Two methods have been presented for synchronizing digital audio transmissions, i.e. for connecting digital audio systems with nominally equal sampling rates. One is based on FIFD buffering and the removal or lengthening of low-level signals: the other is sampling frequency conversion. A far more efficient method, Sampling Rate Synchronization, is proposed, based on a new category of digital audio filters. Its hardware implementation is simple, its total delay very short, and it is well suited for insertion in digital audio systems with auxiliary data.
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