DCAudioDIY.com

This was my understanding as well.

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You and Roscoe have any specifics of DSD filters that the DAC applies. Do all DACs behave the same?

It's been a long time since I read up on DSD, but as I recall they have to use noise shaping to help minimize quantization errors. I'm sure there is lots of information on the subject on-line.

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Well, my phrasing was inapt, but the idea is to have the software upsample the files to your DAC's highest native bit rate and thereby, I assume, reduce the DAC's influence on the digital stream. I don't claim to be an expert at all but that's my understanding.

I'm not an expert either. However, I suspect it's not an issue that has an all encompassing best approach. At the very least the results will depend on your DAC and whatever you may use to upsample before sending the bit stream to the DAC. Some DACs may sound best with a bit perfect stream of the original, some may benefit from upsampling before hand. I've done it both ways over the years, but have settled on sending the original bits to my Holo Audio Spring DAC and letting it convert them into analog. I'm quite happy with the results.

If I'm going into my DEQX, I up/down sample everything to 96k since that's what it uses internally for all its processing. The DEQX I'm using will accept up to 24/192, but still processes at 24/96. I've also found that it can take a noticable amount of time to switch between some bit rates and miss the first note or two of the new bit rate track. Feeding it a steady diet of 24/96 eliminates that problem.

In any event, my opinion is your mileage will very likely vary from anyone else.

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I agree with you, Dave. I like the way my DAC sounds whatever it is fed and I haven't detected any real benefits from upsampling. The fact that these software upsamplig parameters are usually adjustable is another reason I'm not eager to fool with it.

However, my experience with the Chord M Scaler is that it makes a significant difference in the sound of 44/16 recordings. It is not simple upsampling, but a very high sample rate digital fit to original digital data points, basically like a spline curve fit based on several seconds window of data at a time. Basically trying to work backward from the Nyquist sampled data to recover the original waveform. It seems to recover more detail, spaciousness, with a reduction or elimination of digital grain. It may be subtle at times, but difficult to listen without it.

David

HQPlayer manual explains why software resamplers are needed.



That is an excellent justification. Technology is continuously improving. Upgrading SOTA DAC every 2/3 years is an expensive proposition. However, a software like HQPlayer which costs a couple hundred dollars will be updated periodically by its publisher. So, a software solution with "suboptimal" DAC which gives close to "optimal" DAC performance at fraction of the cost and continuous software updates makes lot of sense to me.

Most recent generation DAC's have multiple digital filter settings accessible to the user. They all sound different to me. Linear Phase, Minimum Phase and Apodising filter types and mixes of them.

I listen to them all and decide on which I like the most.

This is the same for upsampling/oversampling rates to the DAC as there are more filters involved. The higher sample rates move the digital artifacts higher out of the audio band. Most filters also use dither to increase the bit depth from 16bits to 24 or 32bits. No extra data, just a noise floor that works better with filters.

In the case of things like crossover filters, it also moved the digital fold over frequency for the digital filter implementation higher out of the audible band. Why I like DSP systems running at 192KHz vs my old DEQX that runs at 96KHz. Never got on with most of their crossover filter types as well.

Lots of filters in the digital chain to get what you want.

I believe you.