Sound card clipping?

[cowgod2007]

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!

 

[Idonno]

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

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

while 120V AC is indeed 60Hz or cycles per sec.

as well as the fact that far more often than not your statement

do know why you are having a problem with 60hz? because that is the frequency of electricity.

is mostly wrong since

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.

 

[Guest]

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

while 120V AC is indeed 60Hz or cycles per sec.

as well as the fact that far more often than not your statement

do know why you are having a problem with 60hz? because that is the frequency of electricity.

is mostly wrong since

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?

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.

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.

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?

bubbye.

 

[Idonno]

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!

grandiose advise.

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

 

[Guest]

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!

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.

grandiose advise.

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

need a dictionary? grandiose
that is ever so relevant now.

 

[gamerk316]

Ok people, thats enough.

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.

 

[Idonno]

Ok people, thats enough.

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.