I am no DSD expert (ask @j_j ) but yes it is a one-bit stream. IIRC the idea was to output the delta-sigma's single-bit stream directly then use analog anti-image (low-pass) filters, compared to conventional delta-sigma DACs that use a (mostly) digital filter. It is basically a sigma-delta modulator without the output decimation (low pass) filter. It was originally a 64x bitstream but there are faster versions now. One-bit delta-sigma modulators do have some issues, including higher distortion and susceptibility to "tones" (signal artifacts created by repeating digital patterns in the modulator), that are not a (normally) problem for multibit designs. They also require much higher sampling rates to achieve the same resolution as a multibit design. Virtually all modern delta-sigma converters are multibit designs with digital processing to provide nearly ideal performance.Aha I see . The trend is there less bits larger aperture time , Can i correctly assume DSD as 1 bit ? . Thankyou for your time btw and linking to older articles in ASR archives.
DSD has very high sampling frequency , its used a lot in misleading advertising , no one mentions aperture time which would be a truer measure of resolution in time ? Ad copy only compares fs and call it a win ? It gives you a larger number for your ad .
This ASR at is best real knowledge by experts . I would actually trust J-J and Don here .
This is bit funny (sorry for rambling ).
I or any other audiophile can usually not calculate bandwidth product, bandwidth, gain , or stability margin of an analog circuit of any kind ? (In school 30 years ago i could possible make a crude bode diagram for our control system excercises ).
But we trust some experts when they tell us this amp has a fr response of for example 5-100kHz +- 3dB and are happy acknowledging our limited knowledge and are compelled by the experts arguments .
Likewise I could not check the calculations LIGO has to make get their results but i trust it is peer reviewed and thoroughly scrutinized.
But digital audio ? If the experts outright tells us how sh*t works we should believe them on the same premises , but here many questions everything ? even people who should know better .How can you even design a NOS ladder DAC (for audio ) without any filter and somehow miss that's it's wrong ? I cant design any kind of DAC but even with limited knowledge I can se that this approach has problem .
But fundamentally the time resolution is a function of how closely you can sample the signal, that is how close together in amplitude two points can be. That is a function of converter resolution, i.e. the least-significant bit (lsb) step size, and the signal frequency -- not the sampling rate (though Nyquist must be met). Time resolution is thus how big a step (the lsb, set by the number of bits the converter delivers) and how fast the signal goes through that step (set by the signal frequency).
The link in my signature goes to a post linking a number of introductory articles on sampling and conventional and delta-sigma DACs (among other things) that may provide more background. DSD, like any noise-shaping method, loses resolution as frequency increases (hopefully well above the audio band, thus the need for high sample rates and noise filters).
HTH - Don
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