Naked Science Forum
General Science => General Science => Topic started by: Geezer on 28/09/2010 18:55:22
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When my neighbour is listening to music, I can keep the higher frequencies out of my house by closing the windows and doors, but I can still hear the thump of the low frequencies.
I think it's because the low frequencies exert considerable pressure on the walls and windows of my house so that they re-radiate some of the low frequency sound into my house, whereas the higher frequencies tend to be either reflected or absorbed by the walls and windows.
What do you think?
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High frequencies are very directional so they are relatively easy to deal woth whereas low frequencies are omni directional and thus radiate all over the place. This is why a sub woofer can almost be placed anywhere in a room without ewe knowing where the thumping bass is coming from. ....so..in addition to other elemets for which I am sure klevur peeps here will discuss.... I am sure this must have something do with it.
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High frequencies are very directional so they are relatively easy to deal woth whereas low frequencies are omni directional and thus radiate all over the place. This is why a sub woofer can almost be placed anywhere in a room without ewe knowing where the thumping bass is coming from. ....so..in addition to other elemets for which I am sure klevur peeps here will discuss.... I am sure this must have something do with it.
That's very true! The LF stuff is more omnidirectional. I'm wondering how it can get through the walls and windows if they are all closed. Sound travels on air wavys, so if there is no way for the air to flow, how does the sound get in?
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High frequencies are very directional so they are relatively easy to deal woth whereas low frequencies are omni directional and thus radiate all over the place. This is why a sub woofer can almost be placed anywhere in a room without ewe knowing where the thumping bass is coming from. ....so..in addition to other elemets for which I am sure klevur peeps here will discuss.... I am sure this must have something do with it.
That's very true! The LF stuff is more omnidirectional. I'm wondering how it can get through the walls and windows if they are all closed. Sound travels on air wavys, so if there is no way for the air to flow, how does the sound get in?
hang on !!..ewe expect me to know more stuff ?....lol !!
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...i think ewe are right in the first place with the wall absorption thing !...the sound then travels through the walls...which is not nice !...
.....don't expect any science from me in the way of an explanation ! [;D]
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Damping (http://en.wikipedia.org/wiki/Damping) is proportional to the speed of oscillations (the frequency), so the rapid oscillations (high frequencies) die-off first (temporally and spatially), leaving the slow oscillations, the low frequencies.
Sound travels on air wavys, so if there is no way for the air to flow, how does the sound get in?
Sound also travels through solid (walls, floors, string (http://en.wikipedia.org/wiki/Tin_can_telephone)).
[Surely you've seen the cowboy movies where the indian native American puts his ear to the railroad track to tell if the "Iron horse" was coming].
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Low Frequency sound isn't actually omnidirectional - it just seems like that to us because we i.e. our ears, are very small in comparison to LF sound wavelengths. In dry air, at 20°C a 20Hz sound will have a wavelength of 343m / 20 = 17.15m (conversely, a relatively high pitched sound at 15kHz will have a wavelength of just ~0.02m i.e ~2cm).
It is the long wavelengths of LF sound that make it so difficult to block. I can't remember the exact formula, but to absorb a 20Hz sound, with its 17.17m wavelength, needs something with absorbent features that are several metres in size.
It is precisely because LF sounds are so hard to absorb that fog horns are low pitched - high frequency sounds, although possessing more energy for a given amplitude (because after the amplitude is taken into account the energy of a wave is dependent upon its frequency) have a much smaller wavelength and are absorbed more easily.
It is believed that some whales use infrasound (iirc down to just a couple of Hz) to communicate over hundreds of km. Once again, it is the very long wavelengths that allow the waves to travel so far without being absorbed.
Low frequency EMR is also similarly hard to block and VLF/ELF (Very/Extra Low Frequency) radio is used to communicate with submerged submarines at depths between 10m-40m (30ft - 130ft). However, such low frequencies also have a very low bandwidth and are not capable of high-speed communications. They also require correspondingly large aerials/antennae, both for transmission and reception: the transmission arrays can cover several square km and the subs have to trail a correspondingly long aerial behind them.
http://en.wikipedia.org/wiki/Vlf#Details_of_VLF_submarine_communication_methods (http://en.wikipedia.org/wiki/Vlf#Details_of_VLF_submarine_communication_methods)
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Thanks Lee!
Assuming a rather substantial brick wall, does the sound "travel through" the wall, or does it cause to wall to move slightly in sympathy?
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Sound even travels through a bank vault door or submarine hull ...
(https://www.thenakedscientists.com/forum/proxy.php?request=http%3A%2F%2Fnews.bbcimg.co.uk%2Fmedia%2Fimages%2F48429000%2Fjpg%2F_48429293_bae_systems_thru_hull_montage.jpg&hash=adb0179373f60c0b3d0c3ae14b50fad3)
http://www.bbc.co.uk/news/science-environment-10701182
And pregnant women ...
(https://www.thenakedscientists.com/forum/proxy.php?request=http%3A%2F%2Fupload.wikimedia.org%2Fwikipedia%2Fcommons%2Fthumb%2F1%2F16%2FBaby_in_ultrasound.jpg%2F220px-Baby_in_ultrasound.jpg&hash=6b45941f51bcc844984d57ee6073aa07)
http://en.wikipedia.org/wiki/Ultrasound
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I think that the brick wall must move slightly, but not as much as you'd think because the bricks and mortar are porous, so the air can move within the pores of the wall material. Even if you painted the wall, to make it airtight, the film of paint would flex and transmit the pressure to the internal air cavities within the material.
I've got to confess that this isn't an area I'm overly familiar with - I'm just remembering bits of stuff I looked into a long time ago - but the key thing I do remember is the relationship between the wavelength and the feature size of the absorbing or reflecting material, which iirc, pretty much dictates whether the sound is absorbed or reflected.
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RD's submarine case is interesting too. Hopefully there is not too much air entrained in the steel hull [:D]
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Can I just say that I told LeeE and RD to write all that klevur stuff !!
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Good points, Lee. I wonder if the thickness of the wall also has to do with setting up standing waves to cause reflection. Another point is that the absorption coefficient of an object generally has a part that's proportional to its thickness, measured in wavelengths. Since bass wavelengths are large, the wall won't absorb terribly well.
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I think the low frequency oscillating air pressure couples energy into the wall to produce a sympathetic oscillation in its outer surface. The amplitude of the sympathetic oscillation is very small, but it sets up an acoustic wave that propagates through the wall by compressing/decompressing the material.
When the acoustic wave terminates at the inner surface of the wall, it produces a small deformation at the surface that couples acoustic energy into the air inside the room.
Shorter wavelengths tend to be reflected from the outer surface of the wall if it is hard, or dissipated into the material if it is soft.
Shorter wavelengths will propagate through a rigid wall reasonably well if the energy is effectively coupled into the wall (e.g. tap a solid wall with a hammer) but shorter wavelengths in air can't couple much energy into a dense wall.
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Shorter wavelengths will propagate through a rigid wall reasonably well if the energy is effectively coupled into the wall (e.g. tap a solid wall with a hammer) but shorter wavelengths in air can't couple much energy into a dense wall.
Good point! So is there an easy explanation why tapping on a wall with a hammer is so much more effective than blasting high frequency noise at it from a stereo? Does it just have to do with the density of the hammer being so much greater than the density of the air or is there something else at work?
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Shorter wavelengths will propagate through a rigid wall reasonably well if the energy is effectively coupled into the wall (e.g. tap a solid wall with a hammer) but shorter wavelengths in air can't couple much energy into a dense wall.
Good point! So is there an easy explanation why tapping on a wall with a hammer is so much more effective than blasting high frequency noise at it from a stereo? Does it just have to do with the density of the hammer being so much greater than the density of the air or is there something else at work?
Yes. I think the difference in density between a hammer and air has a lot to do with it. It's not so much that walls can't transmit high frequencies. I think they can. It's more a question of how effective the energy coupling method is.
There is something a bit different from RF/light going on too, I think. Higher frequency light has more energy than lower frequency light. Sound sort of goes the other way. There is a lot of energy in a LF sound wave. That's why LF sounds can travel for much greater distances - I think!
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http://en.wikipedia.org/wiki/Anechoic_chamber
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http://en.wikipedia.org/wiki/Anechoic_chamber
Ah yes! We built an RF chamber for RF susceptibility and interference work, but it also worked quite well as an acoustic chamber.
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Shorter wavelengths will propagate through a rigid wall reasonably well if the energy is effectively coupled into the wall (e.g. tap a solid wall with a hammer) but shorter wavelengths in air can't couple much energy into a dense wall.
Good point! So is there an easy explanation why tapping on a wall with a hammer is so much more effective than blasting high frequency noise at it from a stereo? Does it just have to do with the density of the hammer being so much greater than the density of the air or is there something else at work?
Excellent point is a hammer striking a wall actually is setting an energy node point, size of the hammer head, directly coupled on the surface of the wall.
While airborn noise is couples the total exposed area surface displacing the pressure to a wider area.
Think of this a pressure of lbs/in2
The node point is the point of origin of the energy to be spent.
What is easier to send cleanly through a board with the same reasonable instantanious pressure, a nail or a steel face plate
Maybe I should say Focal point not node point.
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Just like RF? At the bottom of the URL page is a video that explains something of interest to audio low pitch sound
http://bbamusic.wikispaces.com/Audio+Standing+Waves+-+Room+Modes
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There may be another factor.
A higher frequency sound wave will interact with the wall in such a way that the maxima and minima of the wave encounter the wall at significantly different times - if you see what I mean.
So, the wall will receive a rather mixed signal as maxima and minima will be distributed across the surface of the wall. These will tend to interfere with, and cancel each other.
The lower the frequency, the greater the spatial separation between maxima and minima, so there will be less interference and canceling.
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http://en.wikipedia.org/wiki/Anechoic_chamber
Ah yes! We built an RF chamber for RF susceptibility and interference work, but it also worked quite well as an acoustic chamber.
I experienced a small full Anechoic/acoustic chamber. I felt a little bit dizzy at the first time entering and having the door shut. A bit claustrophobic?
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There may be another factor.
A higher frequency sound wave will interact with the wall in such a way that the maxima and minima of the wave encounter the wall at significantly different times - if you see what I mean.
So, the wall will receive a rather mixed signal as maxima and minima will be distributed across the surface of the wall. These will tend to interfere with, and cancel each other.
The lower the frequency, the greater the spatial separation between maxima and minima, so there will be less interference and canceling.
YES you hit it on the head. I seen this idea in my minds eye but could not put it to coherent words.
I believe the pyramid shape blocks of the acoustic chamber design enhances deflection not reflection reducing the energy of the low freq longwaves
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Sure would be something to use a targeting optics to be able to view the vibration at a long distance, convert and invert the audio signal duplicate through DSP and send it back at the same magnitude to its audiable source. Somewhere along the path of travel the should have some canceling affect.
I don't think it would be a cost effective idea
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Sure would be something to use a targeting optics to be able to view the vibration at a long distance, convert and invert the audio signal duplicate through DSP and send it back at the same magnitude to its audiable source. Somewhere along the path of travel the should have some canceling affect.
I don't think it would be a cost effective idea
I think you can do something like that to eavsedrop on conversations by pointing a laser "microphone" at a window. Not that I actually have one or anything!
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... eavsedrop on conversations by pointing a laser "microphone" at a window.
Allegedly laser microphones (http://en.wikipedia.org/wiki/Laser_microphone) can be jammed by taping a vibrating "marital aid" to the window [:I]
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Use rubber, then let it bounce, ah, the sound?
Wasn't that what they used amongst other, trying to soundproof a room from those laser microphones? Stripes of rubber, hanging like a curtain?
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Another allusion to marital aids?
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No, I actually have some remembrance of them using this technique Geezer.
All Embasssy's have a room like that.
Don't know what they call it though?
the R*'er room
*Hurt*
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There is something a bit different from RF/light going on too, I think. Higher frequency light has more energy than lower frequency light. Sound sort of goes the other way. There is a lot of energy in a LF sound wave. That's why LF sounds can travel for much greater distances - I think!
Ah, this is where the amplitude comes in. For a given amplitude HF sound does have more energy than LF sound but LF sounds are typically produced at much greater amplitudes than HF sounds.
If you watch an audiophile loudspeaker with the covers removed, while playing music with a high bass content, you'll be able to see the LF/bass driver move quite clearly but you'll not see any visible movement of the mid or HF units. I've seen bass/LF cones travel over 1cm (each way) but the extremely light weight metal dome HF drivers commonly used in audiophile loudspeakers (which are typically around one inch/25mm in diameter) would collapse if they tried to match the travel of the bass/LF driver cones (even though HF domes have a much smaller area than LF cones they are trying to move the air much more quickly - in fact up to a thousand times more quickly - and they can only move that quickly be having very little inertia, which equals very little mass, which in turn means very little material).
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... eavsedrop on conversations by pointing a laser "microphone" at a window.
Allegedly laser microphones (http://en.wikipedia.org/wiki/Laser_microphone) can be jammed by taping a vibrating "marital aid" to the window [:I]
That shouldn't work, because the 'marital aid' will be operating at a much lower frequency than speech (Iirc, one of the limits of our sense of touch is that pressure pulses above ~20-30Hz in frequency are felt as a continuous pressure, which would make a HF 'marital aid' ineffective - speech is typically several hundred Hz and upwards).
In any case, laser microphones work by using the vibrating surface to modulate the stable laser frequency, so it would be trivial to filter out the stable LF frequency of the 'martial aid' as well.
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There is something a bit different from RF/light going on too, I think. Higher frequency light has more energy than lower frequency light. Sound sort of goes the other way. There is a lot of energy in a LF sound wave. That's why LF sounds can travel for much greater distances - I think!
Ah, this is where the amplitude comes in. For a given amplitude HF sound does have more energy than LF sound but LF sounds are typically produced at much greater amplitudes than HF sounds.
If you watch an audiophile loudspeaker with the covers removed, while playing music with a high bass content, you'll be able to see the LF/bass driver move quite clearly but you'll not see any visible movement of the mid or HF units. I've seen bass/LF cones travel over 1cm (each way) but the extremely light weight metal dome HF drivers commonly used in audiophile loudspeakers (which are typically around one inch/25mm in diameter) would collapse if they tried to match the travel of the bass/LF driver cones (even though HF domes have a much smaller area than LF cones they are trying to move the air much more quickly - in fact up to a thousand times more quickly - and they can only move that quickly be having very little inertia, which equals very little mass, which in turn means very little material).
Ah! Thanks Lee.
I do have a qestion though. I'm pretty sure most of the power from the amplifier is going to the LF speakers, whereas the HF speakers need hardly any power at all.
So, would that not mean there is a lot more energy produced by the LF speakers? Or are the HF speakers just much more efficient?
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I'm not sure that most of the energy from the power amplifier is going to the LF unit; in high-end audiophile systems using active crossovers you'll usually find that the same power amplifiers i.e. the same model, with the same power rating, are used for each frequency band and driver; the HF, Mid and LF drivers will have the same power available to them.
In an active crossover loudspeaker system the crossover sits between the preamplifier and the power amplifiers (note the plural), whereas a passive crossover sits between the power amplifier and the loudspeaker drivers. A passive crossover then, must not only split the signal into separate frequency bands, before it is fed to the drivers, but it must do so upon a potentially very high power signal - up to several hundred Watts (mostly due to the currents being used, not the voltage) - without sapping an excessive amount of that signal energy.
An active crossover though, as it doesn't need to cope with such high currents, can be made with more finely produced and higher tolerance components, but it does then mean using a separate power amplifier for each of the active crossover frequency band outputs. Active crossover audiophile stereo systems using six identical power amplifiers are not unknown - three per side, one each for LF, Mid & HF.
Active crossovers are also normally used in high-power P.A. and sound reinforcement systems too, where powers in the kilowatt range may be found - trying to pump a kW through some of the components in a passive crossover would make a good impression of a bar heater.
(sorry about having to digress into active/passive crossovers but the statement about using the multiple identical power amplifiers for each frequency band and driver might not have made much sense without understanding the system)
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I'm not sure that most of the energy from the power amplifier is going to the LF unit; in high-end audiophile systems using active crossovers you'll usually find that the same power amplifiers i.e. the same model, with the same power rating, are used for each frequency band and driver; the HF, Mid and LF drivers will have the same power available to them.
In an active crossover loudspeaker system the crossover sits between the preamplifier and the power amplifiers (note the plural), whereas a passive crossover sits between the power amplifier and the loudspeaker drivers. A passive crossover then, must not only split the signal into separate frequency bands, before it is fed to the drivers, but it must do so upon a potentially very high power signal - up to several hundred Watts (mostly due to the currents being used, not the voltage) - without sapping an excessive amount of that signal energy.
An active crossover though, as it doesn't need to cope with such high currents, can be made with more finely produced and higher tolerance components, but it does then mean using a separate power amplifier for each of the active crossover frequency band outputs. Active crossover audiophile stereo systems using six identical power amplifiers are not unknown - three per side, one each for LF, Mid & HF.
Active crossovers are also normally used in high-power P.A. and sound reinforcement systems too, where powers in the kilowatt range may be found - trying to pump a kW through some of the components in a passive crossover would make a good impression of a bar heater.
(sorry about having to digress into active/passive crossovers but the statement about using the multiple identical power amplifiers for each frequency band and driver might not have made much sense without understanding the system)
Thanks Lee!
I'm pretty sure the power input capacity of HF speakers is a lot less than the power input capacity of LF speakers (for a particular matched set of course), and I do know that heat dissipation can be a real issue with LF speakers, but I've never heard of it being an issue with HF speakers. I'm fairly confident the amp is dumping a lot more power into the low end.
Perhaps the coupling efficiency between the speaker and air is much greater at higher frequencies?
Slightly later:
Come to think of it, I seem to remember that there is an inverse relationship between the efficiency of LF speakers and the amount of distortion they produce. Unfortunately, I can't remember why! Possibly something to do with the stiffness of the cone?
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A simple reason for using identical amplifiers with a system using an active crossover is that by using identical amplifiers the distortion produced by the amplifier is identical for each channel, and is not so objectionable as a single amplifier that is sonically different. As well remember the midrange and tweeters will use power as well, although they generally dissipate more as sound than heat, whilst the big low end dissipates more power as heat than as sound.
As to blocking out low frequency, it is easier to block out high frequencies as you are effectively absorbing more energy per wave per unit thickness the higher in frequency you go, thus a rubber block that absorbs well at 10kHz would have to be 10 times thicker to offer the same absorbtion at 1kHz, and 100 times thicker at 100Hz.
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That may be a good point about the sound vs. heat produced, although iirc, ferrofluid was first used in tweeters to help dissipate heat (by thermally coupling the voice coil to the magnet), implying that heat dissipation was less of a problem in woofers, possibly because a woofer/LF driver won't need to move as quickly as an HF unit and could afford to use a heavier voice coil than a tweeter. However, that does imply that similar levels of energy are being used in each driver, for if bi/tri amping works with identical amps then the individual drivers will have similar impedances and draw similar currents.
Also...
As to blocking out low frequency, it is easier to block out high frequencies as you are effectively absorbing more energy per wave per unit thickness the higher in frequency you go, thus a rubber block that absorbs well at 10kHz would have to be 10 times thicker to offer the same absorbtion at 1kHz, and 100 times thicker at 100Hz.
dunno if the relationship is so linear but looking at it from the energy angle is a very good point too.
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LeeE, ferrofluid is used in tweeters as they have a combination of a very small and low mass coil, but with a very high power load applied, as well as a very short travel. The fluid thus is able to stay in the gap, and is not going to suffer enough movement to create shear. In a woofer the coil travels a long way, but is a lot bigger, and has a much larger surface to radiate heat away to the pole pieces and the magnet basket. This means it can stay cool by radiating away heat from a large area, helped by the airflow from the cone pumping the air past the coil breaking any boundary layers there.
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I fear we may be on a bit of a snipe hunt here. The efficiency of loudspeakers (or drivers as they seem to call them these days) is so poor that it's probably useless to try to equate the power input to the acoustic wave pressure in a meaningful way.
It's coming back to me a bit now. Acoustic efficiency usually suffers at the expense of fidelity, which tends to explain why the most high fidelity systems need fairly gigantic power outputs, even though they are only being used in relatively small spaces. If you were ever able to convert, say 100 Watts of electrical energy into acoustic energy in your living room, you'd probably blow out your windows and seriously damage your hearing.
"Most loudspeakers are actually very inefficient transducers; only about 1% of the electrical energy sent by an amplifier to a typical home loudspeaker is converted to acoustic energy."
from - http://en.wikipedia.org/wiki/Loudspeaker#Efficiency_vs._sensitivity
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It's not just because of the relatively low acoustic efficiency of the loudspeakers that audiophile amplifiers tend to be relatively high powered, and in some cases they're not.
The real key is how much current a power amplifier can supply. Generating the voltages, which dictate the amplitude of the cone/dome displacement and hence the audible volume, is relatively easy but the fidelity with which the profile of the cone/dome displacement follows that of the signal comes down to the magnitude of the force driving the cone/dome, which depends upon the current: a greater current flowing through the voice coil will have more authority and accelerate and decelerate the mass of the cone/dome against the air more quickly than a smaller current.
There are quite a few relatively low power (down to 10~15W) audiophile amplifiers, at least in terms of their continuous rating, but they are usually capable of delivering transient current levels well beyond those of a comparably powered non-audiophile amplifier.
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which depends upon the current: a greater current flowing through the voice coil will have more authority and accelerate and decelerate the mass of the cone/dome against the air more quickly than a smaller current.
Sort of. It's a function of the ampere-turns in the voice coil. You can achieve the same force with a lot less current if you have a lot more turns. You could build a speaker driver that used a relatively a small current, but it would have to operate at a much higher voltage, so it really is power that's applied to the speaker to do work on the air.
Of course, the load that the driver presents to the amp is rather complex. The coil produces a back EMF, and the impedance is anything but resistive, so the whole business tends to turn into a bit of a "black art", and quality becomes a bit subjective.
However, as I pointed out, I don't think we can infer too much about the actual acoustic energy based on the power that's being applied to the speakers drivers.
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Hello everyone, new to the forum.
As a sound engineer I can confirm that there is much more energy in the low end in most types of music. Most musical instruments generate most of their energy at the root note, which will be between 40Hz for a low E on a bass guitar up to a few kHz for most instruments. Few instruments have a significant fraction of energy in harmonics, apart from perhaps electric guitar through a marshall stack turned up to 11, or some synthesizer sounds, and cymbals (that contain hardly any root frequency as the wavelength is longer than the diameter of the cymbal so the waves from both sides meet and cancel). Though nearly all music has roots notes for nearly all the notes in any song below 1kHz.
As such the majority of energy produced from the source is in the low end. Obviously if the source has it, an accurate reproduction from loudspeakers will have it also. Such a system will produce a flat response with a sine wave sweep across all frequencies however.
The reason for matching amps in active crossover systems is as previously stated to ensure an accurate sound. All amplifiers have different characteristics, including input impedence etc. Why go to the effort of creating a finely tuned crossover if this is to a certain extent undone and randomised by the inherent tonal differences between two types of amps? The exception is active monitors, which are standard in all recording and mastering studios. Some have lower powered amps for the tweeters, however these are bespoke designs and the manufacturer can ensure electrical alignment of the LF and HF amps, espcially as they can be fed from the same transformer. This approach is possible with seperates amplifiers, but we're then dealing with economics and whether the manufacturer will do the r&d for two types of amps that may or may not be used together being electrically matched for biamping purposes (Rotel do, others don't), or do the r&d for one amp only and make it cheaper therefore and sell more of them.
The question of audiophile amps that have low outputs but large peak outputs is really getting into the difference between valve and solid state amps, and is a whole different area of study. Some of the effects on loudspeaker reproduction come down to solid state amps having a negligable output impedence (and therefore a speaker will have the same frequency response from any such amp) whereas valve amps have an output impedence which affects the electrical q of the whole system. (Solid state amps are effectively removed from the electrical q equation of the loudspeaker, whereas valve amps are not)
Most tweeters are rated at the same wattage or thereabouts as the woofers in the same set of speakers. However they typically are recieving much less energy than the woofers but thats due to what is in most audio signals sent to it. Studio monitors need to be rated the same because even the most experienced engineer will occasionally sent 100w of 10kHz through them, and will need them to still be working when his ears have stopped ringing and he can continue with the session.
The reason why LF drivers are bigger is the same as why LF goes through walls - wavelength. Headphones are small and are tuned to reproduce low frequencies, but the energy levels are so low a small driver can achieve it. In loudspeakers at the much higher energy levels to transfer then energy to the air the loudspeaker must physically move much more air because of the long wavelengths, requiring the speaker either to travel further (driver travel is called Xmax) or be a larger driver.
Sean B is correct with the relationship of thinkness of wall and frequency, because when you see it in terms of wavelength it makes sense. If a wall is as thick as the wavelength it will absorb and reflect to a much higher degree than if the wavelength is much longer then the wall thickness. Of course the wall can be stiffer and reflect more, or be softer and absorb more, and in the case of rockwool absorbers reflect almost nothing but absorb a lot by turning the waves into heat.
LF travels through walls by vibrating in sympathy and re-radiating on the other side. As there is such a large surface area the movement can be tiny but still transfer a lot. For an equivalent surface area of a typical driver a brick wall will produce very little sound, but for a whole wall it is significant.
I hope I've added to the thread, I know I haven't given particularly scientific explanations, but as a pro sound engineer as you'd expect I just get on with it understanding the principles but not needing to know the exactitudes :)
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... eavsedrop on conversations by pointing a laser "microphone" at a window.
Allegedly laser microphones (http://en.wikipedia.org/wiki/Laser_microphone) can be jammed by taping a vibrating "marital aid" to the window [:I]
That shouldn't work, because the 'marital aid' will be operating at a much lower frequency than speech (Iirc, one of the limits of our sense of touch is that pressure pulses above ~20-30Hz in frequency are felt as a continuous pressure, which would make a HF 'marital aid' ineffective - speech is typically several hundred Hz and upwards).
In any case, laser microphones work by using the vibrating surface to modulate the stable laser frequency, so it would be trivial to filter out the stable LF frequency of the 'martial aid' as well.
Evidently these devices do produce audible sound …
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http://www.freesound.org/samplesViewSingle.php?id=51305
Even if the output was exclusively infrasonic its amplitude could be so large relative to the vibrations caused by speech that it would drive the detector onto saturation (http://en.wikipedia.org/wiki/Clipping_%28audio%29#Repairing_a_clipped_signal): i.e. not enough dynamic range to make a comprehensible recording of the speech.
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which depends upon the current: a greater current flowing through the voice coil will have more authority and accelerate and decelerate the mass of the cone/dome against the air more quickly than a smaller current.
Sort of. It's a function of the ampere-turns in the voice coil. You can achieve the same force with a lot less current if you have a lot more turns. You could build a speaker driver that used a relatively a small current, but it would have to operate at a much higher voltage, so it really is power that's applied to the speaker to do work on the air.
Yes, having more turns in the voice coil would allow you to achieve the same force with less current and a higher voltage, but afaik that's not the case; neither active not passive crossover systems would work if the drivers had significantly different impedance unless you used equally different amplifiers, combining ones that delivered low voltage and high currents with ones that delivered high voltages and low currents, which isn't the case. Like I said in an earlier response, the fact that similar amplifiers are used for the different drivers implies relatively little difference in the impedances, voltages and currents applying to those different drivers.
elfabyanos: thanks for your very well informed response. I didn't know about the bespoke monitors with different custom amps for the different drivers. However, I think I have to disagree with your assertion that low-power but high-current capability is really about the different between solid-state and valve amplifiers. Excluding transformerless valve amplifiers, because they're pretty rare and overly expensive beasts, it's true that the output transformers in valve amplifiers result in a higher output impedance than typically found with solid-state amplifiers but the high current capability in relatively low-power solid-state audiophile amplifiers mostly comes from using a combination of over-specified power supplies and higher tolerance and quality components. The higher-spec power supplies are able, on a transient basis, to deliver considerably more (clean) power than their continuous rating would suggest, but this then needs the higher tolerance and quality components in the signal path to cope with those transient bursts without degrading or failing. Normally, in solid-state amplifiers, the peak power is a fixed ratio to the RMS power but in these audiophile amplifiers the peak power can be considerably higher.
A good part of the elevated prices one has to pay for these audiophile amplifiers is because of the higher spec components used in them.
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Like I said in an earlier response, the fact that similar amplifiers are used for the different drivers implies relatively little difference in the impedances, voltages and currents applying to those different drivers.
Er, but I think you'll find the actual impedance is very much a function of frequency (amongst other things). The number of ohms stamped on the label has almost nothing to do with the impedance the driver presents to the amplifier in operation.
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elfabyanos: thanks for your very well informed response. I didn't know about the bespoke monitors with different custom amps for the different drivers. However, I think I have to disagree with your assertion that low-power but high-current capability is really about the different between solid-state and valve amplifiers. Excluding transformerless valve amplifiers, because they're pretty rare and overly expensive beasts, it's true that the output transformers in valve amplifiers result in a higher output impedance than typically found with solid-state amplifiers but the high current capability in relatively low-power solid-state audiophile amplifiers mostly comes from using a combination of over-specified power supplies and higher tolerance and quality components. The higher-spec power supplies are able, on a transient basis, to deliver considerably more (clean) power than their continuous rating would suggest, but this then needs the higher tolerance and quality components in the signal path to cope with those transient bursts without degrading or failing. Normally, in solid-state amplifiers, the peak power is a fixed ratio to the RMS power but in these audiophile amplifiers the peak power can be considerably higher.
A good part of the elevated prices one has to pay for these audiophile amplifiers is because of the higher spec components used in them.
I agree, I didn't make my point particularly clearly. There are two ways to achieve large transients in a musically pleasing way, both are available to valve amps, only one is available to solid state.
With solid state it is as you say using higher quality components to achieve it, but I wouldn't say its (primarily) down to the quality of of the components in the signal path. The real work in a good solid state amp is the quality of the power supply that you mention, its the a double act between the transistor being well engineered to have a good slew rate and also the transformer being good enough to deliver the power when the transistor asks for it. The effect this has is not only audible on low frequency sounds, but can be heard in better clarity in all frequencies, and this is subjectively recognised in the sounds having a better clarity of separation, and a more focused soundstage, due to the accuracy of the timing. It also helps the bass as the LF drivers are mechanical and they have a kind of stiction or lag that a fine quality transient signal can overcome quicker, leading to a clearer and more dynamic bottom end. However in terms of percieved loudness, the ear doesn't take much notice of transients, so a high quality 15w solid state amp is going to sound pretty much exactly as loud as a cheap amp at 15w into the same set of speakers. But, audiophiles with expensive amps tend to have expensive speakers that are more sensitive and have a better frequency range which more than makes up for it.
15w is still a fair amount anyway - in terms of annoying the neighbours this can usually be achieved at 5w RMS through a brick wall if they're trying to sleep. As a rule of thumb, whenever you're listening to music and you decide to turn it up a notch (more than a tiny adjustment, but not enough to disturb the baby) it would typically be about 3db, which is twice the power. Yet to a human, twice as loud is about 10db, or 10 times the power. In this way the difference between a 15w amp and a 50w amp is about 5db - a difference well within the range of differing speaker's various sensitivities. Cheap speakers are typically 83-86db (often with a small resonant peak at 1kHz to up the sensitivity figures whilst giving you nothing below 80-90Hz which is the bottom end of an electric guitar, bass guitar is a whole octave lower) and good ones are between 88-92db. This means an audiophiles 15w hifi is as loud as your average 50w all in one system, and its mainly down to the speaker.
Back to the amp - obviously once this good audio signal has been produced it is important to keep it in good shape - it is relatively easy to obtain components that can 'cope' with the transients. Cheap components cost very little, and there are ones available for all the power requirements one would want at a cheap price because as far as the world of electronics goes amplifiers are not high current. However in audiophile amplifiers expensive ones are used so as not to degrade the sound.
There's also the issue of class of operation which affects the sound quality in general but can affect the transient response in the switching class, whereby the entire circuit is re-phased as the audio signal crosses into negative, because current in a transistor can only flow in one direction. In the class where two transistors are used in opposite phases the transient issue at switching less pronounced, and in the highest quality class of using a single transistor amplifying a DC-offset signal in the positive (the DC being rectified out afterwards to leave the AC audio signal) the transients are little affected at all. Switching however introduces harmonic distortion, especially in the ones that switch the entire transistor circuit - found in cheap cheap cheap stuff.
Valve amps can achieve better transient characteristics in much the same way but I couldn't explain how, but I would assume that the double act of power supply and valve applies.
On to the second way to generate pleasing transient peaks - harmonics (used a lot in the recording studio by the way, to enrich certain sounds or even an entire mix in the mastering process). Valve amps tend to produce 2nd order harmonics, which is an octave and fits in with the original signal as far as the ear is concerned. Transistors tend to generate 3rd order harmonics which is a 5th above the octave above. This means when a guitar's top E is distorted through a transistor you also get a B an octave above. Ok on single notes, on a whole bunch of notes it goes catastrophically wrong. For example in an E major triad chord the middle note generates an Eb, and having notes a semitone apart in the scale is starting to get jazzy to say the least. Have a whole song doing it and its a mess.
Valve amps therefore can be cranked beyond their rated output without too much pain. In fact many listeners will report that the sound actually gets better as the total harmonic distortion rises even to 10%. Another quirk of valve amps is that beyond their rated output they do actually kick out more juice. A Marshall head from the late 60s rated at 100wRMS will be kicking out over 190wRMS at 40% total harmonic distortion. A similar 100w transistor amp will still only be putting out 100w at the same level of distortion, and it will sound awful.
Producing decent hifi though is a dark art, as not only do the engineers need to understand the science, but it is all worthless if they can't make the subjective decisions about what 'sounds' better, as no amount of measurement can tell you that. Hence the endless debates that rage on in the audiophile world (less so in the sound engineer world I have to say).
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Evidently these devices do produce audible sound …
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Even if the output was exclusively infrasonic its amplitude could be so large relative to the vibrations caused by speech that it would drive the detector onto saturation (http://en.wikipedia.org/wiki/Clipping_%28audio%29#Repairing_a_clipped_signal): i.e. not enough dynamic range to make a comprehensible recording of the speech.
When pressed up against a window much of the higher frequencies wouldn't get transferred as much, so the frequency analisys would be skewed to the lower end, though some would. It would still be difficult, as you say because the aid will be so much louder than the speech trying to be recorded. It wouldn't necessarily be saturated though, that would be an issue of calibration at the listener's equipment. The issue would be would the dynamic range be enough to capture the much quieter speech. I would say with 24 bit recording that microphones are easily capable of it would. I am unaware what the dynamic range of a laser would be, I see no reason why it couldn't have the sane if not more dynamic range.
Trying to isolate the aid's vibrations would be done using a denoising software, which usually can be told to take a section as the noise to remove which you would then select as being a section with no speech or other background noise. The software with then use this spectrum as a key and remove the same frequencies by the same amount in the key. I'm not saying its possible, but its amazing what is, I do some audio restoration work and I guess it would be quite similar.
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Anyway, I think I sort of derailed this thread by suggesting that we can get an idea of the acoustic energy based on the power that goes into the speaker. I realize that's highly suspect for the simple reason that so little of the electrical energy supplied by the amplifier is actually converted into acoustic energy by the speaker. Speakers are very inefficient transducers (personally, I think this is a very good thing!)
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Depending on the driver (speaker), most frequencies below 1,000 hz. become a sound pressure, not a sound frequency issue. Because of the pressure created these frequencies are hard to block. Imagine having a 12" thick solid door on your home, that may block out someone yelling at your door however, if they started beating on your door you would hear it. If u need to block these frequencies good luck. Unless you live in a house with 12" thick walls and no windows your always going to hear some of those low frequency noises by way of SPL. ( sound pressure level). You can reduce the noise level with certain types of acoustical insulation but never block it out all together, unless you want to call the police on your neighbor for disturbing the peace. Sorry, hope this helps you at least with information
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Geezer: yes, the impedance of the driver will vary with frequency, but not so far as to take into completely different realms of voltage and current delivery.
elfabyanos: I think we should rule out the case of valve amps sounding better than solid-state amps when there's a significant degree of distortion - I entirely agree with what you say about our perception of odd and even ordered harmonic distortion - but ideally we want no distortion, at least in the reproduction of sound (as opposed to the creation of sound e.g. the 2nd order harmonic distortion in an electric/electronic instrument amplifier).
Other than that, the only things I'd like to add are that better quality components will have tighter tolerances and will be more linear, and that the power supplies will be likely to incorporate larger capacitors, capable of storing more energy and creating a greater reserve for when those large transients do occur.
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Geezer: yes, the impedance of the driver will vary with frequency, but not so far as to take into completely different realms of voltage and current delivery.
Lee - Take a butcher's at this. It may change your opinion. http://users.ece.gatech.edu/mleach/papers/vcinduc.pdf
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elfabyanos: I think we should rule out the case of valve amps sounding better than solid-state amps when there's a significant degree of distortion - I entirely agree with what you say about our perception of odd and even ordered harmonic distortion - but ideally we want no distortion, at least in the reproduction of sound (as opposed to the creation of sound e.g. the 2nd order harmonic distortion in an electric/electronic instrument amplifier).
Other than that, the only things I'd like to add are that better quality components will have tighter tolerances and will be more linear, and that the power supplies will be likely to incorporate larger capacitors, capable of storing more energy and creating a greater reserve for when those large transients do occur.
Yes, the design of the amplifiers is very important, although the speakers and the acoustic characteristics of the room in which they are placed will probably have a far greater impact on the perceived sound quality than anything else. But what does this have to do with the original question?
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Geezer: yes, the impedance of the driver will vary with frequency, but not so far as to take into completely different realms of voltage and current delivery.
Lee - Take a butcher's at this. It may change your opinion. http://users.ece.gatech.edu/mleach/papers/vcinduc.pdf
Geez: I'm simply not prepared to wade through eight pages of an academical paper in search of what, exactly?
C'mon - don't expect me to put all the effort into proving myself wrong - that's your job. And I'm not saying that the paper you referred to is not relevant, but just pick out and quote the bit of it that is relevant to this thread, and then give the link to the entire paper so I can verify it.
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Geezer: yes, the impedance of the driver will vary with frequency, but not so far as to take into completely different realms of voltage and current delivery.
Lee - Take a butcher's at this. It may change your opinion. http://users.ece.gatech.edu/mleach/papers/vcinduc.pdf
Geez: I'm simply not prepared to wade through eight pages of an academical paper in search of what, exactly?
C'mon - don't expect me to put all the effort into proving myself wrong - that's your job. And I'm not saying that the paper you referred to is not relevant, but just pick out and quote the bit of it that is relevant to this thread, and then give the link to the entire paper so I can verify it.
I found the paper quite interesting. I thought you would too. However, if it's bamboozling you, just look at figs 9 and 11 and you can see how much the impedance increases with frequency. No big surprise really. A speaker has a lot of inductance and mechanical resonance, both of which have a significant impact on the impedance at various frequencies.
The bottom line is that you can infer almost nothing about the acoustic energy produced by a speaker based only on the amplitude of the signal across its terminals. Not only is it a very inefficient transducer, but it has a complex frequency response.
Slightly off topic of course, but that's one of the dirty little secrets of the audio equipment industry. You can produce an amplifier that does what an amplifier should do, i.e., faithfully reproduce the waveform that's stuck in the input with a much greater amplitude and capable of driving a low impedance load without distorting it. Unfortunately, the sound "quality" is heavily influenced by the speaker, but the amp has not the faintest clue about what's actually coming out of the speaker.
Turns out, if you introduce various types of distortion into the signal (valve characteristics etc etc etc) you can apparently "improve" the "quality". But the "quality" is largely subjective, and this is a really great thing for the audio industry because it allows them to eternally make "improvements" that obsolete previous generations of equipment just by making a few more spurious claims that no one will ever be able to really quantify - but it makes for good conversation down the pub.
Sorry, but I've been watching this circus for fifty years, and it just keeps going round and round and round.....
BTW - I have no particular interest in proving either of us right or wrong as you seem to suggest. I've already pointed out that I made some bad assumptions in this thread.
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Lee - Take a butcher's at this. It may change your opinion. http://users.ece.gatech.edu/mleach/papers/vcinduc.pdf
Ignore this, there is nothing you can take away from it that will apply to a listening envirnment. A driver's free air resonance and impedence characteristics are of interest only to the loudspeaker's designers.
What your amplifier sees is the impedence curve for your loudspeaker system. The very same 8ohm driver in a closed cabinet might create an impedence of 40ohms at 50Hz, but in a bass-reflex the impendence at 50Hz might be 2.5ohms. Closed box cabinets create a high impendence at system resonance (Fs) off set by the resonance so in a well tuned system there is no significant increase or drop in perceieved output. Indeed there isn't in terms of energy transferred to the air as sound.
Bass reflex loudspeakers have an incredibly low impedence at the port tuning frequency, often accompanied by two peaks either side (in a well tuned system, in a bad one the peaks will be lop sided one way or another), which is typically also near fs. What happens is that the driver vibrates much less than if it was in a closed cabinet, but the air in the port resonates instead and takes over the audio output at those low frequencies. This is also the cause of the drop in impedence.
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Lee - Take a butcher's at this. It may change your opinion. http://users.ece.gatech.edu/mleach/papers/vcinduc.pdf
Ignore this, there is nothing you can take away from it that will apply to a listening envirnment. A driver's free air resonance and impedence characteristics are of interest only to the loudspeaker's designers.
What your amplifier sees is the impedence curve for your loudspeaker system. The very same 8ohm driver in a closed cabinet might create an impedence of 40ohms at 50Hz, but in a bass-reflex the impendence at 50Hz might be 2.5ohms. Closed box cabinets create a high impendence at system resonance (Fs) off set by the resonance so in a well tuned system there is no significant increase or drop in perceieved output. Indeed there isn't in terms of energy transferred to the air as sound.
Bass reflex loudspeakers have an incredibly low impedence at the port tuning frequency, often accompanied by two peaks either side (in a well tuned system, in a bad one the peaks will be lop sided one way or another), which is typically also near fs. What happens is that the driver vibrates much less than if it was in a closed cabinet, but the air in the port resonates instead and takes over the audio output at those low frequencies. This is also the cause of the drop in impedence.
Pardon me Elfabyanos,
But this thread is nothing to do with listening environment. The only reason this even came into the thread was to determine if there is a reasonable correlation between the power delivered to a speaker and the acoustic output. That does not seem to be the case.
I'd suggest you start another thread on the subject of audio quality.
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Anyhoo, despite your loud assertion that we should "ignore this", if you would care to take a minute to review the thread and understand the context in which the reference was provided, you might appreciate that it was to confirm how much the impedance of a speaker can vary with frequency, and how little that has to do with the so called "impedance" value stamped on the speaker.
What you are pointing out is that the enclosure design also has a considerable effect on the impedance presented to the amplifier, and that an enclosure might make it even more variable with frequency than the reference suggests.
Therefore, I suggest, we are actually in violent agreement.
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Yeah I think so. Anyway, I didn't intend to take the thread OT, so to bring it back back I've dug out my college literature.
The explanation as to why low frequencies are hard to block - all sound that is transmitted through a wall is through re-radiation, requiring the wall to vibrate. The question is therefore why does the wall vibrate at lower frequencies as opposed to higher ones. All object tend to vibrate better at lower frequencies than higher ones, and this comes down to the physical properties of the medium. A solid object like a wall does not transmit high frequency sounds well for the same reason that it is a wall - i.e. it is solid and not compressible. If one imagines a wavelength that is twice the thickness of the wall, then to transmit such a sound the molecules on one surface of the wall would have to be traveling in precisely the opposite direction to the molecules on the surface on the other side.
The ability for a brick to have parts of it moving in opposite directions is low, indeed I would think that increase the level of such a sound wave high enough and the brick will just shatter. As the brick resists such movement the energy is instead reflected by the brick into the same room it came from.
Low frequency sounds have wavelengths much longer than the thickness of a wall, therefore the difference in velocities of the molecules in various parts of the brick is much smaller, the brick is therefore not resisting the movement as much, and less energy is reflected. Then the entire wall will accelerate as per mechanics as the force from the wave acts on the mass of the wall. The entire wall must have some give in it to allow it to vibrate, so that the entire wall bends in and out, and as such the middle of the wall which moves further will re-radiate more energy than at the corners where it is anchored to the other walls and rest of the building (in practice this would differ case by case, as buildings are not all as rigid as each other).
There is also a significant element somewhere in here about the velocity of the particles, as for a given audio energy the velocity increases with frequency. The higher speed has implications in terms of mechanics to explain why any medium transmits lower frequencies further than higher ones, as in losses through friction, but I don't know the maths well enough to back it up. Over the short distance of a wall thickness I think this effect would be negligible, but thats a hunch.
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Thanks Elfabyanos! I think that helps a lot.
Would it be correct to say that a wall can act as a sort of diaphragm at low frequencies, but it acts more like a reflector or energy absorber at high frequencies?
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Thanks Elfabyanos! I think that helps a lot.
Would it be correct to say that a wall can act as a sort of diaphragm at low frequencies, but it acts more like a reflector or energy absorber at high frequencies?
Something like a passive flat panel woofer?
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Geezer - yes I would say thats a fair description.
Maffsolo - something like that, but passive radiators are more complicated as they affect the main driver directly as well, unlike the wall which is remote. Passive radiators in speakers are acoustically coupled to the main driver by the air in the speaker cabinet. The passive radiator and the main driver then become one system and the combined impedences and resonances become very different, often in non-intuitive directions.