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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 ...

The 44.1kHz it's the sampling frequency, i.e. the frequency at wich the encoder samples the audio data. It has nothing to do to the frequency of the audio data. You can generate a 1000Hz sine wave, sample it at 44kHz and play it back: what you hear is still the 1000Hz sound. Take a look here. Actually there is a reason to why CD audio data is sampled at ...

I understand why sampling at 44.1 KHz means frequencies approaching 22.05 KHz begin to suffer from sampling artifacts due to the Nyquist theorem. I can't tell the difference, to be honest. Once my signal leaves the digital domain, my analogue gear probably masks a lot of the details in this region and my hearing is definitely not ...

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 ...

This site has a decent table about what the standards are for different HD formats. In general, lower sample rates will have an lower frequency ceiling due to the Nyquist theory, but the average upper bound for human hearing is somewhere between 18k and 20k, so 48k should easily contain all the information necessary for human consumption. The only real ...

I tend to prefer something higher than 44.1KHz because (like you said) it's pretty close to the minimum useful frequency (according to Nyquist), and I try not to assume that I'm going to be using the audio in the same way all the time. The best example I can think of is timestretching algorithms. I'm a big fan of Ableton Live's warping, and at some point ...

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 ...

Pelle ten Cate's answer is great. Another possible concern is that the ultrasonic frequencies might become useful later in processing, even though you can't hear them directly in the recording. If you decide to slow something down for whatever reason, those ultrasonic frequencies will become audible frequencies. If you had filtered them out, the slowed ...

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 ...

Well, it seems like you have to forget everything you know about sampling frequency and frequencies in general. The frequency in terms of audio is the number of times the speaker membrane moves in and out per second. What makes it vibrate is that the power that is sent through the audio cable makes electromagnetism that pushes and pulls to the magnet inside ...

independently of hearing or not, there's the nyquist frequency, but I don't think that that should be to much of a factor to worry about. If you do hear the difference then do something about it, if you don't just let it be - it'll be less drain on your hard drive, RAM and processor, not to mention have to re-apply algorithms when bouncing etc. And why? ...

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 ...

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.

There isn't a single, simple answer for this question. This forum post explains: You can estimate the approximate Audio information [1] content of FM by using the bandwidth and dynamic range. FM pretty much tops out at about 15K and the dynamic range is about 70db. Each bit represents approx 6db so we need just under 12 bits and a sampling rate of 30K. ...

By slow I mean the regular notes have a sort of bite and feeling as if someone were strumming them ridiculously fast (best way I can describe it). When they are played in this low note range the reverberation or super fast strumming effect slows down more and more the lower it gets. Its almost analogous to someone applying and LFO to the bass at ...

The Nyquist-Shannon theorem indicates that the highest frequency that can be sampled is 1/2 of the sampling frequency. However, it does not follow that you can simply adjust your sample rate to extract lower frequencies. The reason for this is aliasing: when a signal is sampled at a rate less than twice the greatest frequency that occurs in the signal, the ...

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 ...

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 ...

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 ...

"48kHz sample rate, A-wtd" is just telling you under what conditions the "114dB" dynamic range reading was achieved. It doesn't really tell you a-lot in all honesty, however almost all modern audio interfaces (including this one) will be more than cable of a decent quality recording and sound reproduction. In regards to sample rates it's not something to ...

As ObscureRobot is getting to, your problem is coming from interferences, particularly from frequencies you can't hear. As you play lower, the frequency difference (in Hz) between the octaves (or whatever harmonic you have playing) decreases, and therefore the apparent LFO rate decreases. For example, An A octave has one note at 220 and another at 440. ...

Good answer Mr Shunz. The Wikipedia article you included has this passage: ... however in some cases ultrasonic sounds do interact with and modulate the audible part of the frequency spectrum (intermodulation distortion). For an explanation of Intermodulation Distortion (IMD), see ...

Yes, they are independent of the bitrate and they both directly affect the "quality" of a digital recording. Sampling Rate At a basic level, digital audio is recorded by approximating the positions of an audio signal over time. The Wikipedia article on sampling rate has a pretty good picture of how this works, which I'll link here: You can see that the ...

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 ...

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 ...

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 & ...

I have asked this question to myself a few times in the past, too. If I can be a bit objective after I gained more experience: The difference between 44.1 or 48.2 and 99.6 will never, ever, amount to 0.2% of the difference between an inspired or non-inspired performance that's captured, or whether it's played in a good or a bad instrument. Bit rate is yes ...

You get far more mileage out of going to 24 bit than you do by increasing the sampling frequency, especially when multitracking. The only reason I would go above 44.1 kHz is if I was especially interested in preserving the phase characteristics of very high frequency sounds (cymbals, for example), since the anti-aliasing filters do not have to be so steep ...

If you can't tell the difference, I wouldn't sweat it. You may, someday, be able to tell the difference, but if you can't now, and your clients can't, I think this is a premature optimization of your setup. There may be a technical reason to do so, such as interchange with other systems/people.

I generate audio in the best quality possible, this means for my equipment 24bit/96 kHz and I am "only" doing a podcast. Reason for this is to have all possibilities open for processing after recording. If this is too much data for you to process, the first step could be to copy the original data to a file with less quality. Please check page 15 of this ...