8

This is just a limited (or clipped through distortion) sound wave added through the normal mixing process, to a lower frequency soundwave. I've created a simple example to demonstrate what I mean. The higher frequency sound wave; Let's pretend it's a nice brass instrument: The brass instrument is then clipped for whatever reason: The ...


4

You will note that this emulates the effect of a DC blocking capacitor, in a record player on clipped vinyl... i.e. no 0Hz being passed. This waveform's clipping slopes exclusively the same way, towards zero. Unless this were a mixture of a triangle, and a square (clipped) wave, of the same frequency, with a phase difference of 90 degrees, Marc W's example ...


4

First of all, let's examine what happens when a signal clips: An audio signal can be represented by a string of numbers, so let's say you wan't to record this signal: 0 0.5 1.0 1.2 1.0 0.5 0 But the signal clips, so what you get is this: 0 0.5 0.9 0.9 0.9 0.5 0 See what's up? Suddenly no signal is present above 0.9... The thing ...


4

You're quite right that it's some sort of nonlinear operation – hard digital clipping in fact. And honestly it's somewhat obvious why this happens: you're superimposing two sine waves, each of which is already peak-normalised. The result of such a combination will in general exceed 0 dBFS, and without special handling (limiter etc.) that means the ...


2

If you have a sample loaded in Live, it generally is warped meaning that Live will generally attempt to preserve the pitch of it no matter how you stretch it in time. This is of course impossible to do perfectly, but it uses some very clever algorithms to attempt it by making educated guesses about the data. The different algorithms are referred to ...


2

There's several different warp mode in Live. Have a look at the cli view panel. If the clip is unwarped, slowing down the tempo will pitch the clip down and make it longer, and vice versa, like in the analog world. If the clip is warped Live will stretch it but keep the original pitch. Those "clipping" you mention are because your clip are warped with the "...


2

I would suggest that if the audio data is to be used in a research project in which having not clipped audio data is of some importance, you should start again. Although you might use some tools to make the clipped files more audible, you actually have no idea of what the data is in the clipped zones (unless that they are over the maximum input level).


2

Ohmicide from Ohm Force is specifically made to provide over-the-top distortion effects. Available as 32/64-bit Audio Unit, VST, and RTAS.


2

See this and scroll down to "Converting and Using Floating Point Samples". One good reason to convert integer samples to floating point samples in [-1,1] is so that you can mix bit-depths of integer samples easily. For instance, if you have a 24-bit file with a sample that is half of positive full scale, and an 8-bit file with a sample that is half of ...


2

Yes, certain compression formats can result in higher peak levels than the same material pre-compression. -1dBTP ("true peak" level, relative to full scale) seems to be the accepted maximum safe level for pre-compressed material when considering this. It's important to ensure you are infact analysing true peak (aka. inter-sample peak) level and not sample ...


1

Yes, the picture shows a heavily clipped signal. Decrease the input level until you never clip. Typical recommendation is to aim for around -10dB full scale when capturing input and to record at 24bit. This will get the best quality "raw" input you can get from the analog signals converted to digital realm. Once you have a good quality input you ...


1

Simply use the non-clipping right channel. It will have sligthly more noise but no clipping compared to the left channel.


1

You have to consider exactly how much headroom you have available to you in a 32-bit or 64-bit float scenario. The answer is - masses! You should be doing your mixing at 64-bit level these days in a modern operating system, so simply make sure that you don't scale your samples to such a level that adding them will clip the output. Remember that a value of 0....


1

A simple, easy way of doing it is by using the amplify effect in audacity. Just select the audio and increase by a couple of dbs and listen to how it sounds (Remember to enable clipping). If you want it more distorted apply the effect again by different amounts until you get what you want. Really what you are looking for is distortion. Any digital ...


1

Clipping and discontinuities can be detected by using the Izotope RX De-clicker and selecting "output clicks only". The resulting timeline will indicate where in the original material the discontinuities are. Not all of them will be full-on faults in the recording, you will have to use your ears to determine which of the candidate artifacts are actually real ...


1

It's related to CPU usage priority & streaming the audio into your system from the USB - this has no relation to the RAM & that's why adding RAM didn't help you . what you are experiencing is called USB Dropouts caused by Interrupts. finding a good guide to help you optimize your mac for audio recording might actually help you more then adding ...


1

Take a look at the command-line utility SoX. The 'trim' effect might be suitable for your use case.


1

Generally no. If the clipping was only one sample the CIA might be able to do it. Clipping causes high order harmonics. Look at the FFT of a square wave versus a sine wave. The more you clip the more that sine wave looks like that bunch of squiggles from the square wave.


1

Is there any way to recover the data. Yes, you can use de-clipping software, e.g. Waves or Izotope plugins created for this purpose. what difference to the frequency spectrum the clipping will make? "As the signal simply "cuts" or "clips" at the maximum capacity of the amplifier, the signal is said to be "clipping". The extra signal which is beyond ...


1

The isotope RX restoration software has some de-clipping features - though any sound coming in at a greater volume that the clipping point threshold is not actually stored and will be digitally synthesised - the best solution would be to redo the recording session.


1

You're not approaching this in the right way. By doing this "iterative normalization" you will be basically changing gain with little to no crossfade between the regions. This is not going to give you a natural sounding result. Sure an overall gain reduction may be necessary, but the initial recording should have enough headroom so that you can process the ...


1

Though this is an old question, a Google search brought me here. Audacity (a free audio editor) actually has a built in tool for repairing clipping, following the logic described in the accepted answer. It works surprisingly well for mild clipping. Link to Audacity docs


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