Many applications in digital audio raise the issue of changing the sampling frequency of a digital audio signal without otherwise modifying its audio content or its pitch. Economics and signal quality make it intuitively appealing to look for a purely digital solution. The requirements for such a solution are severe. For generality, arbitrary sampling frequencies should be tolerated, including a prior unknown sampling frequencies which may drift with time. In addition, the signal quality of 16-bit and possibly even 24-bit digital audio must be preserved in conversion. Finally, automatic bandwidth adjustment as a function of the sampling frequencies is a natural requirement if aliasing effects are to be prevented. A new approach to sampling frequency conversion, based exclusively on digital signal processing and meeting all the above requirements, is presented.
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