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- From: matthews@eecs.ucdavis.edu (Thomas W. Matthews)
- Newsgroups: rec.audio
- Subject: Re: Anti-aliasing on the recording end?
- Message-ID: <21718@ucdavis.ucdavis.edu>
- Date: 23 Jan 93 21:17:09 GMT
- References: <shetline-210193103508@128.89.19.74>
- Sender: usenet@ucdavis.ucdavis.edu
- Organization: Division of Electrical Engineering and Computer Science, UC Davis
- Lines: 31
-
- In article <shetline-210193103508@128.89.19.74>, shetline@bbn.com (Kerry Shetline) writes:
- |> It has obviously become popular in digital playback systems to use
- |> oversampling as a way to provide digital-domain filtering, allowing for the
- |> use of much less severe analog filters. But what's going on these days on
- |> the recording side?
- |>
- |> I realize that the answer to this question would most likely depend on the
- |> application. I imagine that most consumer DAT, DCC, or MD units employ
- |> steep input filters and call the job done. However, I suspect that some
- |> (most?) of the pro gear, and maybe even some high-end consumer stuff, would
- |> use higher sampling rates (>50KHz) with more gentle filters.
- |>
- |> The problem is, you'd need *real* higher sampling rates (as opposed to
- |> computed oversampling), and the processing power to perform the sampling
- |> rate conversion with digital filtering.
- |>
- |> Anyone out there got the scoop on this?
- |>
- |> -Kerry
-
- I have been wondering about this too. I haven't had much
- time to think about it. I think delta-sigma A/D converters
- sample at a much higher rate than 44.1 KHz, and with fewer
- than 16 bits. I once thought this through and understood
- how this accomplished noise-shaping, and finally gives
- 16-bit samples at 44.1 KHz. I wonder if the faster sampling
- can also be exploited to ease the constraints on the analog
- anti-aliasing filter?
-
- Tom Matthews
-
-