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Sound card clipping?

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June 4, 2012 8:50:26 PM

Hey guys,

So I'm recording a very low frequency in Audacity ~60 Hz. However, it should be a perfect sinusoidal wave when I record but it clips at the peaks and the peaks become square-ish rather than round. I'm thinking its the sound card because I've tried the input on different computers and get different, better looking sin waves and recordings.

Can anyone give me some input? Maybe the sound card doesn't record at that low of a frequency? Maybe its distorting the sound? I've tried playing around with the recording volume and lowering it does help, but its still not nearly as good as I would like. My laptop is a dv6t-qe with beats audio (I tried looking up the exact name of the integrated sound card but I can't find that or the specs).

Thanks in advanced!

More about : sound card clipping

June 4, 2012 10:24:46 PM

It's hard to say for sure but, integrated laptop soundcards aren't known for great low frequency range since neither the laptop speakers or the external speakers most people hook up to it can reproduce that range. The fact that the sine wave has a square-ish instead of a rounded peak also suggests that your internal sound card has reached some sort of limiting factor, very possibly the design it's self.

If you really need to record below 60 Hz, I would suggest a external USB soundcard.
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June 4, 2012 10:51:26 PM

Hi cowgod,

1) what is the source of your 60Hz signal?

2) How are you feeding this source into your soundcard? what leads?
and re-check the input connection (line in? mic in? mic boost on? etc)

3) How does the amplitude/volume level 'look' in audacity - do yo need to vertically zoom in alot to see this wave? ....i.e. is the input gain non-optimal?

4) Are you able to boost the signal going into the soundcard i.e. at source (outside the pc)?

5) just an educated guess - are you trying to remove 60Hz mains noise from a recording?

6) what is the soundcard?

7) have you tried recording into a different wave application e.g. sound forge?

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Related resources
June 4, 2012 11:26:45 PM

Idonno said:
It's hard to say for sure but, integrated laptop soundcards aren't known for great low frequency range since neither the laptop speakers or the external speakers most people hook up to it can reproduce that range. The fact that the sine wave has a square-ish instead of a rounded peak also suggests that your internal sound card has reached some sort of limiting factor, very possibly the design it's self.

If you really need to record below 60 Hz, I would suggest a external USB soundcard.


That's what I was thinking. We have a cheap microphone in our lab that records from 40Hz - 19,000 Hz but starts to distort/clip at 40-70Hz, similar to what happens on my laptop. We're thinking the same, lowend sound card is the problem but I just want to make sure before I jump to conclusions because the clipping is slightly different (could just be how its processed).


mesab66 said:
Hi cowgod,

1) what is the source of your 60Hz signal?

2) How are you feeding this source into your soundcard? what leads?
and re-check the input connection (line in? mic in? mic boost on? etc)

3) How does the amplitude/volume level 'look' in audacity - do yo need to vertically zoom in alot to see this wave? ....i.e. is the input gain non-optimal?

4) Are you able to boost the signal going into the soundcard i.e. at source (outside the pc)?

5) just an educated guess - are you trying to remove 60Hz mains noise from a recording?

6) what is the soundcard?

7) have you tried recording into a different wave application e.g. sound forge?



1) I'm doing research and I'm using transisters and resistors to transform electricity from a power main into a signal that is able to be read through a computer. Basically a "microphone for electricity"

2) Through the standard 3.5mm jack. Checked input connections, all seems to be in order

3) I do need to zoom in to see it but I don't need to increase/decrease scales to see anything fine.

4) N/A signal is constant

5) nope, sorry lol

6) Its an integrated soundcard (don't know too much about soundcards and specs). I tried doing a little research, but couldn't find much. Its on my HP DV6T quad edition and it has "beats audio". I've read somewhere the beats audio is just a rebranded intel integrated audio.

7) Nope, but I'll definitely try that. I don't see it being too big of a difference though
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Anonymous
June 4, 2012 11:44:14 PM

two quick thoughts:
have you tried MOVING the microphone?
can you put a parametric EQ in the signal chain? (actually a compressor would better.)
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June 5, 2012 12:50:46 AM

You're not responsible for putting that damn mains noise into certain venues (and recordings) my old band used to play in? - a pain in the ass trying to get rid of it! lol

A few more daft questions......

1. Are you recording amplified audio directly from a speaker? If so, is the speaker rated to deliver a 60Hz audio signal with plenty headroom (using a dedicated bass speaker?)

2. Check that your mic is recording in it's optimal axis position and that it is placed optimally next to the speaker.

3. Is it also possible that 60Hz is getting too close to the mic's low limit? (a higher quality mic will better guarantee signals between these limit boundaries)

4. Try a different mic

5. Have you tried feeding the signal (for test purposes, must be the same signal the mic captures) into an Oscilloscope to confirm it's quality (will help with optimal mic placement)?

6. If all else fails, try a dedicated soundcard/audio interface - borrow one to try out.

7. Was this set-up identical when used with any of the different computers? and, what soundcard or motherboard were they using?

8. One last idea, try recording the output of a tone generator with this set up, sweeping down to the mics lower limit, then checking the waveform.

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June 5, 2012 2:16:02 PM

Anonymous said:
two quick thoughts:
have you tried MOVING the microphone?
can you put a parametric EQ in the signal chain? (actually a compressor would better.)


There isn't really a "microphone". If you look at my previous post, its taking a constant frequency (electricty from an outlet) and converting it into a readable "noise" frequency/signal that is fed directly into the computer through the standard audio jack. I don't know much about audio stuff but I would suspect using an EQ wouldn't help much? But idk.


mesab66 said:
You're not responsible for putting that damn mains noise into certain venues (and recordings) my old band used to play in? - a pain in the ass trying to get rid of it! lol

A few more daft questions......

1. Are you recording amplified audio directly from a speaker? If so, is the speaker rated to deliver a 60Hz audio signal with plenty headroom (using a dedicated bass speaker?)

2. Check that your mic is recording in it's optimal axis position and that it is placed optimally next to the speaker.

3. Is it also possible that 60Hz is getting too close to the mic's low limit? (a higher quality mic will better guarantee signals between these limit boundaries)

4. Try a different mic

5. Have you tried feeding the signal (for test purposes, must be the same signal the mic captures) into an Oscilloscope to confirm it's quality (will help with optimal mic placement)?

6. If all else fails, try a dedicated soundcard/audio interface - borrow one to try out.

7. Was this set-up identical when used with any of the different computers? and, what soundcard or motherboard were they using?

8. One last idea, try recording the output of a tone generator with this set up, sweeping down to the mics lower limit, then checking the waveform.


1) No, its not amplified. But its definitely processed through transister and resistors.

2) N/A the mic doesn't actually record any "sound" so positioning and other noise interferences won't matter

3) N/A there's no actual mic lol

4) We would have to build another lol

5) No, might be something to look into. I would need to get an oscilloscope first though...

6) Yea, what do you think about an external USB soundcard or something of the sort?

7) No, setup was different (different soundcard, on a dell desktop)

8) Will try, won't hope for too much.



At this point, I'm thinking its the physical limitations of the sound card/codec/processor.
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June 5, 2012 6:00:11 PM

Does anyone know a way to get the input through a higher quality, more sensitive mic and then into the PC? So instead of "mic" input --> PC, it would go like input --> high quality mic --> PC
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June 5, 2012 7:21:41 PM

Also, any opinions on external sound cards? Will an external card be of higher quality/same quality as an internal? If I go expensive enough, will external sound cards be able to accurately measure/read low frequencies like 60Hz?
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Anonymous
June 5, 2012 10:35:36 PM

ok, first off you did mention a mic:
Quote:
That's what I was thinking. We have a cheap microphone in our lab that records from 40Hz - 19,000 Hz but starts to distort/clip at 40-70Hz, similar to what happens on my laptop.

but didn't understand you weren't using a mic.

now let me put it to you this way; when something is called a MIC IN it means it is configured for a MICROPHONE.

whatever you are doing you don't know what you're doing.

please stop.
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June 5, 2012 11:06:52 PM

[looniam......I'm pretty much thinking the same thing ;) ....his response to the following will confirm]

Is an actual microphone involved? (and thus, a digital-analogue conversion step....followed by an analogue-digital conversion for the pc)- when you said earlier:

"We have a cheap microphone in our lab that records from 40Hz - 19,000 Hz "
.....and now say
"Does anyone know a way to get the input through a higher quality, more sensitive mic"
.....it pretty much suggested one was, hence our replies.

Internal/External sound cards/audio interfaces : there are many - the more expensive ones, of course, use higher quality components. I currently use a TC Electronic Studiokonnekt48 for various projects. This would be a bit of an overkill for your specific application, however, I would recommend seeing if you can hire a good quality audio interface from your local pro music shop.

Amongst other benefits, the D-A/A-D converters (if you need to use them!) will be much higher quality and, importantly, operate with a much higher resolution. The resulting signal can be passed into your pc using an appropriate link (e.g. firewire in my case, usb, or spdif, etc).

For the cost of 1 day's hire I would try these out (guaranteeing quality components 1st) - you may find you don't have to spend too much on a permanent replacement, heck, you might find that a cheaper creative x-fi card would do.

Back to the mic question (if one does exist, lol), for - as you say - a "a higher quality, more sensitive mic", see the same shop as above, but of course, ensuring you ask for a model fit for purpose.



Ah.....one last thing......take a few detailed photos of your set-up & post here.

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June 5, 2012 11:17:45 PM

Anonymous said:
ok, first off you did mention a mic:
Quote:
That's what I was thinking. We have a cheap microphone in our lab that records from 40Hz - 19,000 Hz but starts to distort/clip at 40-70Hz, similar to what happens on my laptop.

but didn't understand you weren't using a mic.

now let me put it to you this way; when something is called a MIC IN it means it is configured for a MICROPHONE.

whatever you are doing you don't know what you're doing.

please stop.


Yeah, I don't. I'm working at an electrical engineering lab at a university doing research. Maybe I'm the reason why America is so far behind in science and you can help us back up?

And let me put it to you this way. We aren't looking at sounds. We are looking at electricity, specifically from an outlet. We built our own hardware to transform it down to a usable level and feed it directly into the audio jack so it doesn't blow up our computer.



mesab66 said:
[looniam......I'm pretty much thinking the same thing ;) ....his response to the following will confirm]

Is an actual microphone involved? (and thus, a digital-analogue conversion step....followed by an analogue-digital conversion for the pc)- when you said earlier:

"We have a cheap microphone in our lab that records from 40Hz - 19,000 Hz "
.....and now say
"Does anyone know a way to get the input through a higher quality, more sensitive mic"
.....it pretty much suggested one was, hence our replies.

Internal/External sound cards/audio interfaces : there are many - the more expensive ones, of course, use higher quality components. I currently use a TC Electronic Studiokonnekt48 for various projects. This would be a bit of an overkill for your specific application, however, I would recommend seeing if you can hire a good quality audio interface from your local pro music shop.

Amongst other benefits, the D-A/A-D converters (if you need to use them!) will be much higher quality and, importantly, operate with a much higher resolution. The resulting signal can be passed into your pc using an appropriate link (e.g. firewire in my case, usb, or spdif, etc).

For the cost of 1 day's hire I would try these out (guaranteeing quality components 1st) - you may find you don't have to spend too much on a permanent replacement, heck, you might find that a cheaper creative x-fi card would do.

Back to the mic question (if one does exist, lol), for - as you say - a "a higher quality, more sensitive mic", see the same shop as above, but of course, ensuring you ask for a model fit for purpose.


Sorry, we were using a handheld mic (Olympus ws-700m) that we fed the signal directly into, then put on our computer. We also tried feeding the signal directly into our computer's audio jack. Both have clipping but to various degrees. Sorry for the confusion. And I think I've narrowed down the problem to the sound card and I'm going to do some testing with various external sound cards.
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Anonymous
June 5, 2012 11:35:42 PM

cowgod2007 said:
Yeah, I don't. I'm working at an electrical engineering lab at a university doing research. Maybe I'm the reason why America is so far behind in science and you can help us back up?

And let me put it to you this way. We aren't looking at sounds. We are looking at electricity, specifically from an outlet. We built our own hardware to transform it down to a usable level and feed it directly into the audio jack so it doesn't blow up our computer.


again what you are completely failing to understand is that you are taking a signal that is extremely harsh, noisy, and has a lot of ripple and trying to process it through sensitive bandwidth processing.

do know why you are having a problem with 60hz?

because that is the frequency of electricity.

now, did i just saved america?

if you want to listen to electricity get a oscilloscope!
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June 5, 2012 11:41:34 PM

Anonymous said:
again what you are completely failing to understand is that you are taking a signal that is extremely harsh, noisy, and has a lot of ripple and trying to process it through sensitive bandwidth processing.

do know why you are having a problem with 60hz?

because that is the frequency of electricity.

now, did i just saved america?

if you want to listen to electricity get a oscilloscope!


Actually, with the right transistors/resistors/circuitry, the signal is easily readable. I have a computer and the input is read in perfectly with no clippings. And the frequency only fluctuates between 59.90Hz and 60.10 Hz.

And oscilloscopes are all fine and dandy. Until you want to do mass testing.
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June 6, 2012 7:49:22 AM

Anonymous said:
do know why you are having a problem with 60hz?

because that is the frequency of electricity.

now, did i just saved america?
No you didn't save America because your answer was only slightly correct and mostly incorrect.

while 120V AC is indeed 60Hz or cycles per sec. other AC voltages are not and (normal) DC has no frequency.

I believe the OP want's to record the sound of electricity not measure the cycles per sec which is a different thing all together. I'm not sure why the OP wants to record the sound of electricity but, I'm sure he has his reasons.

So in the spirit of the OP's question(s) and of course saving America LOL, I'll try to answer the best I can.

As long as your mic is capable of going lower than 60Hz, IMHO the only other hardware that could be limiting you is your onboard recording hardware (and possibly software).

I always buy higher end soundcards that are all capable of 20Hz however you need far less if you just need one channel capable of recording low frequency's and playback is not much of an issue so it's real for me hard to give you a recommendation.

Certainly something like this: http://www.amazon.com/M-Audio-MobilePre/dp/B0041OSWX8 should do the trick but you might be able to find something cheaper if you look around.
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June 6, 2012 10:32:03 AM

Exactly! - taken straight off my previous posts...lol....though I'm sure were finally getting there ;) 

Is this the mic: http://www.olympusamerica.com/cpg_section/product.asp?p... ---> you may have introduced serious component/converter quality issues in this single link.

Please, again, detail exactly how you use this mic - do you record using the mic or mic-in feature - and describe how.

In any case, you are including (very) basic audio engineering into your setup !! fact !! - thus the use of audacity or, as I use (and other pros primarily use), sound forge.

Now, since we've established the use of a mic (finally), this of course means recording an analogue signal (if using the 'mic' component...see above, converted from your digital source - see later)......coupled with a digital conversion to go into the pc !!! ........now please re-read messages before. I doubt that any signal will 'pass through' this 'mic' in a pure format - i.e. converters will be at play irrespective of how you use this mic.

Here's another (educated?) idea!! - try bypassing these D-A and A-D converters (assuming your 'original' signal sits - at some point early - in the digital domain) and see if you can pass the signal into an audio interface that accepts a digital input (most do!) ----> with this you are removing a set of variables (converters + mic) ---->sounds dooable/good? You may (or may not) need to play about with getting a digital 'lock' - I'll not say any more on this however, you are working in an electronics uni department (e.g. frequency converters hint hint)

The use of an oscilloscope must be included for initial setup / validation purposes, thereafter you're good for the mass run.

p.s. please send some of your profits from this to us (mass run etc) ;)  ;) 

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June 6, 2012 3:30:03 PM

Idonno said:
No you didn't save America because your answer was only slightly correct and mostly incorrect.

while 120V AC is indeed 60Hz or cycles per sec. other AC voltages are not and (normal) DC has no frequency.

I believe the OP want's to record the sound of electricity not measure the cycles per sec which is a different thing all together. I'm not sure why the OP wants to record the sound of electricity but, I'm sure he has his reasons.

So in the spirit of the OP's question(s) and of course saving America LOL, I'll try to answer the best I can.

As long as your mic is capable of going lower than 60Hz, IMHO the only other hardware that could be limiting you is your onboard recording hardware (and possibly software).

I always buy higher end soundcards that are all capable of 20Hz however you need far less if you just need one channel capable of recording low frequency's and playback is not much of an issue so it's real for me hard to give you a recommendation.

Certainly something like this: http://www.amazon.com/M-Audio-MobilePre/dp/B0041OSWX8 should do the trick but you might be able to find something cheaper if you look around.



Yea, we're looking into doing testings with cheap external sound cards and if the clipping is consistent across all machines, then we know the problem lies in the sound card and we will have to get a higher quality one.




mesab66 said:
Exactly! - taken straight off my previous posts...lol....though I'm sure were finally getting there ;) 

Is this the mic: http://www.olympusamerica.com/cpg_section/product.asp?p... ---> you may have introduced serious component/converter quality issues in this single link.

Please, again, detail exactly how you use this mic - do you record using the mic or mic-in feature - and describe how.

In any case, you are including (very) basic audio engineering into your setup !! fact !! - thus the use of audacity or, as I use (and other pros primarily use), sound forge.

Now, since we've established the use of a mic (finally), this of course means recording an analogue signal (if using the 'mic' component...see above, converted from your digital source - see later)......coupled with a digital conversion to go into the pc !!! ........now please re-read messages before. I doubt that any signal will 'pass through' this 'mic' in a pure format - i.e. converters will be at play irrespective of how you use this mic.

Here's another (educated?) idea!! - try bypassing these D-A and A-D converters (assuming your 'original' signal sits - at some point early - in the digital domain) and see if you can pass the signal into an audio interface that accepts a digital input (most do!) ----> with this you are removing a set of variables (converters + mic) ---->sounds dooable/good? You may (or may not) need to play about with getting a digital 'lock' - I'll not say any more on this however, you are working in an electronics uni department (e.g. frequency converters hint hint)

The use of an oscilloscope must be included for initial setup / validation purposes, thereafter you're good for the mass run.

p.s. please send some of your profits from this to us (mass run etc) ;)  ;) 



How would I bypass the D-A and A-D converters? I'm not too knowledgeable on the intricacies in sound cards but don't you need to go through the DA/AD converters to turn the analog signal into a digital one, which allows it to be read into a computer? The signal is no help if I can't analyze it/modify the signal on a computer.

P.S. No profits on my side. I'm researching in a public university so unless a professor decides to patent/copyright this idea, I think this research/information will be public domain.
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Anonymous
June 6, 2012 6:15:39 PM

Idonno said:
No you didn't save America because your answer was only slightly correct and mostly incorrect.

and i am incorrect where?
this is not simply a matter of being able to throw money at it to solve the problem. the problem is using an incorrect method of gaining a "readable" sample.
until the OP can LOWER the SIGNAL STRENGTH it does not matter how fancy or expensive the sound card is, it will inherently overload the 60 Hz. (the frequency of electric in north america.)

now if the OP is successful in their endeavor it will be a red letter day for live audio engineers world wide; for they would never have to worry about a ground hum again.

until then have fun chasing your tail . .

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June 6, 2012 6:20:12 PM

Anonymous said:
and i am incorrect where?
this is not simply a matter of being able to throw money at it to solve the problem. the problem is using an incorrect method of gaining a "readable" sample.
until the OP can LOWER the SIGNAL STRENGTH it does not matter how fancy or expensive the sound card is, it will inherently overload the 60 Hz. (the frequency of electric in north america.)

now if the OP is successful in their endeavor it will be a red letter day for live audio engineers world wide; for they would never have to worry about a ground hum again.

until then have fun chasing your tail . .


I am lowering the signal strength...We build custom hardware to transform the power since we get our electricity directly from an outlet and feeding that into our computer could possibly blow it up or at the best case, destroy it. I think I said that somewhere...no more bickering ok?
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June 6, 2012 7:13:53 PM

cowgod2007 said:
How would I bypass the D-A and A-D converters? I'm not too knowledgeable on the intricacies in sound cards but don't you need to go through the DA/AD converters to turn the analog signal into a digital one, which allows it to be read into a computer? The signal is no help if I can't analyze it/modify the signal on a computer.
There is no way to do this. All storage medium on a computer is digital only, this includes very temporary storage like ram (not usually considered storage). There is simply no way for analog to be recorded on a PC without first converting it to digital then back to analog before it is played through speakers.

The only way to "bypass the D-A and A-D converters" on your sound hardware is to feed it an already converted (to digital) signal on the way in and a digital signal on the way out that still needs to be converted to analog before it can be played.

While this could be of benefit if the onboard DAC's are of particularly poor quality and all PC speakers with a digital input can accomplish the final conversion to analog, you would still need to have a DAC for the initial conversion from your analog mic to digital for your PC. A separate DAC for this (initial conversion) would cost as much or more than a external sound card that could handle both conversions. Also most mid-level soundcards have much better DAC's than PC speakers or onboard soundcards have and you are probably measuring the low end frequency at the PC/before the speakers anyway.
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June 6, 2012 7:47:47 PM

Anonymous said:
and i am incorrect where?
this is not simply a matter of being able to throw money at it to solve the problem. the problem is using an incorrect method of gaining a "readable" sample.
until the OP can LOWER the SIGNAL STRENGTH it does not matter how fancy or expensive the sound card is, it will inherently overload the 60 Hz. (the frequency of electric in north america.)

now if the OP is successful in their endeavor it will be a red letter day for live audio engineers world wide; for they would never have to worry about a ground hum again.

until then have fun chasing your tail . .
Easy looniam, I was just pointing out that electricity has more variables than 120V AC. With the whole "saving America" thing, I thought my response was lighthearted. If it came across as anything else, it wasn't meant that way.
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June 6, 2012 8:50:42 PM

Idonno said:
There is no way to do this. All storage medium on a computer is digital only, this includes very temporary storage like ram (not usually considered storage). There is simply no way for analog to be recorded on a PC without first converting it to digital then back to analog before it is played through speakers.

The only way to "bypass the D-A and A-D converters" on your sound hardware is to feed it an already converted (to digital) signal on the way in and a digital signal on the way out that still needs to be converted to analog before it can be played.

While this could be of benefit if the onboard DAC's are of particularly poor quality and all PC speakers with a digital input can accomplish the final conversion to analog, you would still need to have a DAC for the initial conversion from your analog mic to digital for your PC. A separate DAC for this (initial conversion) would cost as much or more than a external sound card that could handle both conversions. Also most mid-level soundcards have much better DAC's than PC speakers or onboard soundcards have and you are probably measuring the low end frequency at the PC/before the speakers anyway.



That makes more sense. That's what I was thinking because currently my input is analog. Sorry for not specifying, I thought it was assumed.
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June 6, 2012 9:03:21 PM

No probs, cowgod

I'm starting to repeat myself, and, getting the impression that the full start-finish responses that have been posted haven't been fully read before some comments.

e.g -->

"bypassing these D-A and A-D converters (assuming your 'original' signal sits - at some point early - in the digital domain) and see if you can pass the signal into an audio interface that accepts a digital input (most do!) ----> with this you are removing a set of variables (converters + mic)"

--> in reading my earlier posts, and specifically, the repeated unanswered question "Please, again, detail exactly how you use this mic - do you record using the mic or mic-in feature - and describe how. " (this mic offers >1 input method!!!).................

-->We all need to know how you are using this mic:
1. Placing it against a speaker? - if yes, please describe the setup
2. Feeding a 'signal' into it's 'mic in' socket (see - I did have a glance over its manual) - if yes, are you performing a D-A conversion process beforehand (loaded question)? - if yes, then you 'might' not need to and thus you could bypass at least 2 DAC steps IF you could get the ORIGINAL digital signal 'directly' into your pc - you may need to employ a suitable D-D conversion to get the signal to talk to your pc spdif (if it has one) or the audio interface D input. --> thereby BYPASSING any D-A then A-D steps. This is just a suggestion and you may need a custom D-D converter (unless an appropriate one exists)....and this is where your field comes back into play.

p.s. of course pc's operate in the D domain....... ;) 

Anyway, please don't be offended by any 'implied' tone - it's sometimes damn difficult condensing a multitude of thoughts down into a few lines - oftentimes we all over-assume in life :) 
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June 6, 2012 10:12:55 PM

I think your best next step would be to see if anyone at the university has a decent external sound card you could borrow and try.

P.S. No offense by any 'implied' tone "mesab66" and please don't take any on mine. Just trying to save my country. :D 
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Anonymous
June 7, 2012 2:01:48 AM

Idonno said:
Easy looniam, I was just pointing out that electricity has more variables than 120V AC. With the whole "saving America" thing, I thought my response was lighthearted. If it came across as anything else, it wasn't meant that way.


if you are going to tell me i am incorrect, then kindly point out the flaw. after 12 years of audio engineering in a live production environment; i do know what electricity is thank you very much. i also know a thing or two about signal processing and what can and cannot be processed by a given means. so if those experiences and the knowledge believe i gained through such has led me to draw erroneous conclusions; i beg of you to be so kind as to straight them out as opposed to just saying they are wrong without a explanation.

its the sake of the country at stake ya know or "Idonno"?

just don't try to feed me a hubris statement and claim to being lighthearted :non: 
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June 7, 2012 3:20:41 AM

Anonymous said:
if you are going to tell me i am incorrect, then kindly point out the flaw. after 12 years of audio engineering in a live production environment; i do know what electricity is thank you very much. i also know a thing or two about signal processing and what can and cannot be processed by a given means. so if those experiences and the knowledge believe i gained through such has led me to draw erroneous conclusions; i beg of you to be so kind as to straight them out as opposed to just saying they are wrong without a explanation.

its the sake of the country at stake ya know or "Idonno"?

just don't try to feed me a hubris statement and claim to being lighthearted
:non: 

I'm sorry your appear to be so insulted but, I think I was very clear. There was nothing hubris about my statement what-so-ever. I quite clearly pointed out the one instance where you were right
Quote:
while 120V AC is indeed 60Hz or cycles per sec.
as well as the fact that far more often than not your statement
Quote:
do know why you are having a problem with 60hz? because that is the frequency of electricity.
is mostly wrong since
Quote:
other AC voltages are not and (normal) DC has no frequency.


And while I'm glad you have 12 years of audio engineering, that is a far cry from electrical engineering when it comes to understanding electricity. I could boast about my credentials but since anyone can say whatever they like here (true or not) I'll refrain from that.

You made a mostly incorrect statement and I pointed that out in a lighthearted manor. For the life of me I don't know why you want to make a big deal of it. :pt1cable: 
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Anonymous
June 7, 2012 4:48:23 AM

Idonno said:
I'm sorry your appear to be so insulted but, I think I was very clear. There was nothing hubris about my statement what-so-ever. I quite clearly pointed out the one instance where you were right
Quote:
while 120V AC is indeed 60Hz or cycles per sec.
as well as the fact that far more often than not your statement
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do know why you are having a problem with 60hz? because that is the frequency of electricity.
is mostly wrong since
Quote:
other AC voltages are not and (normal) DC has no frequency.

voltages doesn't matter as far as frequency, the only difference is it is either 60 Hz in the Americas and part of Japan or 50 Hz in the rest of the world. the only fault i may of made was assuming the OP was in america, where it is 60 Hz. but to say frequency is dependent on voltage, as you eluded to, is entirely incorrect. and of course since the OP stated the electric was being drawn from a wall outlet well that certainly disqualified DC then didn't it?
Idonno said:

And while I'm glad you have 12 years of audio engineering, that is a far cry from electrical engineering when it comes to understanding electricity. I could boast about my credentials but since anyone can say whatever they like here (true or not) I'll refrain from that.

since you appear to not know what is a qualified audio engineer i'll be brief and assure you that it behooves the engineer to have a knowledgeable understanding of electricity since all the equipment uses it, either DC or AC, such as condenser microphones, mixing consoles, signal processors and amplifiers. and most all are made up of electronic components. so when in the field and something "breaks", ceases to function, it is not exactly going to fix itself along with being able to maintain the gear when there is downtime back at the shop. yes there are a lot of hacks that think after a few beers they can do an outstanding job at making a crappy local band sound like U2 or some kid with their laptop and a M-box with pro tools that they are now a recording studio; thats why i specified "live production audio engineer"; big difference.
Idonno said:

You made a mostly incorrect statement and I pointed that out in a lighthearted manor. For the life of me I don't know why you want to make a big deal of it. :pt1cable: 


so no, i made no incorrect statement. and yes it is very arrogant to claim someone is incorrect in a know it all manner to then give grandiose advise. see you may not be aware but when you poke someone they just might poke back :) 

i am appreciative of you time, thank you.

though since anyone can boost any type of credentials, whether true or not, it probably would be worthless to suggest taking the signal and running it through another channel, reversing the phase and them mixing both channels together. you know all about cancellation, right? and then i could say of course the signal would be zero then so they would then need to decrease the gain on one of the channels to get something.

but i could be just blowing smoke up folks arses, huh? :pfff: 

bubbye.
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June 7, 2012 7:19:04 AM

Quote:
you know all about cancellation, right?
Right, any low level audiophile is aware of the effects of reversing the phase on an audio channel. But that has absolutely nothing to do with what I said you were wrong about and changing the subject won't make you correct anyway. So stop crying. you were wrong. Get over it! :pt1cable: 

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grandiose advise.
Now that's funny, I don't care who you are! :lol: 
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Anonymous
June 7, 2012 7:01:00 PM

Idonno said:
Quote:
you know all about cancellation, right?
Right, any low level audiophile is aware of the effects of reversing the phase on an audio channel. But that has absolutely nothing to do with what I said you were wrong about and changing the subject won't make you correct anyway. So stop crying. you were wrong. Get over it! :pt1cable: 


it doesn't have anything to do with what you said to me, you can't understand that is directed to the OPs problem? actually there are quite a few low level audiophiles that can't comprehend signal cancellation. i moved on because i showed where is was completely correct even though you refuse to acknowledge it.

again, since north american AC electric is 60Hz in frequency; what signal the OP gets no matter how many resistors or transformer they use will overload at 60Hz because that will remain constant no matter what voltage or amperage. just look at the OPs posts and you will see it correct. as they say, the proof is in the pudding. or go here: Circuit Theory/Frequency Response (if you need me to cite my claims then so be it. i have worked with special people before)

now if the OP can introduce a reverse phase and adjust the gain; they may be able to receive something of a measurable signal. however there is a hazard that though it will reduce the level of the frequency, it will potentially double the magnitude depending on the gain adjustment.

obviously this all far beyond your scope of the fine job you do suggesting expensive high end computer components for folks that come here looking for build advice and have the funds for it. this is a college kid who has limited means and resources; your not helping.

the best help is what i first said, stop he doesn't know what he is doing.
Idonno said:

Quote:
grandiose advise.
Now that's funny, I don't care who you are! :lol: 

need a dictionary? grandiose
that is ever so relevant now.
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June 7, 2012 7:59:39 PM

Ok people, thats enough.

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There is no way to do this. All storage medium on a computer is digital only, this includes very temporary storage like ram (not usually considered storage). There is simply no way for analog to be recorded on a PC without first converting it to digital then back to analog before it is played through speakers.


Not entirely true; you could represent an analog waveform as a mathematical function, hence preserving its quality even when digitized [assuming you can accurately reproduce the original signal]. And its possible to record at a high enough quality level where you get *essentially* a good enough quality where any differences in quality are simply not distinguishable to human ears; your only real limitation is file size in that regard.
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June 7, 2012 9:45:28 PM

gamerk316 said:
Ok people, thats enough.

Quote:
There is no way to do this. All storage medium on a computer is digital only, this includes very temporary storage like ram (not usually considered storage). There is simply no way for analog to be recorded on a PC without first converting it to digital then back to analog before it is played through speakers.


Not entirely true; you could represent an analog waveform as a mathematical function, hence preserving its quality even when digitized [assuming you can accurately reproduce the original signal]. And its possible to record at a high enough quality level where you get *essentially* a good enough quality where any differences in quality are simply not distinguishable to human ears; your only real limitation is file size in that regard.


Finally some intelligent discourse. :sol: 
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