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RE: What if you don't care about 24/192?

The Westcott pdf was the one Ed posted and what my comment was originally aimed at.

His main point is in relation to the "available" bandwidth relative to the theoretical Fs/2 bandwidth when implementing a practical system. However, whilst this is absolutely true, the concept of inadequate "accuracy" when applied to audio recording of vinyl (just to make sure this post is still considered Vinyl related!!) is somewhat misplaced especially in the context of the ADCs available for the audiophile. The Lavry paper provides the mathematical analysis to prove that this is simply not the case.

ADCs such as the PCM4202, 4220 etc utilise sigma-delta modulators running at a very high oversampling ratio dependent on sample rate (for example 128Fs for 44.1kHz, 32Fs at 192kHz) thus reducing the required order of the input buffer filter. The oversampling ratio is, of course, an inherent requirement given the limited number of bits in the quantiser in order to achieve the desired noise performance after the application of noise-shaping.
The PCM4220 datasheet recommends (at a minimum) a first order filter. Obviously for better anti-aliasing performance a higher order may be desirable. However, in my view, this does render the argument presented in the Westcott document somewhat irrelevant to the discussion on optimum sample rate for vinyl transcription. The output word at the desired Fs is the result of the decimation filter and therefore not subject (to the same extent) to the issues that Westcott asserts.

Therefore choosing Fs significantly greater than the required value for the signal offers no advantage beyond the reduction in ringing time for the reconstruction filter of an impulse response. The audibility of the filter impulse response depends on the cumulative effect of ALL filters from recording to the final analogue output stage.

As an extreme example, the Benchmark ADC1 runs the ADC at a sampling rate of 216kHz (IIRC) irrespective of the desired Fs. The desired output sample rate is then the result of passing the data through an SRC. Therefore the "accuracy" of the information is not dependent on the selected sample rate for recording, but more on the SRC algorithm...but that's another subject altogether!
You therefore only need to choose Fs based on the bandwidth requirements of your signal. In principle a high Fs would always seem best, but there are tradeoffs based on the clock accuracy and jitter which Westcott completely ignores.

On to the phono stage, I'm still trying to decide on my approach. To ensure good overload margin, I started off on the Single Gain stage/Active RIAA approach and I started off looking at DC servo based designs. Then I saw the elegantly simple Holman design presented in his paper to the AES - "New Factors in Phonograph Preamplifier Design" AES May 1976, Vol 24, Number 4.



Regards Anthony

"Beauty is Truth, Truth Beauty.." Keats

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