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Mini Comparison - Vibe (1st gen), C700, PK2, RE0, NE-7M, PFE, ER4S, OK1, TF10, UM3X, SE530, IE8 - Page 7

post #91 of 142
Quote:
Originally Posted by average_joe View Post
OK, so this example:


You state a 50 Hz sine wave. frequency = 1/time. So, 50 = 1/T, solving for T = 2ms. 5ms = 20 Hz. Wouldn't that be basic knowledge for someone that is an engineer/know physics and has been in this hobby for 15 years? Anyways, if a driver is supposed to produce a 20 Hz sine wave (5ms) and the driver somehow (not sure how this somehow works) produces it in 3.5ms, then it produced a 29 Hz wave. What happens with the remaining 1.5ms of source signal? If the driver continues to move, then is the driver an amplifier in itself (how is that possible)?

I can't imagine it, so please take the time to diagram it. Please free hand draw it, take a picture of your drawing with a digital camera or camera phone, and post the pic.
It's been a while since I've studied physics but from that example, then yes if you keep amplitude the same then the frequency has to change, no two ways about it. This would also mean that you would indeed be hearing music different to how you have heard it before (pitch) and if the diaphragm was ready to move on to the next wave instantaneously, would result in a quicker track.

However, mvw2 I understand you can keep the frequency the same even if acceleration and deceleration of the wave is different for two peaks but surely it would mean amplitude would be alot higher or lower as average_joe pointed out and not sure how that translates to 'body' of note as you say. Assuming the peaks and troughs within an individual wave are irrelevant then surely you would experience the same body regardless of the amplitude the diaphragm sent out?
post #92 of 142
Before that we must understand what is sound waves. Why different musical instruments produce different tone/sound characteristics even though at same musical note, example 440Hz? This is because the sound are not made from a simple sine wave. They mixed together and create the sound that we used to hear.

I would like to introduce a tool which is, IMHO, the BEST sound editor I ever use. SND The Snd Home Page . I am using the software to analyze and understand more about sounds. Yes, this software is complicated, but VERY powerful. There are many examples which you can read and understand about sound. My favorite is a birds/animals sound synthesizer program.


Sorry, I have to go now.....

Thank you/
post #93 of 142
Quote:
Originally Posted by communic View Post
It's been a while since I've studied physics but from that example, then yes if you keep amplitude the same then the frequency has to change, no two ways about it. This would also mean that you would indeed be hearing music different to how you have heard it before (pitch) and if the diaphragm was ready to move on to the next wave instantaneously, would result in a quicker track.

However, mvw2 I understand you can keep the frequency the same even if acceleration and deceleration of the wave is different for two peaks but surely it would mean amplitude would be alot higher or lower as average_joe pointed out and not sure how that translates to 'body' of note as you say. Assuming the peaks and troughs within an individual wave are irrelevant then surely you would experience the same body regardless of the amplitude the diaphragm sent out?
OK, #1, I made a mistake in my post (yea, I am human). 5ms is a 200 Hz signal, and if you take the positive part of the wave, it is 2.5ms. So, lets say the positive movement should take place for 5ms, a 10ms total wave, a 100 Hz frequency. If half the cycle was 3.5ms, the wave would be at 143 Hz.

Part of my point was that knowing/remembering/understanding physics is important for this discussion, and to me there have been things that have led me to believe mvw2 is lacking somewhere, not that I am perfect. I guess I proved that there are even differences between understanding and knowing/remembering all the important details.

While on an absolute scale 100 Hz vs 143 Hz does not seem that big, on a logarithmic scale it is a biger difference. Along with how it is produced, it is not a pitch change, it is a change in the base note. As oarnura stated, it seems like new physics is being created, the output is greater than the input without any additional energy. And that violates conservation of energy.

And from my understanding, waves don't have acceleration/deceleration, that is a physical property of sound producing element. The wave has amplitude, wavelength, and shape.

Quote:
Originally Posted by bakhtiar View Post
Before that we must understand what is sound waves. Why different musical instruments produce different tone/sound characteristics even though at same musical note, example 440Hz? This is because the sound are not made from a simple sine wave. They mixed together and create the sound that we used to hear.

I would like to introduce a tool which is, IMHO, the BEST sound editor I ever use. SND The Snd Home Page . I am using the software to analyze and understand more about sounds. Yes, this software is complicated, but VERY powerful. There are many examples which you can read and understand about sound. My favorite is a birds/animals sound synthesizer program.


Sorry, I have to go now.....

Thank you/
I think we are talking about sine waves because they are simple waves, easy to produce and reproduce by drivers. And yes, different musical instruments produce different sound wave shapes with different harmonics.

Thanks for sharing the sound editor, I will check it out when I get a chance.
post #94 of 142
Quote:
Originally Posted by communic View Post
It's been a while since I've studied physics but from that example, then yes if you keep amplitude the same then the frequency has to change, no two ways about it. This would also mean that you would indeed be hearing music different to how you have heard it before (pitch) and if the diaphragm was ready to move on to the next wave instantaneously, would result in a quicker track.

However, mvw2 I understand you can keep the frequency the same even if acceleration and deceleration of the wave is different for two peaks but surely it would mean amplitude would be alot higher or lower as average_joe pointed out and not sure how that translates to 'body' of note as you say. Assuming the peaks and troughs within an individual wave are irrelevant then surely you would experience the same body regardless of the amplitude the diaphragm sent out?
If a driver moves further for a given input power then the amplitude is affected not the frequency. That is why drivers have different sensitivity.

mvw2's example is broken as Average_joe pointed out. If a driver is reproducing a different frequency than it has been sent, the driver is broken.

I am not aware of any physics that can make a body accelerate faster than the force that was applied to move it. Even in a perfect world with no friction we would get the exact amount of energy transfered. If a driver has stored energy from a previous wave then it is possible. That is why we damp drivers to avoid that from happening. The higher the damping the better.

Mvw2 somehow thinks a fast driver is a poorly damped driver which is exactly the opposite of what it is. Mvw2 also thinks it is possible to overdamp.

So far none of what has been talked about translates to how a driver can be too fast. When instantaneous impulse response is what we want in audio. I doubt there is any driver that is better than the laws of physics allows.

I can only take "body of note" to mean all the frequencies required to make a musical instruments note that gives it timbre. All of that information in in the signal from the amp. I can't imagine how a driver can be too fast (unless improperly damped) to change the signal. If a driver changed the signal it is distorting. period.
post #95 of 142
Quote:
Originally Posted by average_joe View Post

And from my understanding, waves don't have acceleration/deceleration, that is a physical property of sound producing element. The wave has amplitude, wavelength, and shape.
If you plot a wave you can calculate its acceleration/deceleration. This is the point mvw2 was trying to make in reference to attack and decay of a note.
post #96 of 142
Quote:
Originally Posted by oarnura View Post

I can only take "body of note" to mean all the frequencies required to make a musical instruments note that gives it timbre. All of that information in in the signal from the amp. I can't imagine how a driver can be too fast (unless improperly damped) to change the signal. If a driver changed the signal it is distorting. period.
I suspect more a case of reverberation albeit on a miniscule scale is what 'body' of note entails seeing as its focal measure point insofar has been attack and more importantly decay.
post #97 of 142
Quote:
Originally Posted by mvw2 View Post
Body isn't abstract because it can be seen. You can look at a wave form and see the body (area under the curve). If you were able to measure air pressure waves and map out intensity versus time, you should end up with the same curve shape and amount of body as the original electrical wave form.
Isn't the area under a curve dependent upon the amplitude when comparing to of the same waveform at the same frequency? OK, I will answer, yes! So are you saying body = amplitude? I am confused as to what you mean.

Quote:
Originally Posted by oarnura View Post
I can only take "body of note" to mean all the frequencies required to make a musical instruments note that gives it timbre. All of that information in in the signal from the amp. I can't imagine how a driver can be too fast (unless improperly damped) to change the signal. If a driver changed the signal it is distorting. period.
Excellent explanation, I understand that! And by all the frequencies, I think you mean harmonics, reflections from the instrument, etc.

Quote:
Originally Posted by communic View Post
I suspect more a case of reverberation albeit on a miniscule scale is what 'body' of note entails seeing as its focal measure point insofar has been attack and more importantly decay.
I am not so clear on what you wrote, maybe it is just me. I think oarnura summed it up well and I give him the x2!
post #98 of 142
Not to change the subject, but does this all apply to William Hung's singing? Sorry, had to break the mood.
post #99 of 142
Quote:
Originally Posted by communic View Post
I suspect more a case of reverberation albeit on a miniscule scale is what 'body' of note entails seeing as its focal measure point insofar has been attack and more importantly decay.
Whatever reverberation a note contains is in the recording. A driver can not add extra "reverberation". That would be distortion.

My main issue here is simple. MVw2 claims the flaw of the phonak is it is too quick. My contention is there is no such thing.
post #100 of 142
Quote:
Originally Posted by communic View Post
If you plot a wave you can calculate its acceleration/deceleration. This is the point mvw2 was trying to make in reference to attack and decay of a note.
Waves don't accelerate/decelerate, drivers do.



Waves lose energy, which is amplitude.

,

This is a good place to do some research/learning in detail. Here is a site that explains a lot of what has been discussed in an easier to understand format, IMO.

Quote:
Originally Posted by tstarn06 View Post
Not to change the subject, but does this all apply to William Hung's singing? Sorry, had to break the mood.
If you want him to sound good, probably
post #101 of 142
You kids are thinking too damn hard.

nG
post #102 of 142
Quote:
Originally Posted by oarnura View Post
Whatever reverberation a note contains is in the recording. A driver can not add extra "reverberation". That would be distortion.

My main issue here is simple. MVw2 claims the flaw of the phonak is it is too quick. My contention is there is no such thing.
And following on what you said, my issue was that IMO the NE-7 soundstage is not in the same league as the PFE soundstage with the grey filters and a good source. And as oarnura pointed out, it is good left-right, but not front back.
post #103 of 142
Thread Starter 
I speak in layman's terms because most of who I speak to on forums are laymen. Most folks in the audio hobby don't know the physics and wouldn't follow the numbers well. This is why I generally talk non-scientifically about scientific things. It tends to make more sense to most people.

The speaker doesn't directly take the signal force. The coil does, but the system doesn't. It reacts to it. Because of this, voltage in does not need to equal force out, nor the acceleration/deceleration rates. Movements are not linear and forces are not necessarily a 1:1 ratio to the input voltage. The non-linearity and difference in movement is what in part makes the differences in sound between speaker A to speaker B.

A source signal has a certain wave shape, not smooth, with variations, thickness, and specific amplitudes. A speaker takes that pattern and attempts to recreate it by moving a mass back and forth and reassembling the input wave into an output pressure wave. There are also spring and dampening forces tied to this system that affect how the diaphragm moves.

The diaphragm does not have to move in conjunction to the input signal. In fact, in a poorly controlled system, it can seem almost random and ends up just a noisy mess, i.e. most crap sub systems people have in their car that run junky, non-linear drivers in tiny, under dampened enclosure setups.

Body is me referring to the shape, area under the curve of the wave. It's sort of like an EQ with a variable Q value you can adjust. Let's say the EQ band is bound to just 1/3 octave, but we can change the Q value and change the shape of the curve, adding or detracting from the area under the curve. My context of attack is the rise of that curve and decay fall of that curve. The area is body. By changing the body, one does not have to change the frequency period. The shape can change without changing the pitch.

Does anyone not agree with this?

I think the biggest gap I have in my written context is the idea that the diaphragm does not move directly in relation to the input signal. Yes, the coil and resulting magnetic field operates directly off the input voltage, but how the diaphragm actually moves doesn't have to mimic the input signal. This depends upon relative spring force, dampening, linearity, etc... This means the diaphragm can move faster then the input signal. It can move slower. It can have too high of an amplitude or too low of an amplitude. The mental hurdle is separating the diaphragm movements from the original source signal and understanding the diaphragm can be moving even when input voltage is zero, moving in a positive direction when input voltage is negative, or visa versa. How it moves depends upon the force, spring, damper, mass relations, and time.

A practical example would be to cut a rubber band, tie a mass to one end, hold on the other end, and bounce it back and forth. Your arm is the motor. What the mass is doing is the diaphragm. The mass is reacting to your inputs, but it doesn't necessarily have to match those inputs. What it's doing at any given time is its own doing relative to the input force, spring, damper system and the particular point in time and action it is doing. It's not linear. It's not direct. It's not instant. This is what a speaker is doing.

The end shape of the movements is the pressure wave. We hope for it to equal the input electrical wave shape, but it generally doesn't. It can be fatter or skinnier, taller or shorter, have random noise or is overly smoothed.

All speakers sound different because they are different systems.

Does any of this not make sense?

A note on dampening. Yes, you can over dampen. I really don't get how you think you can't. If it constrains the movement and prevents the speaker to accurately recreate the shape of the input electrical wave, then it's over dampened. I understand the concept that you can infinitely dampen and if so, the diaphragm would have to move exactly in relation to the motor and exactly at the time of the input signal. However, that's poor thinking. Dampening is a resistance to motion. You apply an input force via the motor and the diaphragm moves overly slow. It accelerates slower and stops sooner then it should. The end sound is tight but constrained.
post #104 of 142
Quote:
Originally Posted by mvw2 View Post
Really? Most setups favor under damped systems. Like a sub when you put it in a box and gear towards a final Qtc of 0.707. 0.5 is critically dampened. 0.3 is over dampened. 0.7 is under dampened. Most setups are desgined to be critically dampened or under dampened to get the lowest F3. Very few setups are of an over dampened orientation, although some folks prefer it to get a tight, controlled type of sound.
Ah car audio rears its head.

The values you quote for Qtc works well for a sealed car sub. It is a good compromise. You are limited in that application by enclosure size. the lower the Qtc the large the enclosure.

Here is an explanation for others who are reading this thread:
Audiogear Reviews - Enclosure Design - Sealed Enclosures

However, the lower the Q value the better. For home audio without something like a servo you can't achieve a very low Q value without causing other problems. The quality of drivers you pick also makes a big difference.

Companies like Rythmik are able to get very low Qtc values without the compromises by using a servo.

Rythmik Audio • Bass Re-radiation
"The conventional solution falls short

Many subwoofers simply ignore the problem because there is little they can do to minimize the problem, and what they can do is not very practical. The conventional solution is to provide a stronger motor to the driver. This gives the driver more control of the cone and a lower Q value. The problem is that this involves a dramatic loss of efficiency and output in order to make a significant improvment.

With our Direct Servo technology, we are able to provide a Q value of 0.28. A conventional subwoofer would have to increase the magnet size by a factor of 3 to achieve the same level of cone control. This would then yield 5 dB less output around fs. A second subwoofer would be required to achieve the previous output level."


Infinite Baffle designs also chose low Qtc values for tighter bass.

IB subwoofer FAQ page
"10) Which Qts/Qtc should I chose?

For an IB, the Qtc of the system is approximately the Qts of the driver. If you like very tight bass, chose a lower Qts driver. If you like the "HT" sound, then chose a driver with a higher Qts. Dual voice coil woofers can be used with a variable resistor in series with the inputs of an unused VC. This allows the "Q" to be adjusted to the listener's wants/needs. One drawback of this is that the power handling is cut by at least 25%. Here's a LINK to the Adire Audio article on RDO (resistively damped operation).

NOTE 1: using RDO or leaving one VC open will decrease the output of each driver -3dB.

NOTE 2: using narrow bandwidth EQ will raise the Qtc of any system. Wideband or shelving filters have significantly less impact on the Qtc

I wrote a post on the forum going into detail as to my personal recommendations for the choice of driver 'Q'. Here's a LINK to that post. There are those rather emphatic that the Qts for a driver operating in free-air must be ~0.7. If that's what you want by all means use driver with that Qts."



So basically you prefer a response similar to your car audio because all you have heard are those systems. Somehow you think that such a frequency response is the only way to go.

The bottom line it the phonak is not flawed. It is more accurate. You prefer an slightly more colored sound.

I am glad it took all that to finally determine that. Cheers.
post #105 of 142
Quote:
Originally Posted by mvw2 View Post
The speaker doesn't directly take the signal force. The coil does, but the system doesn't. It reacts to it. Because of this, voltage in does not need to equal force out, nor the acceleration/deceleration rates. Movements are not linear and forces are not necessarily a 1:1 ratio to the input voltage. The non-linearity and difference in movement is what in part makes the differences in sound between speaker A to speaker B.
The voice coil is connected to cone. The input voltage is a wave of differing polarity. One phase of the singal pushes the cone out. The negative phase pulls the cone in. Your understanding of this basic principle is flawed.

Quote:
A source signal has a certain wave shape, not smooth, with variations, thickness, and specific amplitudes. A speaker takes that pattern and attempts to recreate it by moving a mass back and forth and reassembling the input wave into an output pressure wave. There are also spring and dampening forces tied to this system that affect how the diaphragm moves.
When you say thickness do you mean Wavelength or Background Noise?

Quote:
The diaphragm does not have to move in conjunction to the input signal.
Rubbish. If it doesn't the driver is non linear.

Quote:
Body is me referring to the shape, area under the curve of the wave. It's sort of like an EQ with a variable Q value you can adjust. Let's say the EQ band is bound to just 1/3 octave, but we can change the Q value and change the shape of the curve, adding or detracting from the area under the curve. My context of attack is the rise of that curve and decay fall of that curve. The area is body. By changing the body, one does not have to change the frequency period. The shape can change without changing the pitch.

Does anyone not agree with this?
If the shape of the curve changes the frequency changes. Wavelenth is veloctiy of a wave/ frequency. Sound has a constant velocity in air 343m/s

So a 20 Hz wave will have a wavelength of 343/20 meters or 17.15 meters.

The area under the graph can only change if the frequency changes or the amplitude changes. If a driver is reproducing the 20Hz wave correctly then the only change can be amplitude.

Which would mean driver A is more sensitive than driver B id the amplitude of the 20Hz signal is higher in driver A than Driver B.

So the phonak has less body it would mean it is less efficient than the other IEM.

Quote:
I think the biggest gap I have in my written context is the idea that the diaphragm does not move directly in relation to the input signal. Yes, the coil and resulting magnetic field operates directly off the input voltage, but how the diaphragm actually moves doesn't have to mimic the input signal.
Then you get distortion. Which all drivers do to some extent.


Quote:
A note on dampening. Yes, you can over dampen. I really don't get how you think you can't.
I have explained this already. Your car audio background is limiting your understanding.

Quote:
If it constrains the movement and prevents the speaker to accurately recreate the shape of the input electrical wave, then it's over dampened.
That makes the driver slow. I want you to explain how you can claim a driver is too fast. Go ahead please.

Quote:
I understand the concept that you can infinitely dampen and if so, the diaphragm would have to move exactly in relation to the motor and exactly at the time of the input signal. However, that's poor thinking. Dampening is a resistance to motion. You apply an input force via the motor and the diaphragm moves overly slow. It accelerates slower and stops sooner then it should. The end sound is tight but constrained.
Damping is stopping the driver after the signal is gone. The natural resistance of a driver or damping factor is Qts. You add the electrical damping factor of the Amp. When the coil is moving without a signal it is creating its own electricity which the Amp has to handle. The Amp has to stop this motion.

You are conflating total damping with a slow driver.

I am going to ask you one last time to explain how a driver can move faster than the input signal. Answer that question.

You claimed a driver can be too fast but all you are taking about is a driver reacting too slow. Answer my question above.
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