I have read that the
optimum sample rate to record at is 88.2kHz. The reasons include simple
integer-ratio sample-rate conversion, avoiding the phase shifts and
ringing of anti-alias filtering at 20kHz, and less data to move about
compared to 176.4kHz. Is there any truth in these assertions?
Via SOS web site
SOS
Technical Editor Hugh Robjohns replies: These claims are partially true!
Let’s start with the simple integer-ratio sample-rate conversion issue.
Simple ratios were important in the days of ‘synchronous’ sample-rate
conversion, but that technology went the way of the dodo a long time
ago. It was a relatively simplistic approach that did work best with
simple ratios between the source and destination sample rates. However,
it had limited resolution in terms of the practical word-lengths
achievable, with the noise floor rarely being better than the equivalent
of 18 or 19 bits. Moreover, the approach is hugely wasteful of
computational effort, calculating millions of intermediate sample values
no one has any interest in.
Modern
‘asynchronous’ sample-rate conversion is far more sophisticated and
works by analysing the source and destination sample rates and working
out only the required sample values with huge precision. This achieves
a technical performance that is significantly in excess of any
real-world converter and very close to the 24-bit theoretical level —
and that’s achieved with any ratio of input-to-output sample rate. There
is no measurable difference in performance between using simple integer
ratios or complex ones.
In fact, it’s
interesting to note that some of the best performing D-A and A-D
converters from the likes of Benchmark, Crookwood, Cranesong, Drawmer,
and others, all use non-integer sample-rate conversion as an inherent
part of their jitter-isolation process. For example, D-A converters
using this approach typically up-sample the incoming digital audio to
something like 210kHz, or the rate at which the physical D-A converter
chip achieves its best performance figures: no simple-ratio conversions
going on there, yet class-leading performance specifications!
Moving
on to the second point, there is a (weak) argument for sampling
original material at a rate higher than 44.1kHz in some cases. The
reason is that a lot of A-D systems are designed with relatively
imprecise anti-alias filtering, which typically only manages 6dB
attenuation at half the sampling rate, instead of the 100dB or more that
is theoretically required. It’s done in that way because it makes the
A-D converter’s digital filtering a lot easier to design, and in most
cases it makes little difference. However, aliasing could result if the
input audio contains a lot of strong extreme HF harmonics. Cymbals,
orchestral strings and brass can all generate enough HF energy, if
close-miked, to cause this problem with some converters. In such cases,
switching to a higher sample rate might sound noticeably sweeter.
The
issue then, of course, is how to down-convert to 44.1kHz for release
without suffering the same problem in the sample-rate converter (SRC)
anti-alias filtering. Clearly, a properly designed digital filter is
required in the SRC, and while some software SRCs do this properly, some
don’t. The Infinite Wave SRC comparison web site reveals the scary
truth! (See http://src.infinitewave.ca).
The
176kHz (or 192kHz) quad-sample-rate idea is really just about being
able to say ‘mine’s bigger than yours’. There’s a very good white paper
about sampling theory on Lavry’s web site (www.lavryengineering.com)
where Dan Lavry points out that the higher the sample rate, the greater
the proportion of error in sampling time and the lower the actual audio
resolution. Lavry argues (very sensibly, in my opinion) that the
optimal sample rate would actually be 60kHz. In the real world, 96kHz
can be useful, for the reasons mentioned above, but the quad rates are
a folly and Lavry refuses to support them!
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