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Gear / Technical Help => Microphones & Setup => Topic started by: laptaper on September 12, 2006, 11:38:21 PM
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I'm listening right now to a master FOB cassette of the Dead on 7/2/89 at Foxboro Stadium, which someone asked me to digitize along with some of his other masters. It was taped using Sennheiser 421s and sounds better than the Schoeps source on archive.org. There's a lot of richness and warmth, but also definition and separation in the recording. Is this due more to the mike placement (it sounds like it was right in the Phil zone) or to the mikes themselves? What are the pros and cons of Senn 421's? I'm seeing some being sold on eBay and am tempted to click and buy right now. Thanks.
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from the Dead tapes I have heard, they sound kinda thin to my ears.
http://taperssection.com/index.php?topic=34004.0
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Thanks for the topic link. So the gist that I'm getting from the link you posted is that Senn 421s are generally considered bright with not a great bass response, but in an FOB or similarly high-bass environment they can definitely hold their own (not to mention their physical sturdiness). There was also mention of Healy's regular use of them. That makes sense then, because there is a definite Healyesque quality to this recording, a strong bass but also very clear highs.
Sometime within the next month I'll be seeding the recording in both 16 and 24 bit to shnflac.net. I'll post a link in this topic when I do, it's a source you really want to get.
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The original 421 is one of the best dynamic mics EVER MADE. IMO the design was light years ahead of anything else. The only problem with this mic is its no where near as sensitive as the mics we use now. It was designed for close miking. This dynamic mic like most is subject to proximity effect; the closer you are to the source the better the bass response will be. Now in a very high decibel level rock show with the right placement the low end might be very good, you would need a very loud sound pressure to get these mics to really sing.
The 421 used a phase plug in the center of the mic capsule it had a winding of wire this acted as a humbucking coil to eliminate or reduce EMI. It also helped form a wave guide that acted as a boost for ultra high end thus giving it a very crisp top end unlike most dynamic mics of its era. The humbucking coil they developed for use with dynamic mic coils is still in use today and was a huge factor in lowering the mics self noise.
They also used a Helmholtz resonator, this tuned cavity in the body of the 421 feed by a small tube from the back of the mic capsule helped lower the resonate frequency of the mic capsule it self thus improving its low end response. This mic also used an alnico 5 magnet structure. The 421 of today is nowhere near as good as they once were in fact they use neodymium magnets and have a weight inside the mic so it does not feel cheap. The size of the diaphragm on this mic was 1.5 inches! It’s a very big capsule. Frequency response was 28 Hz to 18 kHz and this mic handles very hi SPL of 148db!
There was also the Sennhieser 409 a very good mic. I would also check out the EV RE20/27 and Shure SM7 they are also very good dynamic mics. Ev was a real pioneer of dynamic mic designs. I often wonder why more people do not try dynamic mics for live recording of loud concerts. Back in the "day" there was little choice most condenser mics required huge power supply's and were not portable so they used mics like 421's.
I'm listening right now to a master FOB cassette of the Dead on 7/2/89 at Foxboro Stadium, which someone asked me to digitize along with some of his other masters. It was taped using Sennheiser 421s and sounds better than the Schoeps source on archive.org. There's a lot of richness and warmth, but also definition and separation in the recording. Is this due more to the mike placement (it sounds like it was right in the Phil zone) or to the mikes themselves? What are the pros and cons of Senn 421's? I'm seeing some being sold on eBay and am tempted to click and buy right now. Thanks.
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I'm afraid I don't understand some of the more technical aspects of your post - I'm still very much a newbie at this, I've never even owned a set of mikes. I do know that I want a pair that has at least a 24 khz response since I want to tape in 24 bit/48 khz. If the Sennheisers only go up to 18 then I suppose I need to look elsewhere. Thanks.
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Big and heavy!
Decent sounding mic that can handle the loudest of situations. But not so good if its quiet.
Don't confuse digital sample rates with microphone frequency response.
Yep just try and stealth with these I dare ya LOL they are huge and ugly and Moke is correct frequency response and sampling rates are not the same thing. Most mics only really do up to 18k that are anywhere near flat *cardioid mics that is* most good omni mics will go all the way up to 30k and be fairly flat.
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Okay, so as not to confuse them, what exactly is the difference between sample rate and frequency response? Doesn't an 18khz response mean that a mike is not going to record anything over 18khz? Or am I missing something here?
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I agree with Chris, that the original 421 was probably the best dynamic microphone ever manufactured. Look at any picture of the stage at a Grateful dead show from the 1980's. Damn near everything was miced with 421's, the kick drum, tom-toms, guitars, vocals...
421's also have a rotating collar that allows you to tailor a high pass filter.
These are great microphones for on stage or studio micing. There are much better options now for ambient recording.
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18khz is above the spectrum for human hearing. most people can't hear over 16khz, so, 18khz is fairly safe.
As far as sampling rates go,... when you mention 48khz, this is referring to the digital convertor gathering 16 bits, or, 24 bits of information, and sampling that lot 48,000 times per cycle before transmitting that digitized data to a bit-bucket (digi-recorder)..
I've generated 20 khz tones in Cool Edit, and although I've had to crank the volume on my stereo pretty loud I have been able to hear them. I'm probably going to invest in speakers and an amp that can put out more than the standard 20 khz most consumer speakers and receivers are rated at. I'm guessing my limit is going to be somewhere between 20-24 khz.
If I were to invest in a pair of omnis to get that 30 khz response, what would be a good pair to go with?
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If I were to invest in a pair of omnis to get that 30 khz response, what would be a good pair to go with?
earthworks
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I'm afraid I don't understand some of the more technical aspects of your post - I'm still very much a newbie at this, I've never even owned a set of mikes. I do know that I want a pair that has at least a 24 khz response since I want to tape in 24 bit/48 khz. If the Sennheisers only go up to 18 then I suppose I need to look elsewhere. Thanks.
Big and heavy!
Decent sounding mic that can handle the loudest of situations. But not so good if its quiet.
Don't confuse digital sample rates with microphone frequency response.
Moke,
Seems like he is referring to Nyquist. If you are taping @ 48KHz, makes sense that you would want a mic that can record frequencies up to half that, or 24 KHz. Digital sampling rate is directly related to the freq. response contained in your recording.
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Okay, so as not to confuse them, what exactly is the difference between sample rate and frequency response? Doesn't an 18khz response mean that a mike is not going to record anything over 18khz? Or am I missing something here?
Don't make the mistake of getting elite-ist in looking at microphone specifications. Specs don't tell the whole story of how a microphone will sound. Find recordings you like and investigate what microphones were used to make that recording. As far as 30k microphones... Earthworks makes some, but you may not like how the sound for recording concerts, because they have an un-hyped sound.
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If I were to invest in a pair of omnis to get that 30 khz response, what would be a good pair to go with?
earthworks
and that's about it too.
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Okay, so as not to confuse them, what exactly is the difference between sample rate and frequency response? Doesn't an 18khz response mean that a mike is not going to record anything over 18khz? Or am I missing something here?
Don't make the mistake of getting elite-ist in looking at microphone specifications. Specs don't tell the whole story of how a microphone will sound. Find recordings you like and investigate what microphones were used to make that recording. As far as 30k microphones... Earthworks makes some, but you may not like how the sound for recording concerts, because they have an un-hyped sound.
I think wbrisette on here has made lots of earthworks tapes, maybe someone more knowledgeable than me can point out some places to d/l them. Chuck is right, they are supposed to be very neutral mics (which I personally like).
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Okay, so as not to confuse them, what exactly is the difference between sample rate and frequency response? Doesn't an 18khz response mean that a mike is not going to record anything over 18khz? Or am I missing something here?
Don't make the mistake of getting elite-ist in looking at microphone specifications. Specs don't tell the whole story of how a microphone will sound. Find recordings you like and investigate what microphones were used to make that recording. As far as 30k microphones... Earthworks makes some, but you may not like how the sound for recording concerts, because they have an un-hyped sound.
I think wbrisette on here has made lots of earthworks tapes, maybe someone more knowledgeable than me can point out some places to d/l them. Chuck is right, they are supposed to be very neutral mics (which I personally like).
BeckyT made some Earthworks recordings too.
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Only bad thing about Earthworks is that they are more power hungry than Bush and Cheney :P
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I've always maintained, and continuosly so, that I'm not qualified enough in any of this to call myself a professional.
WTF is Nyquist?
a principle in digital audio, stating that the highest throughput frequency in a sampling system must be equal to or less than one-half the sampling frequency. so theoretically, the highest frequency reproducible (without generating aliasing noise) by a system operating at a sampling rate of 48 kHz is 24 kHz. In practice, with the need for brickwall anti-aliasing filters, the actual upper-frequency response of such a system is somewhat less.
as for omnis...id go with Gefell mk221 with josephson c617 bodies. cream o' the crop.
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I'm afraid I don't understand some of the more technical aspects of your post - I'm still very much a newbie at this, I've never even owned a set of mikes. I do know that I want a pair that has at least a 24 khz response since I want to tape in 24 bit/48 khz. If the Sennheisers only go up to 18 then I suppose I need to look elsewhere. Thanks.
Big and heavy!
Decent sounding mic that can handle the loudest of situations. But not so good if its quiet.
Don't confuse digital sample rates with microphone frequency response.
Moke,
Seems like he is referring to Nyquist. If you are taping @ 48KHz, makes sense that you would want a mic that can record frequencies up to half that, or 24 KHz. Digital sampling rate is directly related to the freq. response contained in your recording.
Oh come on. You mean if you want to do 24/96 that you want a mic that goes to 48 kHz?
In my opinion, it's actually a good thing to have mics that roll off in the ultrasonic range. Otherwise you end up consuming a portion of your available dynamic range recording signals that you can't hear anyway.
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Oh come on. You mean if you want to do 24/96 that you want a mic that goes to 48 kHz?
I would think so. That seems to be the reason behind the wide-bandwidth Earthworks and Schoeps CMC6xt microphones, to take advantage of high sampling rate digital formats.
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you've got dog ears!
There is this new technology for crowd control, a 15khz buzz, that is going to lay waste to you and your ears! Most "older people" can't hear it. But the younger ages, the targeted ages, will suffer enough to disperse when it is applied.
Not really, I just take better care of my ears. :-) I found out about ten years ago that a small wad of paper stuffed in the lower ear lobe will cut out what I call the "stilleto frequencies" at a concert, while still allowing you to fully enjoy the show and hold a normal conversation without shouting.
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If you're recording rock concerts, keep in mind that instruments, vocals and all that other stuff coming out of the PA cabinets are probably being miked with 57's and 58's (freq. response is 50-15k I believe?) Then there's the response of the cabinets themselves...
Unless you're recording unamplified music in a perfect room, mics like the Earthworks (30K, 40k, etc) are just going to give you a more detailed picture of all the crap you DON'T want.
Maybe I'm too cynical, but I have a hard time believing you can actually hear a true 20kHz tone. If you really can, maybe you should contact Bob Clearmountain about a possible internship.
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Maybe I'm too cynical, but I have a hard time believing you can actually hear a true 20kHz tone. If you really can, maybe you should contact Bob Clearmountain about a possible internship.
qft
Well, all I know is that if I go into CE 2K and select from the menu Generate>Tones, then type in "20000" in the Base Frequency box (right next to where it says "Hz") I can hear a tiny but distinct tone if I crank my stereo all the way up and there's no ambient sound interfering. It's hard to pick out, but I can.
I experimented with 15khz, too, and pity the teenagers (and myself - I'm 42, and if this is what I'll be encountering in certain public places I'll gladly lose that capacity, I came close to nausea when I turned it up). Also, I've noticed in train stations I'll literally cringe and cover my ears if a locomotive with bad brakes comes screeching into the station while others just saunter on their merry way.
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Maybe I'm too cynical, but I have a hard time believing you can actually hear a true 20kHz tone. If you really can, maybe you should contact Bob Clearmountain about a possible internship.
qft
Hmmmmm....
I just ABX'd two files: a 10-sec silent file and a 10-sec 20 kHz tone file. My ABX results: 10/10. It's faint, not easy to hear. But the chances of me guessing correctly 10/10 times (according to the ABX s/w) is < 0.1%.
That said, maybe what I'm hearing is my headphones' attempt to reproduce the 20 kHz tone, but it's actually generating other noise / signal in the process that I'm picking up.
Or maybe my quick test is fundamentally flawed (likely). Hmmmm...
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Maybe I'm too cynical, but I have a hard time believing you can actually hear a true 20kHz tone. If you really can, maybe you should contact Bob Clearmountain about a possible internship.
qft
Hmmmmm....
I just ABX'd two files: a 10-sec silent file and a 10-sec 20 kHz tone file. My ABX results: 10/10. It's faint, not easy to hear. But the chances of me guessing correctly 10/10 times (according to the ABX s/w) is < 0.1%.
That said, maybe what I'm hearing is my headphones' attempt to reproduce the 20 kHz tone, but it's actually generating other noise / signal in the process that I'm picking up.
Or maybe my quick test is fundamentally flawed (likely). Hmmmm...
I would also guess that it is much easier to hear a pure 20 KHz tone than 20 KHz tones that are intermixed with music.
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I think that info is in the Microphone book!
Kavi is all digital now! Sound Devices 722>>>true systems>>>PearlI've always maintained, and continuosly so, that I'm not qualified enough in any of this to call myself a professional.
WTF is Nyquist?
a principle in digital audio, stating that the highest throughput frequency in a sampling system must be equal to or less than one-half the sampling frequency. so theoretically, the highest frequency reproducible (without generating aliasing noise) by a system operating at a sampling rate of 48 kHz is 24 kHz. In practice, with the need for brickwall anti-aliasing filters, the actual upper-frequency response of such a system is somewhat less.
as for omnis...id go with Gefell mk221 with josephson c617 bodies. cream o' the crop.
Thanks, Teddy
I'll bet you didn't learn that from Kavi! ;D
:goes and hugs his d5m:
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too bad the schoeps XTs sound like ass. unless you are a dog , the wide bandwith microphones are worthless, imho.
Oh come on. You mean if you want to do 24/96 that you want a mic that goes to 48 kHz?
I would think so. That seems to be the reason behind the wide-bandwidth Earthworks and Schoeps CMC6xt microphones, to take advantage of high sampling rate digital formats.
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20k on computer speakers?
we need a review of these, if so!
my harman/kardon sound sticks just sort of clicked as the sample started, and then again as the sample stopped.
WHAT?
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Mics hand-held and performance recorded from 9th row center.
It was incredibly painful!
+T for the bulging bicep workout.
IMO, Like others have mentioned this is pointless for recoring anything that goes through a PA. Even when recording acoustic sources there is something to be said for a gradual smooth roll of of the uber high freq. - even with higher sampling rates. I've heard string quartets and jazz drummer's cymbals with plenty of energy up there recorded with ribbon mics that blew me away with the natural sound of the top octave, & ribbon mics are generally way down by 16-18k. I've also been impressed with some very natural sounding Earthworks recordings that had killer airy transients and response far above what I can hear. but for my money, a smooth response is more important than ultimate extension.
Think of it as leaving a little frequency headroom if it makes you feel better about unused bandwidth up there.
Keep in mind that there are always trade off's. Capturing those dog eared frequencies requires mics with very small diaphrams that are generally noisier. Which, like everything else can be somewhat offset with $$$$. Just make sure the trade off's you make are for better sounding, more involving music and not just specs on paper.
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Well, all I know is that if I go into CE 2K and select from the menu Generate>Tones, then type in "20000" in the Base Frequency box (right next to where it says "Hz") I can hear a tiny but distinct tone if I crank my stereo all the way up and there's no ambient sound interfering. It's hard to pick out, but I can.
I've got an experiment for you. Turn your 20 kHz recording up until you can hear it, then pop in a 8 kHz tone instead at the same volume. (Then think about the damage you're doing to your ears with the 20 kHz that you can barely detect and contrast that with the 8 kHz tone that probably has you covering your ears with your hands. :o)
Like I said, I don't understand using part of your valuable dynamic range to preserve sonic information that is so nearly indetectable when compared with the audible portion of the audio spectrum.
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I overlooked something else - regardless of what frequencies I can (barely) hear at full blast, I need to take into account the volume I'm actually going to be listening to my recordings at, which obviously won't be full volume. Given that, should I still be taping at 48khz for the headroom, or is 44.1khz sufficient?
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Like I said, I don't understand using part of your valuable dynamic range to preserve sonic information that is so nearly indetectable when compared with the audible portion of the audio spectrum.
I have a feeling that you are taking up very little dynamic range by adding the ultrasonics, in my experience recording levels tend to be dominated by bass freqs. What do y'all think?
But I think you are definitely right that the importance of this info up in the freq. response stratosphere is pretty insignificant when compared to stuff that is more in the human hearing range.
WRT sampling freq, I always roll at 44.1, just for convenience in burning to CD audio.
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I overlooked something else - regardless of what frequencies I can (barely) hear at full blast, I need to take into account the volume I'm actually going to be listening to my recordings at, which obviously won't be full volume. Given that, should I still be taping at 48khz for the headroom, or is 44.1khz sufficient?
24/44.1 is more than adequate. Especially since the delivery medium will be CD..no SRC-==less processing. If you had a stellar listening system or were recording acoustic music in great halls...i could see the need for higher sampling rates, but with a PA enviroment, the audio equivalent of a trailer park, what is the point of capturing more crappy room?
teddy
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I'm listening right now to a master FOB cassette of the Dead on 7/2/89 at Foxboro Stadium, which someone asked me to digitize along with some of his other masters. It was taped using Sennheiser 421s and sounds better than the Schoeps source on archive.org. There's a lot of richness and warmth, but also definition and separation in the recording.
This GD 7-2-89 Schoeps source is one of the reasons I had the dream of getting a pair of Schoeps. It is also a prominent reason why I got back into taping! One of the better split omni recordings I have ever heard, very life-like feel from the floor (although in the first three minutes of 'Playin', the levels are too low). I have heard amazing 441 tapes (12-30-83 SF Civic) from the Phil Zone, but the 421s I have heard, never had the sparkle or the transient detail like the 441s.
Man, if this 421 source is better than Gaswirt's split cmc42 > Oade m118 > PCM source on archive, I wanna hear it! Anytime this is gonna be torrented soon? It is a great show to boot!
I say just go 16/44.1 and get the show out there if it is as stellar as you say it is!
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;) no problem , Moke!
the audio equivalent of a trailer park
some days are just so worth waking up for!
thanks for brightening my day. :bigsmile: :lol:
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Like I said, I don't understand using part of your valuable dynamic range to preserve sonic information that is so nearly indetectable when compared with the audible portion of the audio spectrum.
The reason to try and capture above 16k?
One word
HARMONICS.
That is why we should always try to capture the full bandwidth of the source.
Chris Church
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The reason to try and capture above 16k?
One word
HARMONICS.
That is why we should always try to capture the full bandwidth of the source.
Chris Church
I prefer to capture only the AUDIBLE harmonics so that I don't waste bits on inaudible portions of the audio spectrum.
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The reason to try and capture above 16k?
One word
HARMONICS.
That is why we should always try to capture the full bandwidth of the source.
Chris Church
I prefer to capture only the AUDIBLE harmonics so that I don't waste bits on inaudible portions of the audio spectrum.
The funny thing about harmonics is that even the ones you do not hear effect the ones you do. So you need to capture it all DAMIT :)
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I'm listening right now to a master FOB cassette of the Dead on 7/2/89 at Foxboro Stadium, which someone asked me to digitize along with some of his other masters. It was taped using Sennheiser 421s and sounds better than the Schoeps source on archive.org. There's a lot of richness and warmth, but also definition and separation in the recording.
Man, if this 421 source is better than Gaswirt's split cmc42 > Oade m118 > PCM source on archive, I wanna hear it! Anytime this is gonna be torrented soon? It is a great show to boot!
I say just go 16/44.1 and get the show out there if it is as stellar as you say it is!
Damn the PCMs, full 24 bits ahead! Trust me, it'll be worth waiting for. ;D Keep checking shnflac.net, I'm putting a new Tim Alexander source up there each night, 16 bit first, 24 the next night. It'll be up sometime within the next month.
As far as harmonics and sample rate are concerned, if I'm correct by harmonics you mean that by upping the samples per second you're creating more room for extra tones to squeeze in, regardless of frequency, creating a fuller sound. E.g. if a recording were a digitized picture, bitrate would be analogous to the potential number of shades a given pixel could have and sample rate would be the number of pixels per square inch. Have I got that right?
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The funny thing about harmonics is that even the ones you do not hear effect the ones you do. So you need to capture it all DAMIT :)
What's that? You say you want to try another ABX comparison? >:D
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The funny thing about harmonics is that even the ones you do not hear effect the ones you do. So you need to capture it all DAMIT :)
What's that? You say you want to try another ABX comparison? >:D
OK sure lets do it, Two tracks ( the same song) One with a low pass at 20k 96db per octave, the other one with out any lowpass recorded at 24bit 192k Does that sound like a fair test?
Let me know if you have trouble getting a hold of a 96db per octave lowpass I can do it here and upload the files already done.
Let me know if you can find a host for the files.
Chris Church
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"You can bring a bowl of popcorn and a glass of water, too." ;D
Cream, "Anyone for Tennis"
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The funny thing about harmonics is that even the ones you do not hear effect the ones you do. So you need to capture it all DAMIT :)
What's that? You say you want to try another ABX comparison? >:D
OK sure lets do it, Two tracks ( the same song) One with a low pass at 20k 96db per octave, the other one with out any lowpass recorded at 24bit 192k Does that sound like a fair test?
Let me know if you have trouble getting a hold of a 96db per octave lowpass I can do it here and upload the files already done.
Let me know if you can find a host for the files.
Chris Church
OK, you do that. I'll be interested to see how you propose to get 96 dB per octave past 20 kHz. How far down do you plan to be at 20 kHz? Wait... don't tell me... You want it to be flat to within 1 dB out to 20 kHz and be -96 dB by 40 kHz.
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OK sure lets do it, Two tracks ( the same song) One with a low pass at 20k 96db per octave, the other one with out any lowpass recorded at 24bit 192k Does that sound like a fair test?
That is one steep slope!!!! :o
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The funny thing about harmonics is that even the ones you do not hear effect the ones you do. So you need to capture it all DAMIT :)
What's that? You say you want to try another ABX comparison? >:D
OK sure lets do it, Two tracks ( the same song) One with a low pass at 20k 96db per octave, the other one with out any lowpass recorded at 24bit 192k Does that sound like a fair test?
Let me know if you have trouble getting a hold of a 96db per octave lowpass I can do it here and upload the files already done.
Let me know if you can find a host for the files.
Chris Church
OK, you do that. I'll be interested to see how you propose to get 96 dB per octave past 20 kHz. How far down do you plan to be at 20 kHz? Wait... don't tell me... You want it to be flat to within 1 dB out to 20 kHz and be -96 dB by 40 kHz.
There are many ways to do it but I can use a digital crossover to get the slope but it got me thinking I think I have to be carefull about this because if I use a digital crossover it will change the sound of the processed signal and make the two samples less like each other. Any ideas out there?
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There are many ways to do it but I can use a digital crossover to get the slope but it got me thinking I think I have to be carefull about this because if I use a digital crossover it will change the sound of the processed signal and make the two samples less like each other. Any ideas out there?
Start with two copies of a harmonically rich 24bit 192khz source that actually has content over 20khz (varify that with a spectral analysis). Low pass one at 20k and the other at 80k, or 96 or whatever, just higher than the other. Check the two low passed copies in a spectral analyzer to varify the presence of ultrasonics and the roll off. ABX the two 24/192 files that have been through the same processing sequence and should differ only in their cutoff frequency.
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thats an awful lot of effort for things that you'll never be able to hear.
Have fun guys.
I could say the same thing about people that place alarm clocks, and french braid pretty little cables and sell them for $900 but I wont go there. It would be nice to live in a world where snake oil audio solutions were not sold to unsuspecting buyers, who think good sound comes from an alarm clock but what do I know. :)
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My clock radio rocks almost all the way up to 10khz!
It's only worth 10 bucks, but I do leave it plugged in while I listen to the stereo in the other room. ::)
You should hear my microwave..
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Here's a fun site, and apparently this is the new thing with the kidz...?
http://www.ultrasonic-ringtones.com/ (http://www.ultrasonic-ringtones.com/)
So, what's the highest tone you can hear? Be honest now!
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Here's a fun site, and apparently this is the new thing with the kidz...?
http://www.ultrasonic-ringtones.com/ (http://www.ultrasonic-ringtones.com/)
So, what's the highest tone you can hear? Be honest now!
I started highest (22.4 kHz) and couldn't hear it, and dropped down from there. I heard the 2nd highest, 21.1 kHz. I kept going down, and thought the 18.8 kHz sample sounded odd, like it was too low in frequency. So I downloaded a handful of them (16.7 - 22.4 kHz) and checked 'em out in Adobe Audition via its frequency analyzer. Well...the 22.4 kHz file...nothing! Flatline waveform, empty frequency analysis result set.
Anyway, as for the others, though I heard the 2nd highest at 21.1 kHz, I'm not sure whether what I'm hearing is the target frequency or other low-level artifacts as a result of the MP3 encoding process. Hmmmmm... :hmmm:
I think I'm gonna try out this hearing test for kicks:
http://www.digital-recordings.com/hearing-test/www-ht-pro/ht_help_p.html
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The funny thing about harmonics is that even the ones you do not hear effect the ones you do. So you need to capture it all DAMIT :)
What's that? You say you want to try another ABX comparison? >:D
OK sure lets do it, Two tracks ( the same song) One with a low pass at 20k 96db per octave, the other one with out any lowpass recorded at 24bit 192k Does that sound like a fair test?
Let me know if you have trouble getting a hold of a 96db per octave lowpass I can do it here and upload the files already done.
Let me know if you can find a host for the files.
Chris Church
OK, you do that. I'll be interested to see how you propose to get 96 dB per octave past 20 kHz. How far down do you plan to be at 20 kHz? Wait... don't tell me... You want it to be flat to within 1 dB out to 20 kHz and be -96 dB by 40 kHz.
There are many ways to do it but I can use a digital crossover to get the slope but it got me thinking I think I have to be carefull about this because if I use a digital crossover it will change the sound of the processed signal and make the two samples less like each other. Any ideas out there?
Chris, let me put this in perspective for you. You get about 6 dB per octave rolloff in the stopband of a lowpass filter per pole in the transfer function for the filter. With a single pole, you'll be down about 3 dB at the cutoff frequency. When you go to multiple poles, you generally do not put all of your poles at the same frequency, but if you did, you'd get about 3 dB attenuation per pole at the cutoff frequency. By the time you get 96 (!!) dB per octave, you're going to have 16 poles in the filter. If all the poles are co-located, then you'll have 48 dB of attenuation at the cutoff frequency. Something tells me that filtered and unfiltered versions will be easily recognizable if you have 48 dB of attenuation at the intended cutoff frequency. In order to get any sort of flatness in the passband and still get a steep stopband attenuation curve, you'll have to resort to Chebyschev or elliptical designs. With at least 16 poles in the transfer function, some of the poles will be extremely high Q poles which means your parts tolerances are going to be obscenely critical. In order to get an analog filter that was flat to 1 dB in the passband and had 96 dB attenuation one octave into the stop band, you'll probably spend the rest of your life designing, re-designing and tweeking and still not get the job done.
In my opinion, the better approach would be to use a digital domain filter to boost the high frequency content of the signal from a microphone whose transfer function is well known and to do so in such a way that the resulting transfer function is flat out to the highest frequency for which you have calibration data for your mic. So if you have 10 dB of rolloff in your mic's frequency response at 25kHz, that portion of the spectrum should be boosted by 10 dB. The transfer function of your digital filter should ideally be the reciprocal of the complex conjugate of the frequency response of your microphone. This will increase the quantization noise in the ultrasonic portion of the spectrum, but it's the only way I know of the simulate what a truly flat broadband microphone would sound like, at least with any reasonable amount of effort. I'd offer to design a filter like this, but I no longer have access to DSP filter design tools and I don't feel like spending enough time to do it by hand when I know darn well that the only result is to modify signals that I can't hear anyway.
I'd also like to point out that it is not ideal to simply correct the amplitude in the stopband. It would be better to flatten the spectrum by using a filter whose transfer function is the reciprocal of the complex conjugate of the frequency response of the device whose output you wish to flatten. By using the reciprocal of the complex conjugate of the frequency of the mic used to do the recording, you not only correct the amplitude response, but you also get as close as possible to a linear phase (constant delay) response. If you only correct the amplitude response then there may be artifacts from the uncompensated phase distortion that can appear as subharmonics within the audible portion of the spectrum and, as you should expect, you can probably hear arifacts if they are induced within the audible portion of the audio spectrum.
If you really want to do this, I'd recommend using FFT processing to transform the time domain waveform into the frequency domain, apply the required amplitude and phase equalization and inverse transform the result to get back to the time domain. This will be an extremely processor intensive process, but it should yield relevant results.
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Here's a fun site, and apparently this is the new thing with the kidz...?
http://www.ultrasonic-ringtones.com/ (http://www.ultrasonic-ringtones.com/)
So, what's the highest tone you can hear? Be honest now!
In addition to the issues Brian pointed out...
How can the crappy speaker in a cell phone be expected to produce these ultrasonic frequencies?
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In addition to the issues Brian pointed out...
How can the crappy speaker in a cell phone be expected to produce these ultrasonic frequencies?
What driver does a cell phone use? Probably something like a tiny piezo transducer. No problem making ultasonic squeals with something like a 50 cent Radio Shack piezo buzzer.
That's not to say it sounds any good.. or cleanly reproduces those tones without all kinds of distortion artifacts.
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Here's a fun site, and apparently this is the new thing with the kidz...?
http://www.ultrasonic-ringtones.com/ (http://www.ultrasonic-ringtones.com/)
So, what's the highest tone you can hear? Be honest now!
I started highest (22.4 kHz) and couldn't hear it, and dropped down from there. I heard the 2nd highest, 21.1 kHz. I kept going down, and thought the 18.8 kHz sample sounded odd, like it was too low in frequency. So I downloaded a handful of them (16.7 - 22.4 kHz) and checked 'em out in Adobe Audition via its frequency analyzer. Well...the 22.4 kHz file...nothing! Flatline waveform, empty frequency analysis result set.
Anyway, as for the others, though I heard the 2nd highest at 21.1 kHz, I'm not sure whether what I'm hearing is the target frequency or other low-level artifacts as a result of the MP3 encoding process. Hmmmmm... :hmmm:
I think I'm gonna try out this hearing test for kicks:
http://www.digital-recordings.com/hearing-test/www-ht-pro/ht_help_p.html
Well according to the test then, you are a dog not a human. I can hear 16.7 easily. I thought I could hear the 18.8 but I wasn't sure for the same reasons as you. I'm sure there is mp3 artifacting in these files anyway. Plus right now I'm listening on crappy headphones plugged into my laptop.
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heres another, Brian http://www.phys.unsw.edu.au/~jw/hearing.html
Here's a fun site, and apparently this is the new thing with the kidz...?
http://www.ultrasonic-ringtones.com/ (http://www.ultrasonic-ringtones.com/)
So, what's the highest tone you can hear? Be honest now!
I started highest (22.4 kHz) and couldn't hear it, and dropped down from there. I heard the 2nd highest, 21.1 kHz. I kept going down, and thought the 18.8 kHz sample sounded odd, like it was too low in frequency. So I downloaded a handful of them (16.7 - 22.4 kHz) and checked 'em out in Adobe Audition via its frequency analyzer. Well...the 22.4 kHz file...nothing! Flatline waveform, empty frequency analysis result set.
Anyway, as for the others, though I heard the 2nd highest at 21.1 kHz, I'm not sure whether what I'm hearing is the target frequency or other low-level artifacts as a result of the MP3 encoding process. Hmmmmm... :hmmm:
I think I'm gonna try out this hearing test for kicks:
http://www.digital-recordings.com/hearing-test/www-ht-pro/ht_help_p.html
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Chris, let me put this in perspective for you.....
.....If you really want to do this, I'd recommend using FFT processing to transform the time domain waveform into the frequency domain, apply the required amplitude and phase equalization and inverse transform the result to get back to the time domain. This will be an extremely processor intensive process, but it should yield relevant results.
SparkE!,
This is way over my head, but it occurs to me that the low pass filtering done by the typical A/D stage is probably not taking anywhere near the optimized approach you've outlined. Could the percieved differences people hear between 44.1khz sampled audio and higher sampling rates be due to artifacts related to non-ideal real world low pass filters? From your discourse it sounds to me like an ideal steep-enough filter gets very complex when the 22khz absolute frequency limit of 44.1khz sampled audio is not so very far above what most of us can hear. I'd guess the same problems arise with a higher rate but just manifest higher in frequency so the artifacts aren't noticed.
I guess what I'm trying to say is maybe people aren't actually hearing those ultra high harmonics, but are hearing artifacts of the lowpass filter.
just grasping..
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Chris, let me put this in perspective for you.....
.....If you really want to do this, I'd recommend using FFT processing to transform the time domain waveform into the frequency domain, apply the required amplitude and phase equalization and inverse transform the result to get back to the time domain. This will be an extremely processor intensive process, but it should yield relevant results.
SparkE!,
This is way over my head, but it occurs to me that the low pass filtering done by the typical A/D stage is probably not taking anywhere near the optimized approach you've outlined. Could the percieved differences people hear between 44.1khz sampled audio and higher sampling rates be due to artifacts related to non-ideal real world low pass filters? From your discourse it sounds to me like an ideal steep-enough filter gets very complex when the 22khz absolute frequency limit of 44.1khz sampled audio is not so very far above what most of us can hear. I'd guess the same problems arise with a higher rate but just manifest higher in frequency so the artifacts aren't noticed.
I guess what I'm trying to say is maybe people aren't actually hearing those ultra high harmonics, but are hearing artifacts of the lowpass filter.
just grasping..
Yeah, when you're talking about a digitial filter, there are all sorts of audible artifacts that can be introduced if you're not careful, but you can also build a much more agressive filter. It's just math. If you use a real analog filter, then you generally won't get audible artifacts unless something breaks into non-linear operation (like a spurious oscillation), but you can't be as agressive in removing undesired frequency components.
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Thanks for the hearing test links Teddy, I'll have to try this on the phones after I get home.
At least there's few hours left to cram before the test..:-\
Maybe we should post sticky links to these somewhere, like in the other websites section of the board, in the name of protecting everyones hearing.
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Yeah, when you're talking about a digitial filter, there are all sorts of audible artifacts that can be introduced if you're not careful, but you can also build a much more agressive filter. It's just math. If you use a real analog filter, then you generally won't get audible artifacts unless something breaks into non-linear operation (like a spurious oscillation), but you can't be as agressive in removing undesired frequency components.
So when an analog signal is digitized at 44.1khz it first goes through an analog lowpass filter, as I understand it. What slope is typically used? or where does the rolloff start if the signal must be way down by 22khz?
And if a digital file is resampled to 44.1khz or lowpassed to remove everything over 22khz, then the steep slope required to keep a relatively flat response up to 20khz is likely to produce audible artifacts when using simple digital techniques.
Do I have it right?
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Yeah, when you're talking about a digitial filter, there are all sorts of audible artifacts that can be introduced if you're not careful, but you can also build a much more agressive filter. It's just math. If you use a real analog filter, then you generally won't get audible artifacts unless something breaks into non-linear operation (like a spurious oscillation), but you can't be as agressive in removing undesired frequency components.
So when an analog signal is digitized at 44.1khz it first goes through an analog lowpass filter, as I understand it. What slope is typically used? or where does the rolloff start if the signal must be way down by 22khz?
And if a digital file is resampled to 44.1khz or lowpassed to remove everything over 22khz, then the steep slope required to keep a relatively flat response up to 20khz is likely to produce audible artifacts when using simple digital techniques.
Do I have it right?
It's not common to use more than about an 8 pole filter for an anti-aliasing filter ahead of the A/D. If it's flat to 20 kHz, then you're not going to be very far down at 22 kHz, so aliasing happens. That's all the more reason to justify putting your band edge lower, perhaps at 16 kHz. That way, you get more attenuation by 22 kHz. For anti-aliasing, you HAVE to use a real filter. Common digital filtering techniques will not work to avoid audible aliasing if you are using a 44.1 kHz sample rate. However if you are using 88.2 kHz or 96 kHz, then your anti-aliasing filter only has to be sufficiently effective by the time it gets to 44.1 kHz or 48 kHz, respectively. You should be able to get 50 or 60 dB of attenuation of products above the 1/2 sample rate frequency if you use one of the higher sample rates. Then you can actually use digital techniques to remove inaudible portions of the signal between the cutoff frequency of your anti-aliasing filter and your 1/2 sample frequency. Of course, it depends if you really want to do that. Some people (like Chris Church) think it's a bad idea to remove the portion of the signal spectrum that is inaudible and that's really the essence of our disagreement. Can you hear it if inaudible information is removed from a signal? I claim that you can't. Chris claims that it makes a difference because somehow the inaudible portion of the signal makes the audible portion more real sounding. He could be right, but I seriously doubt it. If someone comes up with a reasonable procedure to create files to compare without introducing audible artifacts, then I guess we'll see. One file will need to have ultrasonic information in it. The other file should be exactly the same, except missing the ultrasonic information or at least it should not have as much ultrasonic information in it. I doubt that this will be an easy test to construct.
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Thanks for the well worded explaination. +T
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Yeah, when you're talking about a digitial filter, there are all sorts of audible artifacts that can be introduced if you're not careful, but you can also build a much more agressive filter. It's just math. If you use a real analog filter, then you generally won't get audible artifacts unless something breaks into non-linear operation (like a spurious oscillation), but you can't be as agressive in removing undesired frequency components.
So when an analog signal is digitized at 44.1khz it first goes through an analog lowpass filter, as I understand it. What slope is typically used? or where does the rolloff start if the signal must be way down by 22khz?
And if a digital file is resampled to 44.1khz or lowpassed to remove everything over 22khz, then the steep slope required to keep a relatively flat response up to 20khz is likely to produce audible artifacts when using simple digital techniques.
Do I have it right?
It's not common to use more than about an 8 pole filter for an anti-aliasing filter ahead of the A/D. If it's flat to 20 kHz, then you're not going to be very far down at 22 kHz, so aliasing happens. That's all the more reason to justify putting your band edge lower, perhaps at 16 kHz. That way, you get more attenuation by 22 kHz. For anti-aliasing, you HAVE to use a real filter. Common digital filtering techniques will not work to avoid audible aliasing if you are using a 44.1 kHz sample rate. However if you are using 88.2 kHz or 96 kHz, then your anti-aliasing filter only has to be sufficiently effective by the time it gets to 44.1 kHz or 48 kHz, respectively. You should be able to get 50 or 60 dB of attenuation of products above the 1/2 sample rate frequency if you use one of the higher sample rates. Then you can actually use digital techniques to remove inaudible portions of the signal between the cutoff frequency of your anti-aliasing filter and your 1/2 sample frequency. Of course, it depends if you really want to do that. Some people (like Chris Church) think it's a bad idea to remove the portion of the signal spectrum that is inaudible and that's really the essence of our disagreement. Can you hear it if inaudible information is removed from a signal? I claim that you can't. Chris claims that it makes a difference because somehow the inaudible portion of the signal makes the audible portion more real sounding. He could be right, but I seriously doubt it. If someone comes up with a reasonable procedure to create files to compare without introducing audible artifacts, then I guess we'll see. One file will need to have ultrasonic information in it. The other file should be exactly the same, except missing the ultrasonic information or at least it should not have as much ultrasonic information in it. I doubt that this will be an easy test to construct.
Here is a MLS sample flat from 20hz to 90khz with a 20hz LPF butterworth 7th order with a band pass ripple of 0.5
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Yeah, when you're talking about a digitial filter, there are all sorts of audible artifacts that can be introduced if you're not careful, but you can also build a much more agressive filter. It's just math. If you use a real analog filter, then you generally won't get audible artifacts unless something breaks into non-linear operation (like a spurious oscillation), but you can't be as agressive in removing undesired frequency components.
So when an analog signal is digitized at 44.1khz it first goes through an analog lowpass filter, as I understand it. What slope is typically used? or where does the rolloff start if the signal must be way down by 22khz?
And if a digital file is resampled to 44.1khz or lowpassed to remove everything over 22khz, then the steep slope required to keep a relatively flat response up to 20khz is likely to produce audible artifacts when using simple digital techniques.
Do I have it right?
It's not common to use more than about an 8 pole filter for an anti-aliasing filter ahead of the A/D. If it's flat to 20 kHz, then you're not going to be very far down at 22 kHz, so aliasing happens. That's all the more reason to justify putting your band edge lower, perhaps at 16 kHz. That way, you get more attenuation by 22 kHz. For anti-aliasing, you HAVE to use a real filter. Common digital filtering techniques will not work to avoid audible aliasing if you are using a 44.1 kHz sample rate. However if you are using 88.2 kHz or 96 kHz, then your anti-aliasing filter only has to be sufficiently effective by the time it gets to 44.1 kHz or 48 kHz, respectively. You should be able to get 50 or 60 dB of attenuation of products above the 1/2 sample rate frequency if you use one of the higher sample rates. Then you can actually use digital techniques to remove inaudible portions of the signal between the cutoff frequency of your anti-aliasing filter and your 1/2 sample frequency. Of course, it depends if you really want to do that. Some people (like Chris Church) think it's a bad idea to remove the portion of the signal spectrum that is inaudible and that's really the essence of our disagreement. Can you hear it if inaudible information is removed from a signal? I claim that you can't. Chris claims that it makes a difference because somehow the inaudible portion of the signal makes the audible portion more real sounding. He could be right, but I seriously doubt it. If someone comes up with a reasonable procedure to create files to compare without introducing audible artifacts, then I guess we'll see. One file will need to have ultrasonic information in it. The other file should be exactly the same, except missing the ultrasonic information or at least it should not have as much ultrasonic information in it. I doubt that this will be an easy test to construct.
So if we believe what you’re saying then sympathetic resonation does not exist! I believe that ultra sonic information from say an instrument like a horn can go up to 40k, there are sympathetic resonations that are above 20k that will effect what we hear in the audible range. Why should we limit our selves to 20k? When it’s a fact that many instruments produce sound well above that point, how can you say that the information above 20k does not effect the information we hear below it????
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A justifiable reason to use higher sampling rates, say above 44.1kHz, is to have more natural high end extension due to the neccesity of having the anti-aliasing filter. I think that if equipment is engineered with high fidelity in mind, the limits imposed by the filtering can be overcome.
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...real world here...
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dan lavry says it best.
http://www.lavryengineering.com/documents/Sampling_Theory.pdf
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^^ Go Dan! That's a convincing argument.
Chris,
Your proposing that two ultasonic tones will amplitude modulate to create a 'beat' or third tone in the audible range? Similar to the rythmic 'beat' between two close but not quite matching frequencies in the audible range; ie. the 'beat' sound that slows down as two guitar strings are brought into tune, extrapolated up until the 'beat' is in the audible range of frequencies?
I don't think anyone would argue the physical phenomenon doesn't exist.
I've heard of technolgy exploiting this that uses ultrasonic transducers to create a fixed ultrasonic tone and a modulated one so that their interaction formed a third signal in the audible range. It was billed as a way of creating sound with tiny ultasonic transducers for things like laptops. I think the downfall was that there were huge power requirements for the ultrasonic signals which had to be absolutely enourmous in amplitude to generate even a quite low amplitude audible signal. Probably killed all the rodents and insects in the neighborhood of the lab.
If that is indeed the case, the audible range singnals induced by the modulation of the ultasonic components would be most likely too low in amplitude to be heard, and most certainly masked by the signals in the audible range. I don't know much about the technical aspect of such modulation but I believe the carrier and modulating frequencies to do so would also have to be higher than several hundred khz.
Maybe not, I just found this white paper: http://64.227.81.118/hss/pdf/HSSWHTPAPERRevE.pdf (http://64.227.81.118/hss/pdf/HSSWHTPAPERRevE.pdf)
Non technical article version: http://www.elecdesign.com/Articles/ArticleID/5535/5535.html (http://www.elecdesign.com/Articles/ArticleID/5535/5535.html)
That could be your ammo Chris, but I'm not sure it applies.
The military is using similar technology to beam auditory warnings to craft approaching US Naval vessels and to create non-leathal crowd dispersal techniques from afar in addition to the advertising uses of the manufacturer above.
Then again I also found this research by David Griesinger that argues no and address this specific issue:
Power point: http://world.std.com/~griesngr/intermod.ppt (http://world.std.com/~griesngr/intermod.ppt)
or Google html version: http://72.14.205.104/search?q=cache:ymfUfDx8lnoJ:world.std.com/~griesngr/intermod.ppt+modulate+ultrasonics+to+make+audible+sound&hl=en&gl=us&ct=clnk&cd=12 (http://72.14.205.104/search?q=cache:ymfUfDx8lnoJ:world.std.com/~griesngr/intermod.ppt+modulate+ultrasonics+to+make+audible+sound&hl=en&gl=us&ct=clnk&cd=12)
-gone campin'
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Here is a MLS sample flat from 20hz to 90khz with a 20hz LPF butterworth 7th order with a band pass ripple of 0.5
Chris, that's a digital implementation of a Butterworth filter. You don't get skirts that steep with a real filter. And remember you can't use a digital filter to avoid aliasing. You can only use a real filter to avoid aliasing. Notice how the stopband skirt levels out? A real Butterworth filter's skirts keep going down as you increase frequency and it does so at a 6 db per pole per octave rate.
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So if we believe what you’re saying then sympathetic resonation does not exist! I believe that ultra sonic information from say an instrument like a horn can go up to 40k, there are sympathetic resonations that are above 20k that will effect what we hear in the audible range. Why should we limit our selves to 20k? When it’s a fact that many instruments produce sound well above that point, how can you say that the information above 20k does not effect the information we hear below it????
No sympathetic resonance? I said no such thing. Sympathetic resonance occurs at harmonics of the fundamental frequency of excitation. By definition, those sympathetic resonances are at higher frequencies than the fundamental frequency that excited them. If the sympathetic resonances are in the ultrasonic range, I claim that you can't hear them. You seem to believe that somehow they affect how you hear information in the audible range. One of us is wrong.** Let's find out which one of us is that person. But let's make sure that the test itself is valid. You can't allow your signal processing on ultrasonic signals to affect signals in the audible portion of the signal or the test will be invalid.
What we need are two signals. Each should have identical signals within the audible portion of the audio spectrum (both in amplitude and in phase) and only one of them should have significant energy in the ultrasonic portion of the audio spectrum. The trick is how to make those two signals without inducing changes in the audible portion of the audio spectrum. You can't just go imposing some arbitrary amplitude shaping to the spectrum without also re-equalizing the phase in the affected portion of the spectrum. Otherwise, it's likely that you will inadvertently affect the audible portion of the audio signal too.
**Two men say they're Jesus. One of them is wrong. -- Dire Straits
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So if we believe what you’re saying then sympathetic resonation does not exist! I believe that ultra sonic information from say an instrument like a horn can go up to 40k, there are sympathetic resonations that are above 20k that will effect what we hear in the audible range. Why should we limit our selves to 20k? When it’s a fact that many instruments produce sound well above that point, how can you say that the information above 20k does not effect the information we hear below it????
No sympathetic resonance? I said no such thing. Sympathetic resonance occurs at harmonics of the fundamental frequency of excitation. By definition, those sympathetic resonances are at higher frequencies than the fundamental frequency that excited them. If the sympathetic resonances are in the ultrasonic range, I claim that you can't hear them. You seem to believe that somehow they affect how you hear information in the audible range. One of us is wrong.** Let's find out which one of us is that person. But let's make sure that the test itself is valid. You can't allow your signal processing on ultrasonic signals to affect signals in the audible portion of the signal or the test will be invalid.
What we need are two signals. Each should have identical signals within the audible portion of the audio spectrum (both in amplitude and in phase) and only one of them should have significant energy in the ultrasonic portion of the audio spectrum. The trick is how to make those two signals without inducing changes in the audible portion of the audio spectrum. You can't just go imposing some arbitrary amplitude shaping to the spectrum without also re-equalizing the phase in the affected portion of the spectrum. Otherwise, it's likely that you will inadvertently affect the audible portion of the audio signal too.
**Two men say they're Jesus. One of them is wrong. -- Dire Straits
Well I know I am correct so I win! >:D
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Okay, the recording that got me all excited about Senn 421's (remember those? ;-)) is now seeding at http://www.shnflac.net/details.php?id=836332f70b10bc45fbf711a399c2d4c20dabac68. 16 bit today, 24 bit tomorrow. You can see for yourself if I'm right or if I need a hearing aid. :-)