.

>but i still think measure with measure software is not so real world and most speakers look nice but sound not good.

I essentially agree with you.

As I recently wrote here, as far as the precision measurement and adjustment of time alignment, I prefer simple and straightforward measurement by using Adobe Audition or Audacity with no black-box type procedure included; we need to precisely measure the "relative delays", not the absolute delays, between the SP drivers in actual room air sound "only using the recorded air sound".

As for the tone burst signals to be used in time alignment and wave-shape matching, I found 8-wave sine burst in suitable (not too high) gain would be nice for low Fq zone, I mean excitation of sub-woofers and low Fq of woofers, as shared in my post #493 and #494. This is also suitable for observation of the tightness and cleanliness of the low Fq sound given by subwoofer and/or woofer, as shared in my post #503.

For the time alignment between woofer and midrange squawker, at 500 Hz - 1 kHz zone, on the other hand, the single sine wave pulse should be much suitable for precision time-position identification and also for the detailed comparison and matching of the wave shapes of the air sound in 0.1 msec precision, as shared in my post #504.

For the "measurement of transient characteristics of subwoofer and woofer", the comparison of the air-sound shapes with 8-wave and 3-wave excitations, in suitable (not too high) gain, were found to be really useful as shared in my post #495.

Furthermore, the 8-wave excitation of woofer in "unusual high gain" also well excites the room acoustics for visualization of reflecting waves or standing waves caused by a certain plane (or portion of a wall) which would be useful in identifying the sound reflecting plane or wall for possible sound insulation treatments if needed, as shared in my post #498 and #502.
 

bennybbbx

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I used for my tests studio one. there is a free version of it which is enough for this tests. https://shop.presonus.com/Studio-One-5-Prime it work as every DAW sample exact when play and record at same time when use ASIO drivers or asio for all because it is fullduplex. so can see the exact time in samples of delay.

the longer burst need not do always only for current microphone position to see if there maybe come lots reflections from cone or speaker case or too large distance. much reverb on the test signal look as this. this are much room reflection influence and not a speaker record. I do this with an reverb effect on sine tone to see how room influence look in measure signals. you can see on a perfect test there is near no lower level of 1 cycle see in compare to 2. or 3.rd cycle. on speakers can see much diffrence and on headphones too less

reverb 100.jpg
 
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dualazmak

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Hello again @bennybbbx

Thank you for sharing your experiences with Studio One.

From the wave chart you kindly shared above (but you did not show the time scale), can you identify or read the exact arrival timing of the first and second reflection waves at your microphone just like I measured using Adobe Audition in my post #498?
 
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bennybbbx

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the arriving time i have not measure. I do measure 1 cm away from woofer/mid to avoid room influence. I look only how the 1. wave cycle is in compare to 2. and 3. wave cycle to see how fast the speaker can react on transients. i try to find your measure tones wav. but i did not find them . where are they ? . then i can play them on my speakers. intresting to see how the results compare to your speaker results. i also like try your measure tones with headphones how they look

maybe it is better when distance to speaker is 3.43 cm . because this fit to audio speed 343 m/s better as 1 cm and is near too.
 
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dualazmak

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I prepared the test tone signals by myself as shared in my posts#493, #494, #495 and#504. I will be happy sharing them with you. Will contact you through the PM system soon....
 
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mecedo

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Thank you @dualazmak for sharing your knowledge. It helps me to save much time and avoid many mistakes in my journey with my audio setup optimization :)
 
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dualazmak

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Again, "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in HiRes music tracks, and EKIO's XO-EQ configuration to cut-off such noises

Hello friends,

I rather intensively investigated and shared this topic and my counter measure; "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in HiRes music tracks, in my posts #362-#386.

Although quite belated, today I noticed this important and interesting thread;
SOUND LIAISON, PCM DXD DSD free compare formats sampler. A new 2.0 version.
https://www.audiosciencereview.com/...pare-formats-sampler-a-new-2-0-version.23274/
where Sound Liaison generously described about sharing "free compare formats sampler. A new 2.0 version" for which they wrote that "We tried (and succeeded quite well) in recreating the sound the analogue vintage equipment added to the voice by using the tools we have in our digital workstation. But now without added high frequency noise."

These "free compare formats sampler. A new 2.0 version" tracks were newly prepared by Sound Liaison after amirm's specific post and amirm's YouTube video clip on Sound Liaison's original "compare formats sampler (1.0 version)".

Today I shared the following information in my post here on the interesting remote thread entitled "DSD is better than PCM!", and I do hope it would be allowed to share also in my project thread here, since the topic and information are still closely related to my system setup and configuration.

Now, I am just very much interested in "quality control, QC" of these "SOUND LIAISON free compare formats sampler 2.0 version", I quickly analyzed them by using MusicScope 2.1.0, as follows;
WS003465.JPG


WS003464.JPG


WS003463.JPG


We may clearly see that these "version 2.0 compare formats sampler" tracks contain much less UHF (ultra high frequency) "noises" in comparison with their original version 1.0 tracks.

We can also find, however, that the HiRes format tracks still have "sound" or "noise" components beyond 25 kHz, I mean in 25 kHz to 176.4 kHz frequency area. Since I (we) can hear up to ca. 22 kHz, the "value" (and "meaning") of the sound over 30 kHz would be the subject of our further discussion.

At least in my latest audio system setup and configuration, PCM 88.2 kHz WAV or FLAC 88.2 kHz (up to 44.1 kHz in each of the L & R stereo channels) would be "sufficient" enough, since the digital XO/EQ "EKIO" can work up to 192 kHz 24 bit and also having protective -48 dB/Oct high-cut (low-pass) LR filters at 25 kHz to eliminate any UHF sound/noises over 30 kHz (please refer to my posts #362-#386).

Consequently, even though my digital music library (ca. 25,000 tracks) consists of so many formats of DSD (8x, 4x, 2x,1x) FLAC (44.1 kHz- 352.8 kHz) and WAV/AIFF (44.1 kHz - 196 kHz), now I usually set "JRiver's DSP studio" in 88.2 kHz output of all the music tracks stereo 2-channel (on-the-fly conversion) as well as I set EKIO in 88.2 kHz sampling rate.

The simple reason for my current choice of 88.2 kHz, instead of 96 kHz, is that majority of the tracks in my music library is CD ripped non-compressed 44.1 kHz 16 bit AIFF format. As for my music library organization strategy and policy, you would please refer to my post here in a remote thread.

And, I should not forget about sharing the important subjective impression that all of the 15 tracks in "SOUND LIAISON, PCM DXD DSD free compare formats sampler. A new 2.0 version" sound really amazingly nice!
 
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dualazmak

dualazmak

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Hello friends,

I assume it would be somewhat worthwhile sharing my latest audio sampler playlist of 72 tracks (out of about 25,000 tracks in my digital music library) I currently use for check and fine-tuning of my system;
WS003470.JPG
 
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dualazmak

dualazmak

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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.

Abbreviations
in this post;
SW: sub-woofer, Yamaha YST-SW1000 (L & R)
WO: woofer, Yamaha NS-1000’s 30 cm JA-3058
SQ: squawker, Yamaha NS-1000’s 8.8 cm Beryllium dome JA-0801
TW: tweeter, Yamaha NS-1000’s 3 cm Beryllium dome JA-0513
ST: super-tweeter, Fostex metal horn T925A

Hello friends,

One month has passed since I established my latest system setup and configuration shared in my post #508, and we (my wife and me) have been intensively checking the total sound quality through our daily listening sessions using the latest audio sampler playlist shared in the above post #519 as well as many other beloved music tracks.

Yesterday evening, while having our enjoyable listening session, my wife encouraged me to write this post sharing our wonderful subjective impressions after the establishment of perfect (0.1 msec precision) time alignment of all the SP drivers (SW, WO, SQ, TW, ST: 5-way 10-channel) which already has been shared as follows;

- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507

Even though all of you may agree with me that it would be rather difficult to objectively measure and prove the "disappearance of SPs" in front of us in our listening environment, "the amazing disappearance of SPs" is our most astonishing subjective impression after establishing "the perfect (0.1 msec precision) time alignment of all the SP drivers".

This "SP disappearance phenomenon" reasonably resulted in wonderful, the best ever in our experiences, 3D (three dimensional) sound stage spreading in our stereo listening environment, which I assume all the recording engineers would like to present for all the listeners with all of the tracks and genres we listen to; from solo lute, solo soprano, solo piano, jazz trio, to full orchestra huge sound stage.

Furthermore, we believe that "the perfect (0.1 msec precision) time alignment of all the SP drivers" is also greatly contributing to the much improved sonority cleanliness and tightness of the total sound, as you may agree after reading through my posts #493 through #507; we can easily understand that "the better time alignment between the SP drivers" should effectively minimizes the intermodulation distortions of SP drivers especially around the crossover frequencies, and also it much contributes to tighter solid bass sound especially around the crossover between SWs to WOs. Our subjective hearing impressions in this regard, i.e. cleanliness and tightness, are again considerably exceeding our expectations.

Consequently, now I strongly believe that the establishment of precision (0.1 msec accuracy) time alignment between all of the SP drivers is one of the critical and important aspects (or conditions) in our multichannel multi-way multi-driver multi-amplifier stereo audio system.

At least for the coming several months (or longer period?), my latest system setup and configuration shall remain unchanged, and we will carefully continue our subjective hearing tests and listening sessions.
 
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