First, there is no such thing as the perfect low-pass filter required by Nyquist’s theorem. Higher sample rates are also widely used, but their necessity is debated. Digital audio: theory and reality. it is *possible* to reconstruct the exact signal up to nyquist (assuming perfect sampling to begin with), but is it likely to be implemented that well in your battery powered portable music device? 44.1kHz would cause a noticeable drift over time, hence 48kHz. (The sampling theorem is clear about this.) Certainly nowhere near as bad as tape hiss or the grunge you get after only two or three copy generations. Local music shops & gear makers are open online and ready to ship your dream gear. And, at the low amplitude end, distortion of small-signal components is very high, contributing to the “brittle” sound that many people describe with digital audio. Digital audio is a representation of an analog audio signal used by computers and digital devices to record and playback sound. Just as analog audio is defined by the values of frequency and amplitude, digital audio has two main two parameters: sample rate and bit depth. Today converter chips are so good, and so cheap, that even $30 sound cards are audibly transparent. Another big problem is finite wordlength effects—we’re using 16-bit samples, not the pure numbers of the Nyquist theorem, so we have to compromise the sample values. Of course a signal cannot be perfectly bandlimited if it’s finite, but that information is more pedantic than practical (the site’s tag line is “Practical signal audio signal processing”), and this article extremely general (not an integration or summation to be seen). there’s also the argument that transients affect the quality of audible frequencies. Your email address will not be published. The Basics Digital Audio at it's most fundamental level is a mathematical representation of a continuous sound. Nice article, and it’s great to see “digital” being defended, though I have a few quibbles: That some people (including me) like the sound of subtle distortion on some types of music is not an argument for preferring analog recording mediums. Watch video series, download programming examples, and connect with Professor Bennett on social media to ask your questions! Why? EH Media LLC. Practical digital audio signal processing. Of course, this alone isn’t a reason to forsake digital’s many conveniences. Yes, every volume (or other) operation adds a tiny amount of distortion, but it’s really REALLY tiny with modern software that processes using 32 bit FP math. No, I don’t disagree one bit. : You could sample a sine wave at twice it’s frequency and get the zero crossing point every time, yes? Proponents claim the ultra-high frequency content subtly increases fidelity and adds “air” to the signal, while critics argue that 44.1 is good enough and that anything higher simply creates larger files and the potential for artifacts when dithering down to lower sample rates. Yes, you are right that calc’s are done in 32-bit float, but accuracy of that calculation is not the problem. Other systems, such as DTS and Dolby Digital, also exist but are more prevalent in the film and technology industries.


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