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Clipping occurs when the signal exceeds the maximum dynamic range of an audio channel. If you're recording a sine wave, clipping looks like somebody cut the top and bottom of the sound wave: (http://en.wikipedia.org/wiki/Clipping_(audio)) Analogue equipment will often add other artifacts along with clipping and rarely will the peaks be cut cleanly as with ...

I think it is possible, but depends in how clipped the signal is. Let me explain: think about a softly clipped signal. Clipping is present only in the greatest peaks, and therefore appears for a short time lapse. This kind of method could detect the clipped intervals and ''soften'' them, based in the previous behaviour of the signal. An statistical ...

You're right in that when your digital signal "clips" at 0dB, it's an absolute maximum limit. 16-bit audio has 2^16 possible volume levels, and once you get to 2^16, there's no higher number to express your sound, so it maxes out, or clips. With analog signals, your "clipping" limit is less well defined. The term distortion in an amplifier just refers to ...

"Clipping" or "hard clipping" distortion generally refers to trying to record at a volume level that is higher than the maximum value your hardware will check for. Since it can't store this value, it simply records the highest value available. If you look at the waveform, the peaks will look flattened off, as if someone clipped them off with a set of ...

That's a well-known problem. First of all, you shouldn't be using the microphone connector on the laptop. These inputs are usually not only particularly susceptible to buzzing noises, but also to all kinds of distortion and aliasing. Use an external audio interface, there are very affordable USB ones available. Using an interface with XLR inputs, connected ...

Nope, the sound information is missing and there is no way to recover it. (At least not that I'm aware of.) Even if there was, it would have to basically be completely guessing at what should be there. (Edit: there is software that will make the guess, and that's what the answer with the waveforms is illustrating. It is worth noting that it is a guess ...

The signal levels generated by most keyboards should not be enough to damage the microphone/line-in jack when plugged in directly. However, I do recommend you not turn the keyboard up past the point where it begins to distort just to make sure. To get the best recording level, First set the volume on the keyboard as high as possible without causing ...

Check this Wikipedia article The article doesn't go into full detail, but let's address yo your problem: YES Audio loses quality very easily specially when changing its sampling rate and in your particular case it probably has to do with aliasing, which is a consequence of re sampling audio. Basically - as people tell me at least - you should only raise ...

With analog equipment, if the signal level rises above the maximum input level of the equipment, it will begin to distort. The corresponding output level may or may not continue to increase, depending on the equipment. At some point, you will begin to damage the equipment. Equipment that has Line-Level inputs will typically not accept levels much beyond ...

I'm just going to answer the levels part of this question, as you computer should be able to handle a LINE level signal to the LINE IN plug, but I can't be certain without knowing the hardware. Best practice in setting levels is to turn every right down, then plug it in. Set you instrument to 75% (rule of thumb) and your input gain to 75% (or unity gain). ...

If you can achieve 143dBSPL in a studio, I would be very seriously more concerned about your own ears than the microphone. 130dBSPL is the estimated threshold of pain, roughly 4.34 times quieter than 143dBSPL (if my math is correct). If you take Inverse Square Law into consideration, if you feel the mic is being subjected to too much level, move it further ...

Any audio signal path, digital or analog, will have a maximum amplitude. If you feed it a signal, or amplify it more, you will get "clipping distortion". You avoid it by doing something that you should always do at all times: You should make sure that at each part of your signal path the path should be as loud as possible without distortion, but no louder. ...

The question about the nature of clipping phenomena was discussed many times here on this site (for example here) and on dozens of other places which are easily googled. So as it was already discussed, clipping - is just few points where signal amplitude exceed the available range of finite amplitude levels. This causing the wave signal to become less ...

NO, resampling – provided it's done properly – does not degrade the quality. In any sampled audio signal, the assumption is that, prior to sampling, it was ensured that the signal only contains frequencies up to the Nyqvist frequency before sampling; for this purpose AD converters contain low pass filters, which don't work perfectly but can work very good, ...