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16

I always use double sampling rates if possible, for two important reasons. First reason: to get rid of the characteristics of the anti-imaging filter when working with analog sound sources. What is an anti-imaging filter? Let's say I am recording on 44100 Hz. If I would record a sine wave of less then 10 KHz, you could clearly see the sinewave when you ...


10

For projects which are strictly digital, ie, using pure digital synthesis and not recording any material which would be converted from analog -> digital, is there any advantage to using sampling rates above 44.1kHz? Yes. Some examples: Creation of frequencies you don't want Aliasing from digital synthesis Many square/sawtooth/triangle wave ...


6

That's Nyquist-Shannon sampling theorem. Think about it in the other way: imagine you are given a set of samples (the black dots in the image). The most intuitive way to recover the original analog signal would be by joining these samples. But you can easily notice that there is not a unique way of joining them. In fact, there are infinite possible ...


4

Sounds above the Nyquist limit (in this case 4000 Hz) will seem to fold back into the allowed range. For example, a tone 100 Hz above the limit will appear as a phantom tone 100 Hz below it. So a frequency of 4100 will seem to appear at 3900, one at 4200 will appear at 3800, and so on. This is called 'aliasing' or 'folding'. The apparent 3900 tone is an ...


4

To have headroom for effects is a theoretically (and practically) valid reason to have a higher sampling rate than twice the human hearing limit. The reason for this is easily visualised by comparing with image editing – if you only have say 800x600 px image with an overall shot of a high contrast brick wall, fishnet, striped textiles, or other finely ...


3

Another good reason to use a higher sampling rate is to work around deficiencies of plugin implementations. Many plugin writers do not properly take into account the bandwidth-expanding effects of nonlinear signal operations, and as a result you can get aliasing effects before you leave the box. For example, a compressor is basically a voltage-controlled ...


2

For what it's worth, the mathematical rationale, at least to the needs of the audio world, is generally described by the Nyquist-Shannon sampling theorem, sometimes just referred to as the Nyquist theorem, which in basic language just states that to fully reproduce a waveform with a max frequency n Hz, you need 2n samples per second.


2

The classic PC Speaker is a very simple system that was designed to be inexpensive and use off-the-shelf parts of the early 1980s. The core problem is that the PC speaker system consists of a cheap speaker, a timer, and digital pulses. Using this system, you can get quite a lot of interesting sounds out that weren't originally intended. But you are limited ...


2

The result of using a sampling rate that is too low for your program material is easiest understood with a graph. If the blue wave is the actual sound you are attempting to record, in your example, say, 4.5KHz, with a sample rate of 8,000 samples/sec, the signal can only be recreated with the information that was recorded (in red). So to answer your ...


1

Interesting question. First, here's a question for you: can your supercomputer actually calculate and render, in real time, an animated zoom on images so complex? My guess (only a guess) is that it will need more than 1/30 sec. to complete each frame (let's assume 30 frames per second as a minimal frame rate in which to achieve a smooth zoom). If that's the ...


1

There isn't a good answer to this. It depends on the quality of the speaker and the quality of the sound chipset that is sending the output to the speaker. 20khz is a typical estimate of average human hearing range, so I wouldn't be that surprised if the sound card is more accurate. Generally they try to cover the range from 20hz to 20khz as the "typical ...


1

Yes, sample clocks vary, and this can cause recordings on different devices to drift over time. If you're trying to figure out the frequency at which something was recorded, you could do a bandpass filter around 50 Hz or 60 Hz (depending on which country you're in), and then analyze the hum which is picked up in the recording. If you're in the US, the hum ...


1

What file format are you passing back and forth? The sample rate is stored in the file. No need for a reference tone. One of you is likely recording at a different sample rate than the other. Probably one is recording at 44.1kHz, and the other 48kHz, or 88.2/96, etc. Now, audio interfaces do vary in sample rate frequency. 44,100Hz can be 44,099Hz on ...


1

When recording with multiple tracks, I believe the bit depth is more important than the sampling rate. So for example, 24bit would be better that 16bit. This has to do with the way that your tracks are mixed down together and something called "rounding errors" when there are not sufficient bits. Most hardware and software can now easily support 96k & ...



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