Originally posted by soundhound:
... PCM systems use A/D and D/A converters using "1 bit" technology and megahertz sampling rates because it is a very reliable, stable and cost effective way to convert analog to digital and digital to analog...
Some basic and probably ignorant questions, but questions that I believe will help many people like myself understand digital signals in sound reproduction...
... Could someone describe the basic digital format attributes (or data structure) for sound encoding?
For a single audio track, assuming a typically wide frequency range (10Hz - 30kHz), with lots of instruments playing at the same time, how is this information stored digitally?
For example, if a typical CD is recorded in 44.1 kHz/16-bit, is there a complete picture of the audio spectrum being created 44,100 times a second? Does the does the bit depth describe amplitude only, frequency, both, or even more?
I believe that 44.1, 88.2, 96 & 192 kHz are sampling rates, but what are the attributes being sampled at this rate? Frequency; amplitude; both, or something else? Soundhound refers to megahertz sampling rates (1,000 kHz); how are these comparable? Are higher sampling rates useful for oversampling/error correction, to be used during the read/processing of the digital signal? It would seem reasonable that high sampling rates might be especially helpful in accurate description of high frequencies, but is it true?
... And I believe that a 24-bit recording can provide 256 x (times) the incremental information than a 16-bit signal... but what attributes are being stored here? ... And of what value are those extra 256 x (times) incremental steps? I understand that oversampling can be useful in error correction, is oversampling information contained in the extra bit depth? ...Does the greater bit depth provide just more information, or a way of verifying accuracy of the information, or both?
...And the attributes themselves... are discrete frequencies described digitally, or are there narrow frequency format slots (or bins) for frequencies throughout the audio spectrum, with each slot (bin) being so narrow that it closely approximates a smooth frequency/amplitude curve? I would guess that each frequency slot (bin) would have an amplitude assigned to it.
Or is the sampling rate high enough so that the bit depth just describes amplitude (or what else)?
Would a good analogy be a tape recording machine... where tape speed is comparable to sample rate, and tape density (dynamic range) is comparable to bit depth?
I would greatly appreciate a short primer, or a reference to a novice friendly source for this information.
Thank you...
Allan