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The limits of human hearing is 20kHz, right? And a 24/192 recording goes up 9 times above that, right? What benefit is that adding? Don't DACs filter out high frequencies anyways? I have not had the opportunity to compare hi-res to redbook, but I am confused by the principle of hi-res audio. Any explanations?
The Nyquist rate is the minimum sampling rate required to avoid aliasing, equal to twice the highest frequency contained within the signal.
That makes sense. I read an article that said 44.1kHz was chosen originally because 20kHz is the limit of human hearing, double it and you avoid aliasing, or something like that. Something about the Nyquist rate too...http://en.wikipedia.org/wiki/Nyquist_rate
I think you answered your own question. 40KHz is the MINIMUM sampling rate to avoid anti-aliasing and above that you then get more samples across the entire audible range, so as to better approximate the analog curve. Thus, hirez wouuld accomplish BOTh objectives,
Technically, sample rate has no effect on dynamic range.
Audiophiles have claimed since the beginning of digital audio that vinyl records on an analog system sound better than digital audio. Indeed, you can find evidence that analog recording and playback equipment can be measured up to 50khz, over twice our threshold of hearing. Here's the great mystery. The theory is that audio energy, even though we don't hear it, exists as has an effect on the lower frequencies we do hear. Back to the Nyquist theory, a 96khz sample rate will translate into potential audio output at 48khz, not too far from the finest analog sound reproduction. This leads one to surmise that the same principle is at work. The audio is improved in a threshold we cannot perceive and it makes what we can hear "better".
Increasing the sampling frequency as it applies here means that you are using more data points to recreate the entire sound (within the whole range of human hearing). It is not just adding more data points that are above our normal range of hearing.
Right, but these two things correlate. If a sound's frequency is, say, 5,000 Hz only and nothing else is present, the sampling rate needed to capture it is 10,000 Hz. You're not going to get any more information out of that one sound by sampling at 100,000 Hz because there isn't anything there.
Again, I'm not sure I follow what you are saying. You give an example of a sound at 5000Hz, as in the frequency of the sound. The other 2 numbers are sampling rates. You can say the 10,000Hz sampling rate means you are recreating that 5000Hz sound using data points taken every 1/10,000th of a second.
Will there be an audible difference in the sound? I dont know. But remember, an analog sound with a frequency of 5000Hz still has a sampling rate with a frequency that is theoretically infinite (i.e. it is a continuous wave function that is not chopped into bits of data).
I'm not sure about anyone else, but it helps me to think of digital audio resolution much the same way as I think of the resolution of a digital photo. If I'm taking a picture of someone's face, I know that I can get more detail into the picture by zooming in. If I'm zooming out, their face only takes up maybe 50% of the image in the photo and I get less facial details (16 bits). If I zoom in, their face may then occupy 80% of the digital image, and I'll get more detail (24 bits). If I want to print that digital image on my printer, I know I can get a better print if I increase the dots per inch (DPI) setting on the printer. If I use a setting of 300 dpi (44.1khz) the picture may look OK from a distance but if I look closely I'll note some missing or blurred information in the picture. The image won't look as sharp. However, if I increase the dpi setting on the printer to 600 (96 khz and up), the image will look much sharper on the paper, and the details will be rendered much more accurately.
