Why NOS actually may make sense.

[DF96]

For most audiophiles, listening experience can only tell them whether they like a particular sound. This tells us almost nothing about to what extent that sound is faithful to the original source. For that you need theory and measurements, combined with results obtained from well-controlled ears-only tests.

Theory tells us that filterless NOS cannot reproduce the original signal without HF droop and accompanying images. Theory cannot tell us whether some people might prefer this lack of reproduction.

 

[hollowman]

No. of taps

ESS also supports the implementation of custom filters by uploading custom coefficients, but the number of taps that are supported is not what you would call impressive.

Somewhere on Stereophile's website is a video interview with chief designer of mbl (German high-end company).
In it he briefly discusses no. of taps. Bottom line being something of a Goldilocks -- not too many, not too few.

Refs:

A Conversation with Juergen Reis (MBL) & John Atkinson (Stereophile) -- part 1 --YouTube

A Conversation with Juergen Reis (MBL) & John Atkinson (Stereophile) | AudioStream - YouTube

 

[Ken Newton]

Somewhere on Stereophile's website is a video interview with chief designer of mbl (German high-end company).
In it he briefly discusses no. of taps. Bottom line being something of a Goldilocks -- not too many, not too few.

The number of 'taps' refers to the effective number of MAC (Multiply- ACcumulate) computations utilized by an particular FIR digital filter engine in calculating each sample value output by the filter. The key correlation being, the greater the number of taps the sharper can be the filter's frequency cut-off slope. As you might imagine, this is of particular importance to brickwall anti-alias and anti-image filter implementations.

A Goldilocks philosophy regarding the number of filter taps translates into a Goldilocks philosophy regarding filter slope sharpness. Many commercial DACs feature a soft sloping digital filter mode.

 

[hollowman]

A Goldilocks philosophy regarding the number of filter taps translates into a Goldilocks philosophy regarding filter slope sharpness. Many commercial DACs feature a soft sloping digital filter mode.

If all were perfect, the more the merrier.
However, the greater the number of taps, the higher demands on power-supply and regulation. Some of you may recall early digital filters (SAA7220, etc.) that gobbed massive current (200mA) and ran hot.
Add to that more computation increases things like transistor-switching noise.
Don't ask me what the Goldilocks zone is. Maybe keep adding taps (in that FPGA) 'till it sounds less good, then subtract a few ...

 

[MarcelvdG]

One advantage of NOS DACs over most of the commercial oversampling DACs has not been mentioned yet: most commercial oversampling filters have no headroom for overshoots and clip when they are subjected to peak sample normalized audio recordings.

Audio That Goes to 11 - Benchmark Media Systems, Inc.
Why "Audio Goes to 11" - Benchmark Media Systems, Inc.
Intersample Overs in CD Recordings - Benchmark Media Systems, Inc.

Possible solutions are digital volume control before any interpolation or sample rate conversion takes place, replay gain tagging, renormalizing recordings to a lower level or post 34 of this thread:
Solving the intersample DAC clipping problem for about ten euros
Mind you, it is neither tried nor tested.

 

[NATDBERG]

The system has already been abruptly, sharply bandlimited by the anti-alias filter prior to sampling. If there was spectral energy beyond Fs/2, it was removed ... sharply ... to prevent aliasing. If there was no spectral energy beyond Fs/2, the anti-alias filter had no effect (in frequency or time). In short, the ringing is already "baked-in" ... as it must be, in any sampled-data system ... long before any post-DAC filtering.

Sure, some may find some pleasing colorations from rolled-off treble, ultrasonic images or nonlinear phase distortion. But, at least in the ideal sense, there's only one way to preserve magnitude and phase accuracy below Fs/2.

Many modern studio ADCs use relaxed anti-alias filters for high sample rates so nothing is baked in if you use the right ADC..

This is the main reason for high sample rate recordings in my opinion, which can then be played back well on NOS dacs - you can get a TDA1541 to play 386KHz via Ian Canada's projects for example (you just get a bit truncation problem instead i guess, 24 -> 16).