MOVING FROM "HERTZ" or "Hz" TO "YA"

HERE IS THE DEFINITION TAKEN FROM WIKIPEDIA:

From Wikipedia :

Definition

The hertz is equivalent to one cycle per second. The International Committee for Weights and Measures defined the second as "the duration of 9192631770 periods of the radiation

corresponding to the transition between the two hyperfine levels of the ground state of the caesium-133 atom"[4][5] and then adds: "It follows that the hyperfine splitting in the ground state of the caesium 133 atom is exactly 9192631770 hertz, ν(hfs Cs) = 9192631770 Hz." The dimension of the unit hertz is 1/time (1/T). Expressed in base SI units, the unit is 1/second (1/s).

In English, "hertz" is also used as the plural form.[6] As an SI unit, Hz can be prefixed; commonly used multiples are kHz (kilohertz, 103 Hz), MHz (megahertz, 106 Hz), GHz (gigahertz, 109 Hz) and THz (terahertz, 1012 Hz). One hertz simply means "one cycle per second" (typically that which is being counted is a complete cycle); 100 Hz means "one hundred cycles per second", and so on. The unit may be applied to any periodic event—for example, a clock might be said to tick at 1 Hz, or a human heart might be said to beat at 1.2 Hz.

The occurrence rate of aperiodic or stochastic events is expressed in reciprocal second or inverse second (1/s or s−1) in general or, in the specific case of radioactivity, in becquerels.[7] Whereas 1 Hz is one cycle (or periodic event) per second, 1 Bq is one aperiodic radionuclide event per second.

Even though frequency, angular velocity, angular frequency and radioactivity all have the dimension 1/T, of these only frequency is expressed in hertz.[8] Thus a disc rotating at 60 revolutions per minute (rpm) is said to have an angular velocity of 2π rad/s and a frequency of rotation of 1 Hz. The correspondence between a frequency f with the unit hertz and an angular velocity ω with the unit radians per second is

ω = 2 π f {\displaystyle \omega =2\pi f\,} $\omega =2\pi f\,$ and f = ω 2 π {\displaystyle f={\frac {\omega }{2\pi }}\,} $f={\frac {\omega }{2\pi }}\,$ .

The hertz is named after Heinrich Hertz. As with every SI unit named for a person, its symbol starts with an upper case letter (Hz), but when written in full it follows the rules for capitalisation of a common noun; i.e., "hertz" becomes capitalised at the beginning of a sentence and in titles, but is otherwise in lower case.

End of Wikipedia:

__________________________________________________________

OK... > I DON'T CARE <

I've been thinking about this for a little while now, and I would like to present to you the >NEW< numbering system.

It is called "YA". I know it sounds >STRANGE< but hare are the reasons why I have opted to go with "YA" instead of others...

The numbering system is based on the NEW system (Aurora). And as such, the >seconds< it >WAS< based on, is >DIFFERENT< now... It's > L O N G E R < then it was before
The "Y" in "YA" is divided by >3<. >Y<, 1 2 3. Do you see it? Like our >EARS<. We have the hearing over >3< decades
The "A" in "YA" represents the beginning. Like the "A", the beginning of the alphabet
For those who haven't figure it out yet: YA is my initials: Yvan Arshoun, just in case...
And because it sounds >STRANGE<... Which is cool, I think...
AND finally, no I don't want my name here (BUT It is so perfect with all the reasons listed that I figured why not!)

Hertz vs. YA

>Hz< are >ELIMINATED!!!< (well almost)

>Hz< ARE REPLACED WITH >YA<.

Why do you asked? YYYAAA... It's POSITIVE! That's why... AND the new numbering system is > L O N G E R < than what we are used to. That's the >WHOLE< reason behind >NEEDING< the new system...

OK, here is the >NEW< and >SLOWER< numbering system...

SUBMULTIPLE (hardly used, but I'm pasting them anyways [taken by Wikipedia, so if there is a error, it's not my fault, I have a Secondary 5 diploma, remember?]) :

10^-1 YA, dYA, deciYA

10^-2 YA, cYA, centiYA

10^-3 YA, mYA, milliYA

10^-6 YA, µYA, microYA

10^-9 YA, nYA, nanoYA

10^-12 YA, pYA, picoYA

10^-15 YA, fYA, femtoYA

10^-18 YA, aYA, attoYA

10^-21 YA, zYA, zeptoYA

10^-24 YA, yYA, yoctoYA

MULTIPLES (commonly used, commonly the "Kilo" to "Tera" [for now])

10^1 YA, daYA, decaYA

10^2 YA, hYA, hectoYA

10^3 YA, kYA, kiloYA

10^6 YA, MYA, megaYA

10^9 YA, GYA, gigaYA

10^12 YA, TYA, teraYA

10^15 YA, PYA, petaYA

10^18 YA, EYA, exaYA

10^21 YA, ZYA, zettaYA

10^24 YA, YYA, yottaYA

More to come, including the >EARs<. You'll be surprised at how WELL the ears react to the new system...

Stay tuned!

> LET ME PRESENT TO YOU, MY "AURORA 93" ANALOG SYNTHESIZER <

MORE DETAILS TO COME SHORTLY

A CLOSEUP VIEW: https://img1.wsimg.com/isteam/ip/feda7228-79a8-4009-9b06-c11fc8dd850b/AURORA%2093%20NO%20FRAME%2075%20COMPRSSION.jpg

In the mean time, for those of you wanting to >BUILD< you own circuits (Under the name "AURORA 93") here is the BEST PERSON (I think anyways) for a no nonsense approach. His name is >FORREST MIMS<...

Here is Forrest Mims, the worlds BEST electronics scientist. He is the author of "Getting Started in Electronics", "Circuit Scrapbook", "Engineer's Mini-Notebook", "Engineer's Notebook" and others (I have a few of his books from Radio Shack). Here is his web page: https://www.forrestmims.com/

SOUND

I wonder if mp3 formats is truly the best compression... Because the higher the frequency of the details you hear. SO: If you have a fixed frequency (48,000Hz as an example), the maximum details would be less in the higher frequency...

I think it would be BETTER if we have an INDEPENDENT range, and REDUCE the "details" in LOWER frequency. We are going the OPPOSITE way that our ears work!!!

EVEN if you "fix it", all the professional and the audiophile equipment, YOU STILL have a >>> DIGITAL <<< maximum frequency... You are wasting bandwidth on low frequency. I say that we should GO THROUGH A REBUILDING OF OUR AUDIO SYSTEMS.

Now I think that the curve for the ear is different then a straight line. I think it would be a bit odd. But I feel that we should "reprogram" everything audio at the best "EAR" response.

I know there's a (or few) program that do that... They are not the ideal curve, but better than FIXED frequency. MP3 Players do that(almost).

When you go in the settings menu, you can give the player the choice between a FIXED frequency, or a VARIABLE frequency. I really encourage you to select "VARIABLE frequency". That way they will BE at a VARIABLE frequency.

But for the source of almost all the recordings, it will be guaranteed that there was a 44100Hz (or 48 000Hz or more) FIXED frequency. WE NEED TO REMOVE the >FIXED< frequency, PERIOD. Computers have become SO FAST (for now... AND they are going to be SOOOOOO much faster in the extremely near future that we will all >>> FREAK OUT <<<), that it's not a problem any more.

The >RESONANCE< in >DIGITAL< systems is audible. Yes it true. We (kinda) "ignore" it, but its true. The reason why I say that is: Ever listen to the same audio clip but at 2 >DIFFERENT< sample rates? Try it, you see what I mean. That is because the >SAMPLE RATE< determines the >MAXIMUM HI FREQUENCY< we can hear. So the >MAXIMUM< sample rate of 44 100Hz is a different output than a 48 000Hz sample rate. You can hear it well on a good sound system. You can tell especially on hi hi hi frequencies. That's the pure DIGITAL sound effect.

And they are trying to move away from the digtal "effect" by "pushing" the sample rate >AWAY< from anything audible. They are playing cat and mouse with the sample rate. It's going to cost more and more money to get it higher and higher all the time... Like I said go >ANALOG< for the audio. You still could have digital systems for memory and such, but keep it analog for the audio, that's all...

So I say: go analog. In going purely analog there's >NO< sample rate at all (as long as the audio doesn't pass thru a digital system).

For studios (for now), all our "44 100Hz" or more would need to be REDONE (ya I know, it's completely insane). However we could still keep the old format like that. But REDO EVERYTHING in the near future.

New sound card, new CD/DVD format, etc. etc. etc... All with a VERY high bandwidth. The GREAT MAJORITY of sound cards don't do that. I'm trying to think of a model that does that but nothing come to mind...

See, what is really cool about having a VARIABLE frequency, if that it >FOLLOWS< she sound that comes in. Right now it >DOES NOT FOLLOW THE SOUND!!!<. Even if you >think< it does, it >DOES NOT!< because it is >SET AT SET AT A MAXIMUM RESOLUTION< decided by the sound card... PERIOD.

For those who don't believe me, open your PC and check the sound card. Does it say if it has a maximum frequency? Yes it does, of course. It's the >sample rate< of 44100, 48000, 96000, Etc. That's why I say it should be >REMOVED< and >REPLACED< with a >FOLLOW THE EARS< instead.

That way a 20hz vs a 20,000hz will have the SAME AMOUNT OF DETAILS. No matter what we give to it. Right now it goes against ALL COMMON SENSE!

20hz GETS THE LLLOOONNNGGGEEESSSTTT wave, and the 20,000hz has a SUPER >>>TINY<<< WAVE. WE ARE GOING AT IT THE COMPLETE >>>OPPOSITE<<< WAY OUR BRAIN WORKS!

I know it sounds completely crazy, BUT IN THE IMMEDIATE FUTURE WE NEED TO ELIMINATE ALL "FIXED" FREQUENCY... PERIOD!!! Like that you'd have EVERYTHING follow your >>> E A R S <<<

THEN WE WILL BE ABLE TO HAVE A PLEASANT RECORDING AND LISTENING.

https://en.wikipedia.org/wiki/Hearing_range

SPEAKERS, SPEAKERS, SPEAKERS !!!

SPEAKERS:

When you see a speaker with "spectacular" power rating... Let me tell you the truth about speakers: THE MAXIMUM SPL (Sound Pressure Level) you can get, is exactly what the Xmax (MAXIMUM EXCURSION in Millimetres) says (for Bass that is), mixed with maximum power (in certain cases).

YOU CANNOT GET MORE >LINEAR< BASS THAN THE MAXIMUM SPL says on it's data sheet (mixed with the cabinet gain/loss. You can get more bass if you go over Xmas, but you are losing linearity [pretty bad IMO]). Now there are a large number of things that come into play, A LOT, BUT if you have a bass driver, the max bass you can get is that. So a *5000 Watt* speaker (ESPECIALLY if they DON'T specify the Xmax) is COMPLETE NONSENSE. Look for the Xmax.

What the speaker builders don't say (for the most part, BUT certain companies do say it) is the COMPRESSION of their speakers. That's why a speaker with LOW COMPRESSION and a WELL DESIGNED BOX will sound amazing (with all parameters optimized).

After a certain volume, the compression starts to rise. To the point where you would need TWICE the power the get the same output. YES IT'S TRUE! If you're trying the push the limits you're better off with you money spent on good LOW COMPRESSION boxes. You have to look at everything when buying speaker.

Now that will not be a problem for home use, BUT can be a problem in PA gear (for concerts). This applies to buying a cheap speaker box to "cut corners". It is a REALLY bad idea (if you are playing for many people). In fact, you can CUT the price of amps by 1/2! As a matter of fact, speakers are generality THE MOST IMPORTANT piece of gear you can buy! PERIOD. If you have GOOD speakers, then you can listen to every piece of equipment and decide if its good or bad for you.

Also, it would be a good time to say that now would a good time to move to CONCENTRIC speakers. They have NO PHASE DISTORTION when it comes to mid/hi. I know the are other things to consider, but for home/studio use I think now would a great time to shift our focus on CONCENTRIC loudspeakers.

YOU CAN USE >HEADPHONES< TO BE LIKE A PAIR OF SPEAKERS

Four those who didn't know, sound engineers have their headphones put aside because "speakers" are different.

Well I have a solution actually! AND it may let your "expensive" pair of loudspeakers be, ummm, not necessary anymore. The problem is VERY simple, and we already have this system in plenty of hardware. OK here it is:

There is no DELAY between the Left ear and Right ear. The ears are COMPLETELY separate in headphones. WE NEED TO allow the ears to SHARE the sound when they play. Here is what I think:

Until engineers start putting this piece of gear on the headphone outputs, we can start building boxes, and very fast. You have a tiny box built. This box contain an analog circuit (or digital if that's what you want). It functions like we were in the front of our speakers, but your not. You can listen to your mix as loud as you want, but you bother absolutely nobody. Here is what I've come up with:

You sit down and put you headphones on
A >set< amount of DELAY in the headphones between the >Left< and the >Right< ear
The DELAY is shaped like your skin (face) affects it, and CUTS certain frequencies
IF you want, you can put a "silent" bass driver. You'll need a clip (someone must know where to get them)... Here it is: https://www.youtube.com/watch?v=AiyBJ3_l33o . Now I have 2 of the "Aura"s placed in my basement ceiling just in front of my TV (on the MAIN floor) for when I watch concerts. My floor is 1.5 inches thick, with the beams across the floor. AND I TELL YOU, when the show starts, and the kick drum hits, there nothing that I can say to make you understand how it feels. It quite unreal. The floor shakes A LOT, you feel like you were at the concert, and much more...
You SHOULD press a button to make the sound go back to "normal" headphones. And you press, again, to put it in "speaker"mode. This includes the delay to the ears, PLUS the room which you hear the "bleeding" through the headphones (TWO microphones "parked" over our shoulders. So we hear very very close to the real deal!). You switch between the 2 so you can mix with the Headphones and with "Speakers" (fake speakers)
Maybe have a minute amount of reverb (to imitate you room...). It would have the 3 dimensions of the room, the clarity of the room, all entered in the units

So that's it really. You could buy the box anywhere between I'd say 25$ and about 100$. It's a relatively simple design, it already exist, except that I found no body that has figured out how to make headphone work like speakers... A small amount of tweaking and your set!

CONCERTS:

No more "DELAY" in the sound (concerts will be "instant audio" no matter how far you are), no more "SPEAKERS" yes yes yes, you heard me right, no more bugging people cause you like you music LOUD, The sound is bad, etc. etc. etc...

WELL, IT'S OVER!!!

Ah yes! check this out! Now that we understand how to make the headphones work like speakers (coming soon, oh yes!), we can make headphones act like the speakers at concerts, BUT all in teensy weensee speakers.

You have a antenna (small) placed in the middle of the room (probably up in the ceiling). The antenna shoots the concert sound to the listeners Ipad, or any other mobile device. They tune into a channel on the unit.

Now it would have to be a channel that is COMPLETELY different from any band in use now. it would be a GENERAL WAVESTATION (Call it the "GW" if you will). It does NOT require a username, its FREE, it has NO DELAY, it's ON ALL THE TIME. It's for CONCERTS and all kinds of events. BUT eventually it would make it's way EVERYWHERE. It's simple (no risk of hacking) It's a "General Wavestation"!

MANUFACTURER OF THIS "NEW" TECHNOLOGY:

It will be possible to have SPECIFIC headphones that would be for CONCERTS (or you're everyday headphones actually. Lets call them "CONCERT READY" headphones).

An APP you get on the mobile device, will determine the "CORRECT" frequency response of your headsets, the way the band WANTED you to hear them. It will include a "FM BASS BOOSTER" to give you the bass I talk about below. A "Decibel Setting" that you "can" accept to play (or not, it depends on how you like your music).

The way it works is: If you bought CONCERT READY headsets, these are very very specific headsets. It has a flat (almost) frequency response, GREAT bass, non resonating frame, a set of "2" microphones that lay on the shoulders (or if the manufacturers choose not the have them it's OK too). This is CONCERT READY sound. You can still go at the show with regular headsets, BUT you will not hear the sound the way it was intended to. My guess would be 100$ and up for a pair.

You could buy a earphone measurement system but it would be too expensive (unless if a company decides to offer something at a good price [?]). Better go with a good pair of "CONCERT READY" headsets.

HOW THE 2 1/2 DIMENSION WORKS:

The way it works is like FM, but at frequency NOT used. (Which make me think: WHY did they remove certain frequency from the public? The decided to go "digital" but almost immediately restricted the use of certain frequencies [and kept it silent?]. What are they doing with those frequencies? Could it be a plan for to "trick"us? If you will, you can check it out. Do a little math and determine if their frequencies would be a "magic trick", or am I imagining this? Check my "Laser TV" page, you'll see what I mean. Well it's everywhere on my sit. AND check out the medias, I am registered to 4 news channels, and they have a different opinion on many things. It's hard to tell who is telling the truth).

OK sorry, I got distracted: "The way it works is like FM but at frequency NOT used". You choose a frequency not used at all. I'm guessing a high enough the require SMALL components on the board, BUT not high enough to cause harm. I think we ought to leave a hole (or holes) in the frequency(ies) affecting the human body. The human body functions at certain frequencies (size of Plurons, speed of the Plurons and Unitons, the size of the human body, etc.). If we can avoid coming it "contact" with the Human body, then we are free to use it (within reason). Keep in mind that the "Sine Wave" (FIXED frequency or verify if the VARIABLE frequency does not come in contact with the human body) is the best wave to have.

So now we have our "GW" wave, ready for all kinds of events!

Notice how the site has SHIFTED the things we perceive, to a higher frequency. so we can hear and see thighs MODULATED on a higher level!

This is an other form of the TRIANGLE, In the sense that we want NORMAL wave, BUT we are "Frequency Modulating" 2 "higher frequency" waves, to get "1 wave" we can perceive. Here is a explanation:

"TWO" ultrasonic frequencies "CREATE" "ONE " waveform (CANNOT be measured with conventional equipment), It's a TRICK!

So you have a TRIANGLE!: "2" ULTRASONIC waves generates = "1" ORDINARY wave, but it is "unreal".

That's the TRIANGLE!

NEW "POINT SOURCE" SPEAKERS (FOR THOSE NOT READY TO USE HEADPHONES YET)

In the 1st drawing, you can see the panel of the drivers pushed inwards (the 1st lines). The remaining is just pieces of the carbon fiber.

On the bottom pictures of the U2 tour, you can see the speakers don't have a carbon fiber >recess< . It is flush. The speakers therfore go to the maximum excursion... Whereas mine instead of pushing the excusion, you have a natural bass boost that you put down on the main equaliser. The speakers therefore don't work as hard. You get more natural sounding mids.

For those not ready to make the move to headphones, this is the concert speaker for you. So in a venue you could have the regular speaker >AND< have the new "Concert Ready™" system for those ready to make the switch.

You don't have to worry about the noise coming trough because you can "Frip The Svitch™" (not too sure about that one) and no sound will come and bother you! It's as simple as that!

And the drivers are surface mount. A lot of loss. MY system has the drivers >pushed in< about 2-3 feet.

EXPLANATION OF COMPRESSION IN (LIVE SOUND) LOUDSPEAKERS

>HI< "COMPRESSION" DRIVERS

Most speakers are made with (low cost) >ROUND WIRE< voice coils. Compression is >HIGH<.

When the coils start heating up, the heat is >STUCK< in between the coils. So the heat stays in the coil. Therefore it can take up to >TWICE< the power of an amp to drive them.

Compression is typically about 5-6 db at full power.

So you typically are carrying >TWICE< the power for nothing. Provided the number of Db is the same. Never thought of it eh? It's true... There's SO much to now

>LOW< "COMPRESSION" DRIVERS

Those speakers are made with (hi cost) >RECTANGULAR WIRE< voice coils. So compression is >VERY< low.

It takes >HALF< of the power amp to drive them.

When the coils start heating up, the heat goes >OUT< of the voice coil. So the heat >CAN NOT< stay in the coil (almost). Therefore almost all the power goes to "music".

Compression is typically about 1.5-2 db at full power.

So you see it might be an option (depends on $$). So you need 1/2 the power and save money on the amps!

ALMOST >COMPRESSIONLESS< SPEAKERS

Just a quick note about compression... It affects speakers more or less depending on a variety of "cooling" they have.

>BUT< here is what >I< have to suggest as a quick remedy... It's not the best deal when it comes to greatly minimize compression (a good cooling system is WAY better that none that's for sure, but here is what >I< have as a solution...):

Put in line with your >amplifier<, a simple >EXPANDER< will just eliminate (almost) any compression in the speakers. You have more >BREATHABLE< speakers... Your speakers will not sound as >held down<.

This is not for everybody, the effect can be quite unnoticeable, but for PRO usage, it could be very good...

So for the companies interested in >compression<, a quick graph, the same as the frequency response, will indicate witch frequency and at what ratio the driver compresses.

Woofer/Mid-Range:

1 line at >1< watt
1 line at >2.5< watts
1 line at >5< watts
1 line at >7.5< watts
1 line at >10< watts
1 line at >25< watts
1 line at >50< watts
1 line at >75< watts
1 line at >100< watts
And so on

Tweeter:

1 line at >.1< watt
1 line at >.25< watts
1 line at >.5< watts
1 line at >.75< watts
1 line at >1< watts
1 line at >2.5< watts
1 line at >5.0< watts
1 line at >7.5< watts
1 line at >10< watts
And so on

I believe this will help us build better speakers. The numbers which I have mentioned here are simply those the I find to be OK, but I'm sure that someone can come up with better values... In looking the them, I get the feeling someone will.

>ROUND< TWEETERS (YAY) VS. >RIBBON< TWEETERS (BLAHG)

Why do people buy >RIBBON< tweeters? To me, it me it makes no sense at all...

You go to the hifi store in order to get the best speakers for you home system. You listen to columns >AT A DISTANCE< with >RIBBON< tweeters and a >ROUND tweeter<. And to your knowledge, the >RIBBON< tweeter sounds THE BEST...

So you buy the columns with the >RIBBON<> tweeters, thinking you get the best for the money, because the sounded so good >FROM WERE YOU WERE LISTENING<.

Let me explain to you WHY you buy the speaker with the RIBBON tweeters instead of (good quality) >ROUND< tweeters.

When you listen to music from a distance you (think) you are listening the the "same" frequency range in both speakers, but it is a (pretty big) >ERROR!<.

You see, >RIBBON< tweeters (the tweeters that come in a >SQUARE< or >RECTANGULAR< shape) have a roll-off of >3< decibels
>ROUND< tweeters (the tweeters that come in a >ROUND< shape) have a roll-off of >6< decibels (TWICE the roll-off of the ribbons)
Yes the data sheet will be accurate at >>1 METER<< (what the data sheets >measure< at). BUT as soon as you GO BACK further than 1 meter, the frequency response starts to >SHIFT<. You then see the effect of the >RIBBON< starting to sound >SWEETER< the more you >BACK UP<
The further you go, the sweeter it sounds. THAT'S >MOSTLY< WHY it sounds "BETTER".

That's the >TRICK< behind RIBBONS. They sound much better (3 db roll-off) than ROUND speakers (6 db roll-off). It's a trick (sorta) behind the 2 technologies.

The best way l can explain it, is with the attached pic (see below).

Lets take this as an example: Lets say you listen to your speakers at a distance of >3< meters (contrary to the >1< meter that they measure on their data sheets, except for studios and PC speakers).

3 Meters (9 Feet) is what I would expect for CORRECT listening, not 1 Meter (3 feet).

So now you can see, there is a >BIG< difference in the measurement. You put the speakers at a GOOD DISTANCE (3 Meters) you can see that they sound >TOTALLY< different. And its obvious, because the >RIBBON< tweeter got attenuated >LESS< than the >ROUND< speaker.

And the tweeter starts to "shift" frequency at a specific frequency. It has to do with the dimensions of the tweeter. It's hard to explain without the formulas. But the frequency at witch this happens makes you "THINK" you are listening to a "better" tweeter. It really isn't. The truth is the frequencies that are (naturally [3db]) *BOOSTED*, make it sound >SUPER HIFI<.

You should have a graph to see the difference between >RIBBON< tweeters and the >ROUND< tweeters. You will see on the >RIBBON<, the response goes >UP< the further away you go.

That's why I wonder how come studios have ribbons. I mean lets just for a second try to "calibrate" you room. The high frequencies of your speakers (for someone listening to their song) gives the wrong impression of how it sounds.

You have the perfect sound were you sit (mixer) but the client doesn't get the real sound. He would have to get closer to the monitors to get the right sound.

On top of that when you "calibrate" your room, just look at the graph. The balance it not right. You have tweeters that are very hard to "balance". The room sounds different at a near distance vs far distance (because of the tweeter).

And OK it's fine for mixing, but your clients have all this >HIGH< frequency that you don't have.

Now I know a lot has to be thought of, but generally >RIBBONS< (depending on the frequency) are simple: 3 db, and >ROUND<: 6 db.

And forget the woofers, they will (for now;)) be at 6db because it would be impossible to get a 3db loss (it's just too large for low frequencies). It's in the upper frequencies where you can "play" with their db.

Now, tweeters are fine at 1 meter, but go back 2 meters and 3 meters, and the tweeters start to change. Again you can see what I mean with the graph.

PS, that's why in concerts, you have >PLENTY< of bass if you're close, BUT you >LOOSE< a LOT of the bottom end the further away you go. I explain this on this sheet later on.

Personally I always go with >ROUND< tweeters. >TO ME< they are better...

PS, I've been trying to get this right, but the "errors" make me go nuts. I fix the errors, double and triple check if everything is good, and it comes up with a bunch of "new" errors... Just ignore them, I will try to figure out what to do.

AUDIO MIXERS - THINGS THAT I THINK WOULD BE GOOD

WHAT > I < WOULD LIKE TO SEE IN MIXERS

You could even have the HI and LOW bands have a VARIABLE frequency. That way you could have 3 gains WITH 3 frequency all on 3 pots TOTAL! The frequency and the gain on the same pots. Save space. So: A CENTRE frequency position would be good if it gets too complicated. And you have plenty of equalizer to not worry about a graphic equalizer.

What I think is (quite frankly) unfortunate is mixers that have a WALL WART. Seeing a wall wart on a "PRO" piece of equipment kinda make me quite blah, I can't even come to the right word for it.

A wall wart, I'm sorry to say that but, is CHEAP (particularly on bigger pieces of equipment). If a wall wart is in the design it's because you trying the keep manufacturing costs LOW. I've got my eye on a very very good mixer, but the darn thing has a wall wart. I'm stuck because if you look at the way it is made (and, yes I know it still works no problem), BUT its a tiny tiny piece of metal making the (bad, In my opinion) contact. Its NOT AS GOOD!

Now,I know for a fact that having your products certified in house would cost too much (until a change is happening). Just for the general public to know... When you have a product that gets certified IN HOUSE, it COSTS money. What REDUCES the cost (and time etc. etc. etc...) is just putting a simple WALL WALT in the design.

What does that mean? It means that the manufacturing companies who simply put a wall wart in the design, they are free tho make whatever they want. They simply BUY the wall wart and they are done (because the companies that make the wall wart, for others, have the obligation to make it safe, CSA etc etc... So THEY have the wall wart certified ). So companies can make a product that is not quite as "bullet proof" that it can be.

BUT it comes at a price... The WALL WART... It is awful. Really awful (my thoughts that's all). If all companies would ELIMINATE the wall wart, your house would look way better, I mean, have you really thought for a second, how much of a mess wall warts cause? Can you think for a second how clean it would be if wall warts were gone? PLUS you would NOT have to carry that "thing" around (If you are going some where)... Imagine, no wall wart...

If everybody agrees that they don't like wall warts, (EXCEPT the physical size is the problem) then it would not be an issue. All the products that HAVE the wall wart, would be a different category than those who DIDN'T have the wall wart. Look at the way their made. It NOT the same as a solid IEC connector. It's simple.

SO that's my view on wall warts: HATE EM, totally.

NEXT INVENTION: THE >> NOISE GATE << ON THE CONSOLE (like the compressor on board all ready). YES that's right a >> ONE << KNOB NOISE GATE. I know what you are thinking, Too many parameters to do it, OR it might not sound good). Well I have a (strong) idea that might work (you can tell me if it sounds OK or not):

You have a "noise gate" that SLOWLY cuts off the sound, BUT, it is done by physiologically reducing the threshold. You hear the sound diminish and loose HI frequency as it goes lower. I've heard the effect on computer noise gate, with a excellent results. The fact the the HI frequency goes down, makes it smooth and (almost) perfect. So the Adjustment would be the THRESHOLD, with the "OFF" function if you don't want the effect . That way it is totally smooth...

A last thing before I go (for now)... MEMORY: A 1Tb >>>MINIMUM<<< MEMORY CARDS. Now if we can go much higher, it would be even better. But their getting there...

NEXT FEATURES: A SEMI-PARAMETRIC equalizer on the SMALLEST number of channels. NOT a FIXED equalization. It frustrating the have companies offer that on higher numbers of channels. If I have room for 2 channels, let me have 2 channels with a FULL EQUALIZER. Not a "Fixed Frequency" equalizer.

And (with the cost of metal now) have the FREQUENCY be on the SAME POT as the GAIN button. One on top of the other.

You could even have the HI and LOW bands have a VARIABLE frequency. That way you could have 3 gains WITH 3 frequency all on 3 pots TOTAL! The frequency and the gain on the same pots. Save space. So: A CENTER frequency position would be good if it gets too complicated. And you have plenty of equalizer to not worry about a graphic equalizer.

If the HI frequency is to remain fixed (because of cost), I think that 10Khz in the HI frequency is the "sweet" spot. 12Khz is a bit hi... The true sweet spot (I think) is 10Khz. You hear everything you need to...

The MID: A SWEEP that ranges from100Hz to 8Khz should be standard (in my opinion). If you vary from that, you no longer follow the curve. In that range of the frequencies everything follow the same line. AND everything >BALANCES< out. The frequencies COVERS the WHOLE human range.

Lastly: The LOW frequency SHOULD be at 80Hz. A 100Hz it a bit hi. 80Hz is PERFECT (My opinion)!

So you see how everything just follows >PERFECTLY<the EAR.

10Khz
100Hz - 8Khz
80Hz

I >THINK< THAT SHOULD BE THE STANDARD.

SO that's what we are missing right now. If these elements would be addressed, WE WOULD NOT NEED A COMPUTER! I >>> LOVE <<< mixing without a computer. It's free, and you get to make the best with what you have. And I BET if you sit down and listen to OLD records, it just sounds just amazing. They had no PCs back then. They went at it by ear. Or of course you paid a heck of a lot of money to get a album done, BUT it was done without a PC .

PS, in case you are using a computer with a OLD hard drive (>HD< not SSD), you would not need to upgrade, despite the new computers out there. Before I changed computers (I had no choice because I was waiting a long time for the site and it had a long time to even load). BUT I had a 13 year old computer, all I did was UPGRADE THE HARD DRIVE, and because of that I was going SOOOO FAST, it's was insane. So if I may make a suggestion... IF you computer is 8 or 10 years old, I would simply upgrade the Hard Drive. This applies to the people who JUST like surfing, and do very basic stuff on there PC. Very basic stuff. BUT only go with some one you trust with this. DON'T DO IT ALONE.

Now not all PCs are the same, but basically it's gonna do the job very well. My mother, I gave her my old laptop (6 gig of ram, and a new 250gig SSD) and you can not tell the difference between her and a new PC surfing the web. YES I know, it's crazy. Especially with UBUNTU https://ubuntu.com/download/desktop/thank-you?version=22.04&architecture=amd64 (Ubuntu download, if you're ready)

Oh yeah and, SSD (Solid State Drive) does NOT need to be big to be fast (ancient Hard Drives, they still make them, go figure, there is a awful list of things I could mention but I will save you the 1 page of reasons why it important to upgrade). With the SSD, You could put the smallest drive possible, and it will run as smoothly as a big on (about 3 TIMES MINIMUM the speed of old ones, AND it does NOT slow down, aside from virus etc.). Plus you get twice the battery life of your laptop (approximately), if not more. So it's up to you to take the decision.

My (13 year old laptop) had 4 gigs of ram. That it. AND if you choose to stream down (remove the JUNK that MS puts it the software) you could go SOOOOO fast, it's not even funny. I think MS makes their software be BLOATWARE. I mean you are STUCK putting all the software on your system. YOU pay HUNDREDS OF DOLLARS for a new new OS and you're making >>>THEM<<< install software you DON'T want(?)! WHERE'S THE LOGIC IN THAT???

OK, here is my OLD PC (For those who are interested):

Asus K7V
4 Gigs Ram (Particularly low amount of Ram by today's standard)
1Tb SSD (BE CAREFUL, NOT > HD <): (Prices for a bran new, standard model SSD (Ordered 1 yr ago) but top of the line SSD, so prices may go DOWN if you choose to go with a "inferior" brand: If you surf the web ONLY, you could go with 128Gb at 30$... Other uses: 256Gb at 55$... 500Gb at 90$... 1Tb at 120$... Prices are very approximate! Speak to somebody who can help you please). Prices in Canadian Dollars
3 1/2" floppy drive
DVD/CD drive
Etc...

Round knobs. THIS IS JUST MY PREFERENCE, not a request. I would be happy if consoles would come with ROTARY buttons instead of sliders.

With the amount of people mixing at home (and having less space) and mixing from their PC, it would only make sense if you offer a smaller version (sliders replaced with ROTARY pots only). It would replace the cost of the metal (which is going thug the roof right now). AND it doesn't let the junk come it to the electronics.

So to recap:

You have a SEALED system against the outside elements
You have a SMALLER system to fit at home
You have a LESS EXPENSIVE board
You don't have to go with sliders because you are MIXING in the COMPUTER

PS: A quick summary of what went wrong with a X32 board I purchased:

The board got here, There was 1 fader that was slow. I mean VERY SLOW. And it was sold as a BRAND NEW unit. Now I have a "new" board that is broken!
The firmware did NOT WORK AT ALL when the board arrived. I thought something was wrong. So I looked online and saw somebody at that said he could not even use the board. Well thankfully I installed the new firmware and it worked. BUT seriously I could not even get the sound to do anything. It was awful. The VERY LEAST have a firmware that SENDS THE AUDIO were it should go. So it was 2 of us that had this problem

Also the mixers should be as SIMPLE as can be. I have an X32 mixer, and I tell you, it's is so complicated (just trying to make sens of it is hard).

They should rethink how it's made. Redo the layout. Right now it's is complicated. If I would have tried it before, I would have purchased another. I mean OK it has a good number of features... But here is my suggestion on how to make, well, ALL products work well for ordinary users and professional (this does not apply to all products):

Right when you start for the 1st time, make the person CHOOSE if it is for a general user or if its for professional use. That way you will sell MORE units to the general user. Its simple enough the get a SIMPLE function out of it.
REDO THE LAYOUT. If you look at certain products you'll see that they put the buttons in a very clear and precise method. That's the method that it we should follow.
REDO THE SOFTWARE. If it's hard to understand, then you lose sales. Again: Choose if the product is for general use or if it's for professional use. We have the software to do it now, so lets do it

Amplifier INCLUDED!

You are asking yourself, "Which are the best speakers for the money right now?". I would have a solution right now, but after listening to them at 100% volume (Ya I know, I'm going deaf actually, but that's not the point). The speakers kinda work, but the have a really bizarre "THING" that happens. Its a mix of off the wall microscopic delay the specific range, witch works fine (a bit) then the range goes off (a bit). Plus the tweeter on the left side does a frickadoodle, and goes down about 5-6 db. BUT only when it fells like it. Its not constant. Buy there something wrong with the left speaker.

Do you think there is something wrong with the speaker, or the problem lies with me? The loss I found it on a site too... The exact (minus the frickadoodle part) was exactly the same.

Any way here is the "best" speakers (without my [and the other] "bug"):

Edifier R1280T: just the RCA inputs
Edifier R1280Ts: RCA inputs WITH SUBWOOFER out
Edifier R1280TDB: RCA + Digital + Optical
Edifier R1280TDBs: RCA + Digital + Optical WITH SUBWOOFER out

It's EXACTLY the same speakers EXCEPT the connectors in the back.

Amplifier INCLUDED!

THE BEST!!! Expensive, I agree, but they are the best.

Amplifier INCLUDED

ACOUSTIC EQUALIZERS

AUTOMATIC ACOUSTIC EQUALIZATION

This is what I consider to be THE BEST audio, OR, rack module you can have. It's an AUTOMATIC EQ. Big brands have it. A good AUTOMATIC Equalizer can make the difference between a gig won or a gig lost. IT'S THE MOST IMPORTANT purchase. Even at home (it is not as important, but still, can make a real difference). Check it out!

Some companies make this, unfortunately, it is too complex. For home (especially) it can make the difference between AWESOME speaker and blah speakers (I must change these speakers, their crap!).

I tell you this, and this is no joke: Put an Auto EQ before your speakers, and you tell me if you need new speakers. I don't understand how we are not able to have EXCELLENT speakers nowadays with the Auto EQ available. The difference can be ENORMOUS!!! Plug it in you audio source, you'll see!

We have very very great audio equipment now (really really good audio equipment). So I ask the manufactures to make a version that is >SUPER SIMPLE< for home use. I can also suggest a curve for home use AND for Pro use. I'll see what I can do...

AUTOMATIC PC ROOM CORRECTION (AUTOMATIC EQUALIZER PLUGIN)

To make sure your PC functions properly (without buying an equalizer) here is a SOFTWARE solution: https://www.roomeqwizard.com/installers/REW_linux_5_20_8.sh . Here is the general website if you need more information: https://www.roomeqwizard.com/ . There are 2 videos to help you out on the page below.

For me, the software I use for >RECORDING< (not for the net though!), is 48000Hz/16bits. I find the others too be unnecessary (unless you want the ABSOLUTE BEST sound possible). The truth is that you'll put your files in mp3, which is quite compressed. So you loose much of the HiFi you recorded at). If you're not sure about what to do right now, then just put 48000Hz/16bits. Its generally THE BEST without going crazy on the gear.

It's nuts, if we where to follow everything they say, we'd keep buying new stuff all the time. There are others, yes, BUT, I use 48000Hz/16bits. Save yourself a new PC. And people won't even notice! Just listen to Sade: https://www.youtube.com/watch?v=_WcWHZc8s2I . The recording was done in 1992, and it sounds absolutely AMAZING! AND she had a LOWER sampling rate then what I suggested.

You can make you speakers sound as good as possible, for studio use. You will get the BEST sound possible with this plugin. You may not need to change you speakers after all!

You'll need a measurement microphone now. I have one and it cost me under 40$, and it's rated among the best ones for doing tests. I would not use this mic for recording however. Yes it is flat, BUT its omnidirectional, it has high noise, etc. Is perfect, however, for just putting the mic in the middle of a room and recording a band, for example. I did it myself and putting it in the middle of the room where the sound was just excellent. You would need phantom power though... Here's the mic in question: https://mediadl.musictribe.com/media/PLM/data//images/products/P0118/2000Wx2000H/ECM8000_P0118_Other_XL.png

Others measurement mics WILL WORK too! But this mic is exceptionally flat. More so than many others tested. But regardless, you will still get "corrected" values that are WAY better than without a mic. Guaranteed!

And try this if you're skeptical: Take ANY >CHEAP< SPEAKER, and run the plugin (or automatic equalizer). And you tell me if you can mix a song or not! Adjust (cut) the bass for your speaker not to push itself too hard. Usually you cut the bass at -3db.

When you look at the graph, when the bass starts to fall, you take the -3db point and that where you put the cutoff. I don't know if I explained it well enough, but that's what you need to do. You'll have a picture of the graph once it starts doing it measurements.

Now when it comes the automatic room correction, it's usually done white a >BASIC< equalizer. What I >THINK< would be an even >BETTER< choice, who'd be (like the "AUTOMATIC ACOUSTIC EQUALIZATION" just above) a "linear phase equalizer"

A linear >LINEAR PHASE EQ keeps the >PHASE< (amount of delay) the same. >USUALLY< it >SHIFTS<, so it is not truly linear. >BUT< when it >IS< linear, it sounds >TOTALLY< natural.

Today most EQ are >NOT< a linear phase EQ. What I suggest is coming forth is a >LINEAR PHASE EQ<. Much >SMOOTHER<...

POWER RATINGS

POWER RATINGS:

The power rating on speakers can LITERALLY be 4 time the REAL rating. The REAL amplifier rating is THE PURE SINE WAVE (RMS) rating. Here are the ratings:

CONTINUOUS POWER RATING: The REAL rating
MUSIC RATING (OR PROGRAM): >1/2< the REAL rating
PEAK RATING: >1/4< the REAL rating

Look at the breaker in the back (That's where you get the real numbers), And they cannot lie about this, or else they have real problem if the don't give right numbers. Look in the back: THAT tells you HOW MUCH POWER you're driving.

Lets say the amp is rated at 100 Watts (LOOK AT THE POWER RATING NEXT TO THE FUSE). You can tell how much you amp is driving MINUS the heat transfer. SO (for an example) lets say you're shopping for a 100 Watt amp:

100W x 90% (for an amplifier rated Class D, the one I think is the best in price overall), is 90W for speakers *TOP*
100W x 60% (for an amplifier rated Class AB, the most popular on the market), is 60W for speakers
100W x 50% (for an amplifier rated Class B, BUT not recommended, high distortion), is 50W for speakers
100W x 25% (for an amplifier rated Class A, super low efficiency BUT excellent sound!), is 25W for speakers

So lets put the REAL ratings only! The >>> CONTINUOUS POWER RATING (the 1st rating of the three) <<< needs to be the REAL power rating!

THAT'S IT!

AMPLIFIER CLASS "X" TO... CLASS "D", OR BETTER

Just had an idea that "maybe" could work...

An amplifier going from say, either class "A", "AB", "C", "H", whatever the class, all giving a FUNCTIONAL class "A" sound, BUT ends up being a class "D" in the heat dissipation.

This could be done via a processor that analyses the input TO the output, and adjusts all the parameters to "correct" the errors, and make it follow the class "D" for heat.

I know there are delays in the way this operates, and also perhaps a "ready built" program for already made models of amps, but you could ALMOST eliminate all the distortion, and the lack in bass, etc. etc...

Perhaps it would be good for processor to take PRELIMINARY measurements, then it would "know" what correction to give. Then it would know what to feed the input. It seems like it would be possible to arrange the output if it has the "near perfect" input.

That way changing amps may be over???

Just something to think about...