Understanding how speakers work and are tested - in simple terms

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because of the great risk that you will read some alluring propaganda about open baffle speakers, and be sucked into the Dark Null of Destructive Interference. It's more dangerous than Scientology and ISIS together, and it's right there in your living room.

Smack dab in the middle!

"Open baffle" is as precise term as "loudspeaker" Yes there is a cult of some kind, but real understanding of basic principles seem to be missing from most of "designers". Dipole radiation has some very special characters and well executed multiway dipole loudspeakers sound wonderful, in an appropriate room. It is a "tour de force" way , be warned!
 
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Will any of these books answer the following :

1.How to consider your listening levels when designing- for example SPL of 75 dB/W/m

2.How to obtain a specific stereo image footprint in front of your speakers

3. What standard music test tones and samples should you download and use to test your speakers?

4. How to assemble a the amplifer + speaker system for best results.

5.How to build satellite subwoofer systems.
 
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Well that's the point. Do I want a piano in my living room or bedroom? What about a rock band or symphony?

With television we have minature images of the real thing, you have 14", 21", 32" etc screens. With audio, we sometimes want a 14" version but without any of the pixels missing and in the right color balance.

What size do you want your music is the question.
 
Well, why not a cinema ? There the image is supersized, and you are plunged into the movie.
Same should happen when reproducing music thru the system : it should create a state of plunging ( duck, dive ... )
And as sound is created by the two sources, the speakers, particular attention should be given to the aspect
 
If a loudspeaker is designed/rated to give only moderate spl level (like TVs, table radios and desktop speakers), distortion will terrible. Hi-Fi sound reproduction is not just about frequency range and directivity, also distortion arising from speaker should be minimal. This is what many call "headroom"

Sure it is wise to think about spl levels that will be used. We all don't neet eg. horn speakers and compression drivers. But KEF LS50 or any other 2-way with single 4-6" bass drivers are not a hifi speakers in my world either. And I don't listen loud!
 
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Equal contours

If a loudspeaker is designed/rated to give only moderate spl level (like TVs, table radios and desktop speakers), distortion will terrible. Hi-Fi sound reproduction is not just about frequency range and directivity, also distortion arising from speaker should be minimal. This is what many call "headroom"

Sure it is wise to think about spl levels that will be used. We all don't neet eg. horn speakers and compression drivers. But KEF LS50 or any other 2-way with single 4-6" bass drivers are not a hifi speakers in my world either. And I don't listen loud!

Can I experience Hi-Fi without listening loud? There was that equal loudness countours graphic somewhere let's see:

https://en.wikipedia.org/wiki/Equal-loudness_contour

When a speaker is designed for flat response, I guess it is designed to boost the bass frequencies to follow the curves shown. By the way the curve is another of those difficult to interpret ones. For the novice.

Also A - weighting. Important to keep in mind when desinging speakers?

https://en.wikipedia.org/wiki/A-weighting

Also Equalization of Sound Systems
 
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Back to my investigation into how speakers work. A recommended book will be reach me soon.

Meanwhile, what I think happens:

As mentioned earlier, a speaker diapgpram creates sound by vibrating at a certain frequency. It is made to vibrate at that frequency by the effect of the current inthe voice coil causing the diabpgrm to move in and out.

A little needs to be said about the time intervals in which this occurs. ( not sure if this is in books) . Taking a 1 Khz pure sine wave tone, this amounts to 1000 cycles pers second. The speaker diapgram will there for be changing direction 1000 times a second. The eardrum in response will also be vibrating at this frequency. It is apparent therefore that the occurence of hearing takes place within a few milliseconds. In fact.

It would be interesting to see what the minimum time interval or pulse the ear is capable of detecting. It must be appreciated therefore that human hearing is a real-time, rapid process.

Complicating the hearing process is the fact that the interpretation of sound requires time. Many of us may have had the experience of hearing something and understanding it later. The sound is then stored in memory until it is interpreted. Could this be taking place in the case of listening to music. I suspect that the analysis of music is not real time but takes place from the data stored in short term memory. In fact, the following article seems to bear this out.

The ear and the primitive brain are known collectively as the low-level processing units. They perform the main feature extraction which allows the brain to start analysing the sounds, breaking down the sensory stimulus into pitch, timbre, spatial location, amplitude, reverberant environment, tone durations, and onset times of different notes.

This data is conducted through neurons in the brain; cells specialized in transmitting information, and the basic building blocks of the nervous system. The output of these neurons connects to the high-level processing units located in the frontal lobe of the brain. It is important to note that this process is not linear. The different regions of the brain constantly update each other with new information.

This Is Your Brain On Music: How Our Brains Process Melodies That Pull On Our Heartstrings

Sound is transmitted as a series of compressions and rarefactions of air molecules. Temperature and pressure of the air will affect the transmisstion of sound.

Another less aprreciated effect is the time sound takes to travel. at 330 metres pers second, sound will take 3.3/330 1/100th of a second the travel the distance of 3 metres, for a 1 Khz tone. This is for direct sound. Barring attenunation of the reclective surfaces, reflective sound will take more time than this 10ms time to travel to the hearer. If the path lenght doubles due to reflection, the time delay will double. Of course repeated refelctions will occur, but multiple echoes such as found in an empty room, are not the best for the enjoyment of music. Sound basorbtion panels on the either side of sepakers on a wall, for example have been demonstrated to clean up the sound considerably in a demo found on You Tube.
 
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Thanks for the corrections. Uploading some diagrams

The diagram shows a short pulse of sound. Assuming this is from a 6 inch woofer, the pulse duration is very short, and shows one compression and rarefaction. The illustration highlights the short time scales in question.

Given the millisecond pulses involved, I am almost sure that sound is processed and interpreted by the brain, not in real time from short term memory. This has no bearing on sound quality, except for reflected sound, where the reflected sound and the original sound arriving a few fractions of a second earlier is stored and processed as a whole.

Anyone know any references in the literature?
 

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It takes longer to percieve a lower frequency signal.

Lower frequencies aside, a reflection will combine with the original if it is close enough. With some delay it can become a problem but with enough delay, it becomes an echo which the brain can process with less effort. Reverberation isn't necessarily a bad thing. The rule of thumb is that the sound should pass behind you before coming back as a reflection.
 
You are on the right track with your assumptions, investigation and questions. You will notice that as you delve deeper into the details, the complexity tends to explode.

I will give you another detail to contemplate:

Obviously, music is not a pure sine wave, it's a combination of different frequencies with different duration and amplitude each, constantly changing, all being reproduced by the speaker's diaphragm. (well, in a multi-way loudspeaker, different drivers reproduce a portion of the audible frequency range).

Now, consider the woofer diaphragm reproducing a low frequency tone as well as a higher frequency tone (say, 70Hz and 500Hz). The diaphragm will superimpose both tones and vibrate both at 70Hz and 500Hz. This translates into an oscilating source (oscilating at 70Hz) producing a 500Hz tone, causing the well known doppler effect, where the higher frequency is sounding even higher pitched when the diaphragm is moving towards the listener and lower pitched when the diaphragm is moving away (back). This causes a vibrato like, flutter effect to the sound that can be called inter modulation.
 
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