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

gene_stl

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I have seen photographs of broken Be domes. They are very stiff and brittle and shatter into pieces (which retain all the curvatures).
If I wanted to clean one, which I think is generally very inadvisable (jmo), I would invert them and squirt them with an anhydrous ethanol propanol mix. and then drive the alcohols off with warm air from a heat gun held at a distance. Without removing the cages.

Of course people can do whatever they want with their toys. There is a UK guy who had a British Racing Green finish put on his NS 1000 M and he had all the Driver frames painted and the phase gadgets too. They look GREAT.
 
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dualazmak

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aladar

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Dear friend what a wonderful project !
I have a very little setup (stereo and multichanne) and my big issue is with the connection with a single pair of speakers. Now I’m using a self made relè board, but not satisfied at all.
I’m very interested in your SP cabling board. Don’t see any physical or electronic switch on it. How do you change the signal from one system to the other ?
Ciao
 
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dualazmak

dualazmak

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Dear friend what a wonderful project !
I have a very little setup (stereo and multichanne) and my big issue is with the connection with a single pair of speakers. Now I’m using a self made relè board, but not satisfied at all.
I’m very interested in your SP cabling board. Don’t see any physical or electronic switch on it. How do you change the signal from one system to the other ?
Ciao

Hello,

By using a screwdriver! This is a primitive but most reliable method with no deterioration to sound quality.
- Speaker cabling board for single-amp and multi-amplifier setups: #004, #137, #250

Also, please refer to my post here and here; "Elimination of magnetic susceptible metals in SP signal handling" is really important and critical in HiFi audio setup.
 
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dualazmak

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Overhaul maintenance of super-tweeter FOSTEX T925A and further signal fine tuning thereafter

Hello friends, Happy New Year from Japan!

The super-tweeter (ST) Fostex T925A was launched in audio market in April 1994, and it has been keeping good reputation until today. It is amazing Fostex still has T925A in their product lineup, current price tag is JPYen 55,000 inclusive of 10% sales tax (somewhat cheaper in several web stores).
WS002898.JPG


Please refer to my post here for the unique physical alignment (positioning) of T925As in my system setup.

I purchased a pair of T925A in 1996, and have been using with great satisfactions over 25 years. Last month, early December 2021, I contacted with Fostex Company, with no expectation of positive response though, inquiring the possibility of overhaul maintenance of my T925As.

They very quickly responded; "We thank you and very proud of your long-year loyalty of T925A. We are very happy to give overhaul maintenance of your T925As!" Then my T925As were hospitalized at Fostex service center for a week, and came back to my home in 10 days. Fostex completely disassemble T925A, cleaned-up every parts, measured, and replaced some of them, including the inner thin connection wires, with new ones.

I very carefully checked the refreshed T925As confirming the better sound quality, and then intensively simulated/calculated the high-pass (low-cut) capacitor configuration in SP level signal going into T925As; now I decided using series of 3 microF and 10 microF capacitors as shown here;
WS002899.JPG


The series of 3 microF and 10 microF make total capacitance of 2.308 microF giving further -6 dB/Oct high-pass (low-cut) at ca. 8,611 Hz for the SP level signal going into T925A (impedance 8 Ohm) which already had -12 dB high-pass (low-cut) at 6,000 Hz by digital software crossover EKIO. (As for the use of 22 Ohm parallel resistors in the lines, please refer to my post here and here.)

All of the other elements of the configuration remain unchanged;
WS002900.JPG


Then, firstly, I measured the frequency response (FR) of the SP high-level signals after the capacitors, just before going into SP drivers, just like I did measured as shared here, this time using another Hi-to-Low converter Audio-Technica AT-HLC130 , and TEAC TASCOM US-1x2HR Audio Interface, and Adobe Audition 3.01 on the second Windows 10 Pro PC;
WS002908.JPG


The FR of SP high-level signals to go into the super-tweeters (STs) and tweeters (TWs) were "measured" in good conformity with my simulation/calculation;
WS002907.JPG


I also measured the SP high-level signals to go into beryllium squawkers (SQs), woofers (WOs), and the line-level signal to go into active sub-woofers giving this total and individual FR curves;
WS002903.JPG


Please note that the sensitivity (efficiency) of T925A is rather high in 108 dB (1m, 1W2.8V), and this means T925A sings 15 dB louder than the Yamaha 3cm Beryllium dome tweeter JA-0513 of 93 dB (1m, 1W2.8V) with a same gain/volume input signal.

As usual, I also measured the actual room air sound at listening position using the measurement microphone BEHRINGER ECM8000, just like I shared here, and here through here;
WS002904.JPG


The total and individual SP driver air sound FR curves at my listening position, with L & R SPs singing, were "measured" as shown in this diagram;
WS002905.JPG


Finally, the latest shape of the best tuned total FR;
WS002906.JPG


The above latest total FR as of January 2, 2022, is essentially identical to the one I shared here, but now I have the refreshed sound of T925A with better/improved variable gain control.
 
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tpaxadpom

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Good find of the dome cleaning video! Two questions for any Russian speakers here (I presume that's the language in the video? Correct me if I'm wrong!):

1. What's the liquid in the syringe he's using to soften the rubber ring seal/glue? Alcohol?
2. The liquid in the plastic bottle must be plain water? Maybe distilled water?

The guy seems almost cavalier using cotton pads to scrub the beryllium dome surface! It must be a lot tougher than I've thought before.
1. Acetone. Alcohol would not break the glue.
2. No. He is using isopropyl alcohol.
He is barely touching the dome during the cleaning process. "You cannot even talk about applying any pressure here..."
After I saw this video I thought of trying this on one of my sets, but decided not to. Mine are dirty but not as bad as the set in the video. Though I decided to clean straighten the metal mesh using the wire that is typically used to hold cables when you bundle them up (very thin metal with very wide flat dielectric). In the process a small metal piece brooke off and was caught by the midrange driver. It took me several ours to take it off. I didn't scratch the dome, but ended up in the cold sweat and perhaps developed some gray hair in the process. Using 2 toothpicks and magnet I was able to remove the piece of wire with help of my wife (need a third hand to hold the magnet).
 

tpaxadpom

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I have seen photographs of broken Be domes. They are very stiff and brittle and shatter into pieces (which retain all the curvatures).
If I wanted to clean one, which I think is generally very inadvisable (jmo), I would invert them and squirt them with an anhydrous ethanol propanol mix. and then drive the alcohols off with warm air from a heat gun held at a distance. Without removing the cages.
I doubt this approach would clean the dome. It took quite a bit of effort for the guy on video and required a mechanical help. In the process you would expose cloth suspension, probably not a deal breaker but still.
 

gene_stl

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I would not probably buy one that looked like it needed cleaning. I have eight midranges (Four bought new) and thirteen NS 500 tweeters, (ten bought new) and four NS1000 tweeters (bought on used NS1000Ms), that don't need cleaning.

If I wanted to buy a driver that needed cleaning it would lower my offer alot. I think it also depends on what is deposited on it. If it was in a smoking environment ages ago you are especially likely to be right. That stuff polymerizes.

It could be possible to invert the driver and flush only the beryllium surface with a strong solvent. If perhaps someone gave you some drivers for free. I would not want to disassemble them. (I regularly disassemble a LOT of tricky things, for a living). Although Acetone is a pretty strong solvent.

As long as I don't have a very good reason to do any of that disassembly I won't. We stopped smoking around our house about 1983.
 
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tpaxadpom

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I would not probably buy one that looked like it needed cleaning. I have eight midranges (Four bought new) and thirteen NS 500 tweeters, (ten bought new) and four NS1000 tweeters (bought on used NS1000Ms), that don't need cleaning.

If I wanted to buy a driver that needed cleaning it would lower my offer alot. I think it also depends on what is deposited on it. If it was in a smoking environment ages ago you are especially likely to be right. That stuff polymerizes.

It could be possible to invert the driver and flush only the beryllium surface with a strong solvent. If perhaps someone gave you some drivers for free. I would not want to disassemble them. (I regularly disassemble a LOT of tricky things, for a living). Although Acetone is a pretty strong solvent.

As long as I don't have a very good reason to do any of that disassembly I won't. We stopped smoking around our house about 1983.
Did you buy them new when they were still in stock? How do your DIY speakers image in comparison to NS1000M. My main set of speakers I have been listening to the most is DIY set with Yammy mid and tweet.
I wish someone gave me the dirty NS1000M drivers for free. I will take an offer you refuse and will be willing to pay a reasonable price. :) I don't think the process of disassembling is very difficult. I was worried more about not damaging the rubber ring more than denting/cracking Be driver. At one point I had to brak super glue joint I made with acetone, it wasn't an easy task but it worked. I am not sure if compressed air will get rid of the dust, but something I can try with NS1000M, other speakers are just too heavy to move down/up by myself.
 
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gene_stl

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Yes the ones I bought new were in 1977. and the early eighties for two of the NS500 ones.
I recorded some classical guitar concerts with crossed cardioids and when I play the tapes back the imaging is as good as I have ever heard. But I haven't compared them with NS1000M which I only got recently. I have been listening to them and like them very much.

If I run across some that I don't want I will pass the situation on to you if you wish. What country are you in?
 

tpaxadpom

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Yes the ones I bought new were in 1977. and the early eighties for two of the NS500 ones.
I recorded some classical guitar concerts with crossed cardioids and when I play the tapes back the imaging is as good as I have ever heard. But I haven't compared them with NS1000M which I only got recently. I have been listening to them and like them very much.

If I run across some that I don't want I will pass the situation on to you if you wish. What country are you in?
I am in US like yourself. There is something magical about these Yamaha drivers. They seem to transfer the emotions that are lost with other speakers. 20 years ago I would likely think they are not worth attention, but now, I simply enjoy the music. In fact I find myself listening to life recordings more than anything else. With other speakers I was blaming recording/mastering quality. Do you have pictures of your DIY speakers you can share with yamaha lovers?
 

gene_stl

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I used those drivers in my speakers unheard. I knew they had to be something special. In the mid 70s they were absolutely the only drivers that appealed strongly to me other than JBL. My speaker building buddy built an all JBL pair similar to some later ones that did not exist at the time.
An S8R system with a ten inch mid bass coupler. Tri Amped then quad amped. I bet a lot of money on the Yamahas and I am glad I won. If they hadn't been wonderful I would have been an angry camper.
 
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dualazmak

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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;
WS003122.JPG


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.
WS003121.JPG


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;
WS003120.JPG


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;
WS003118.JPG


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;
WS003119.JPG


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
 
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dualazmak

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Precision measurement and adjustment of time alignment for speaker (SP) units
Part-2: Energy peak matching method

Note:
The "time alignment measurement and adjustment" described in this post was so far only validated in my own audio setup. You need to, therefore, carefully validate this method in "your" audio setup by comparing the results with those given by the "Part-1: Precision pulse wave matching method", if you would like applying this method in your setup.

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,

The successful implementation of the "Part-1: Precision pulse wave matching method" shared in my above post #493 encouraged me evaluating a further simplified "semi-quantitative visual assessment method" for measurement and adjustment of time-alignment for multiple SP units.

Here, I prepared another test tone signal of 300 ms width consists of multiple-frequency peaks in-single-time-line, just at the center of the time width;
WS003134.JPG


Now you may easily guess what I am intending to do with this signal. The recorded air sound of this signal with my multichannel multi-driver multi-amplifier system should show the same "just on the center-line" frequency peaks, if I would have the perfect time alignment, I mean if there is no relative delay throughout all of the SP units.

This method for time alignment measurement and adjustment, therefore, can be called "energy peak matching method", I believe.

I carefully selected the peak frequencies and their gains for current purposes as shown in this diagram;
WS003132.JPG


Above diagram also shows my validation procedures and results. I prepared and applied "tentative" EKIO cross over configuration as shown, and the line-level output signal from DAC8PRO was analyzed by Adobe Audition 3.01 without delay setting and with intentionally forced to delay setting, as you may easily understand in the diagram.

For this validation procedure, all of the divided crossover channels (in monaural) were fed into DAC8PRO's channel-1 for recording of the analog line-level signal with audio interface TASCAM US-1x2HR using the second PC.

As we can see, the "intentionally forced to delay setting" was properly reflected in the recorded sound (here in line level signal) which can be visually identified and the relative delays can be measured with "a pencil" and "a straight line ruler".

Then, I finally applied this semi-quantitative visual method, i.e. "energy peak matching method", in my audio setup;
WS003133.JPG


I do hope the results in the above diagram are just self-explanatory for your easy understandings.

Just like I found in my previous post #493 "Part-1: Precision pulse wave matching method", the 16 ms delay in sub-woofer sound was visually and semi-quantitatively detected, and it was successfully adjusted by EKIO's 16 ms group delay setting for ST+TW+SQ+WO.

One of the nice features of this method is that I can easily draw the vertical time-center-line having the "very tiny and compact 20 kHz signal spot" as the "marker" for the center-line.

It is also important and indispensable adjusting/controlling the color scale of the spectrum for easy assessment of the energy center in the air recorded low Fq signal bands. This "color scale control" corresponds to the elimination of low Fq background noises, and also to the minimization/elimination of the "sound tailing effect" due to the aftershock and/or after-Helmholtz-resonance in woofer and sub-woofer.

Edit:
You would please jump to my post #504 which is sharing "Precision measurement and adjustment of time alignment for speaker (SP) units
Part-3: Precision single sine wave matching method in 0.1 msec accuracy".
 
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dualazmak

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Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000

Hello friends,

Starting on November 11, 2021, I have been sharing my thoughts on possibility and feasibility of adding a pair of woofer (or a pair of completed SP as woofer) in present multi-channel multi-driver (multi-way) multi-amplifier stereo setup; the related posts are;

- Just thinking about possibility and feasibility of adding a pair of woofer (or a pair of completed SP as woofer) in present multi-channel multi-driver (multi-way) multi-amplifier stereo setup:
#451, #167(remote thread), #170 (remote thread), #183(remote thread), #455, #459-#461, #462-#466, #469, #31(remote thread), #470, #471, #473, #474, #476, #477

And, in my post #470 and thereafter, I shared my great interests on SEAS XM001-04 L26ROY woofer (sub-woofer) unit which looks having nice response in 50 Hz - 500 Hz.

Several of my audio enthu domestic friends (in Japan) and ASR Forum friends, however, kindly pointed that I should carefully look at the specification sheet of L26ROY; the overall sensitivity of 87.0 dB (2.8V, 1 m) is not so impressive, the moving mass of 118 gram is relatively large/heavy among the modern 25 - 30 cm woofers, and the linear coil travel length 28 mm and its maximum of 56 mm are relatively long (long throat). Of course I know and understand that L26ROY would effectively compensate these drawback features with its strong 1.2T magnet as well as proper design of sealed cabinet providing enough air-damping, together enabling the excellent transient characteristics.

On the other hand, after I could establish almost perfect time alignment between my present woofer (WO) (Yamaha 30 cm JA-3058 in NS-1000's sealed cabinet) and my active sub-woofer (SW) Yamaha YST-SW1000, I can subjectively feel slightly improved transient characteristics with my WO, SW and WO+SW sound, as shared in my recent post #493.

Furthermore, as I repeatedly mentioned on this project thread, the WO is now directly and dedicatedly driven (no LC-network) by a powerful nice amplifier Yamaha A-S3000 (please refer to my posts #367-#309 for my intensive tests and evaluation on A-S3000), dynamic power 120W+120W/8 Ohm, max 130W+130W/8 Ohm (1kHz, 10 %THD), damping factor > 250.

I assume, therefore, before actually moving forward towards the new woofer implementation, it should be better for me knowing the details of transient characteristics of my current directly-driven Yamaha WO for 45 Hz - 500 Hz, and the SW for 15 Hz - 55 Hz.

For this purpose, I use Sony Super Audio Check CD's track-14 "Speaker Check by Tone Bust Waves". The measurement microphone BEHRINGER ECM8000 was place at 30 cm from the surface of the SP unit, and the sound was recorded by second PC with Adobe Audition 3.01 using TASCAM US-1x2HR audio interface through USB 2.0 connection.

Since the rectangular tone burst signal consists of 8 waves may give too much excitation to the SP unit, I also prepared 3-wave excitation signals by editing the track-14 using Adobe Audition 3.01.

Just for this measurement, I tentatively treated/modified my listening room environment minimizing the sound reflections, resonances, and hence the standing waves, relatively inaudible (by ECM8000 microphone) level; I will soon touch on this issue in my separate post.

The results of my intensive measurements of transient characteristics are as follows;
WS003137.JPG

As shown above, with the strong excitation by 63 Hz tone burst, I can see the kick-up of the first wave even though 5 dB lower than the second, but in and after the second wave, the WO shows really nice response. Only with 63 Hz excitation, however, WO has one aftershock wave, both with 8-wave and 3-wave excitations, while the "fade-out" pattern thereafter looks nice and short.

With 125 Hz excitations, WO shows fairly nice kick-up and fade-out patterns; I am rather impressed by the fade-out patterns since I had thought the WO has more prolonged fade-out patterns. I believe that the direct drive by powerful amplifier with no LC network, together with the sealed and well damped cabinet design, greatly contributes to this more-than-expected nice transient characteristics with 125 Hz excitation.

Next, 250 Hz and 500 Hz excitation, still the important Fq region the WO covers in my setup;
WS003138.JPG


The kick-up patterns are further better or almost perfect by both of the 8-wave and 3-wave excitation. The fade-out patterns after the strong exact 8-wave response are again really excellent.

Then, with 1 kHz excitation, (even though I set EKIO's crossover between WO and midrange Be-squawker at 500 Hz);
WS003139.JPG


I was really surprised that the kick-up and fade-out patterns are still really nice in 1 kHz, with both of the 8-wave and 3-wave excitation.

I am very happy seeing and confirming that my 40-year old Yamaha JA-8058 30 cm woofer (WO) driven by A-S3000 amplifier is still working very fine and doing really nice job throughout the 55 Hz to 550 Hz Fq region in which I always use it all the way through my multichannel audio project.

The results for sub-woofer (SW) YST-SW1000 excited with 31.5 Hz and 63 Hz tone bursts can be summarized in this diagram;
WS003140.JPG


With the 31.5 Hz excitation, SW gives rapidly declining aftershock three waves, while the entire response is in symmetrical shape with kick-up and fade-out patterns. This pattern/shape in whole is somewhat better than my expectation; I had thought it might be rather asymmetrical with possible slower kick-up than fade-out.

With the 63 Hz excitation, I can see almost no aftershock inertia movement of the 30 cm cone of the SW, but slow low-gain low-frequency Helmholtz resonance noise (slow after-movement of inner air) can be seen after the expected 63 Hz response waves. This observation can be better illustrated in this diagram including the Fq distribution spectrum;
WS003153.JPG


These results on SW transient characteristics suggest that it should be better to use the SW in less than about 50 Hz Fq region, and 50 Hz - 500 Hz region would better to be covered by WO. This finding also validates and encourages feasibility of my present EKIO crossover Fq points; high-cut (low-pass) at 50 Hz for SW, low-cut (high-pass) at 45 Hz for WO.


Now, having these better-than-expected measurement results on transient characteristics of WO Yamaha JA-3058 in sealed cabinet and active SW Yamaha YST-SW1000 up to around 50 Hz, I well realized that I should be very careful in selecting possible new woofer unit and designing (sealed?) cabinet for it, in case if I would go into that direction; the overall transient characteristics of such possible new woofer would need to surpass my present woofer, JA-3058, driven directly and dedicatedly by a powerful amplifier.
 
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JRS

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Those tone bursts in your last post are interesting. I am a bit confused, what is the Fs and fb for the Yamaha woofers? And is active EQ being applied?
 
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dualazmak

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Hello JRS,

>I am a bit confused, what is the Fs and fb for the Yamaha woofers?

Yamaha NS-1000;
https://audio-heritage.jp/YAMAHA/speaker/ns-1000.html
WS003176.JPG


Please note that the above official specification is for whole NS-1000 in the sealed cabinet and using its own LC-network and attenuators (for Be-squawker and Be-tweeter). I have completely eliminated the LC-network and the attenuators, and the 30 cm JA-3058 woofer is now driven directly and dedicatedly by powerful A-S3000 amplifier.


Yamaha sub-woofer YST-SW1000;
https://audio-heritage.jp/YAMAHA/speaker/yst-sw1000.html
WS003177.JPG


>And is active EQ being applied?


Yes and no, in my post #495 on transient measurements, I applied only my standard simple XO but no further EQ using digital crossover software EKIO;
WS003179.JPG


The master volume and relative gain controls are shown in this diagram;
WS003180.JPG


For the signal given to the sub-woofer YST-SW1000 (SW), I have high-cut (low-pass) at 50 Hz -12 dB (LR filter) in EKIO plus high-cut (low-pass) -24 dB/Oct at 55 Hz in YST-SW1000.

For the signal given to the woofer (WO) JA-3058, I have low-cut (high-pass) at 45 Hz -12 dB/Oct (LR filter) in EKIO.

As I shared in above post #495, therefore, SW and WO can sing together at 63 Hz in which the transient characteristics for both WO and SW were measured; of course, for transient measurement of SW at 63 Hz, I gave rather high gain tone burst of 63 Hz.


WS003182.JPG
 
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dualazmak

dualazmak

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Identification of sound reflecting plane/wall by strong excitation of SP unit and room acoustics

Hello friends,

In my previous post #495, I applied rectangular tone burst signals in various frequencies for excitation of SP units for their transient characteristics measurement. Such a tone burst signal effectively excite SP unit, and if the gain/volume is large enough, it would also effectively excite room acoustics by a bunch of the sound given by the SP driver.

The rather intense sharp transient excitation of room acoustics, when measured by a microphone, can be used for identification of specific sound reflecting plane (or wall, portion of wall) in our listening room.

In this post, I share with you one example of my "Identification of sound reflecting plane/wall by strong excitation of SP unit and room acoustics ".

Here in this diagram, I set measurement microphone BEHRINGER ECM8000 at 40 cm from the surface of my Yamaha 30 cm woofer JA-3058, then the SP was excited by somewhat unusually high-gain rectangular 8-wave 500 kHz signal;
WS003158.JPG


The measurement microphone ECM8000 is fast and sensitive enough, and we can clearly "see" the primary and secondary "reflected" sound waves coming into the microphone, arriving 23 ms and 52 ms after the SP excitation, respectively.

As I shared in detail in my post #311, the "physical" views of the listening room environment are like this;
WS003159.JPG


You can see the "suspicious" sound reflecting plane/wall at 4.2 m from the surface of my left speakers which I was measuring by a microphone. I remember that I wrote in my post #311;
> .... behind the listening sofa, you can see another Japanese style tatami-mattress-floor room which is acoustically "dead" and effectively minimizes the sound reflections and resonances. Of course, the sliding doors between the rooms are kept open during the listening sessions as shown in this photo. I often place sound absorption sponge mattress in front of the white wall and the glass door leading to the corridor.

The simplified physical alignment and dimensions of the measurement setup can be illustrated in this diagram;
WS003160.JPG


As shown in above diagram, I calculated that the sound traveling distances for the possible "primary reflection" and "secondary reflection" would be 8.0 m and 17.8 m, respectively. The sound velocity at that time, in the room of 22 degree-C temperature, was 345 m/s (meter/second), and therefore the primary and secondary reflections should be "heard" by the microphone 23.2 ms and 51.6 ms after the transient excitation of the SP unit, if "that" plane/wall at 4.2 m ahead the SP was actually reflecting the sound.

The nice match of these calculated (23.2 ms, 51.6 ms) and measured (23 ms, 52 ms) values clearly indicate that "the portion of the wall" at 4.2 m ahead the surface of the SP actually reflects the sound and hence causes a low-gain 500 Hz standing wave in the listening room unless otherwise some deadening treatment would be placed on the wall.

I believe, if needed, we may apply this measurement method in identifying a specific sound reflecting plane/wall by applying rather strong tone burst excitation of specific frequency into the SP and room acoustics.

In any way, since I did not like to see such sound reflection(s) or standing waves in my "transient characteristics measurement of SPs" shared in my post #495, I just tentatively placed rather heavy-duty sound insulation/deadening treatments on "that wall", as shown here;
WS003161.JPG


I actually heard and found such a heavy-duty treatments gave too much sound deadening in my music listening enjoyments in the room; I well know that our listening environment should not be quasi-anechoic, and we do need suitable preferable enjoyable sound reflections resonances and/or standing waves, but here I will not (should not) be going into the detailed discussion on "how much" and "how intensively" sound insulation/deadening should be applied in our listening room, an eternal and endless theme in home audio setup.

Let me just add one point that I do not always fully ignore such room acoustic issues; I sometimes and occasionally "measure" total room acoustics by REW's wavelet analysis, as I briefly shared in my very earlier posts #020-#022;
WS003162.JPG
 
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dualazmak

dualazmak

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Sure M8, stay safe yourself and good luck with your project!

Hello @Qmuse,

Although quite belated, I thank you so much again for the nice discussion we had in February 2020 on this thread regarding "phase and delay".

Only in case if you are still interested in my project, your comments and/or suggestions on my recent several posts will be highly appreciated;

- 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
- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495, #497
- Identification of sound reflecting plane/wall by strong excitation of SP unit and room acoustics: #498
 
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