Thank you for this explanation.

this is helpful . thank you so much

Incredible timing, thank you Hans! The DAC I am considering has upsampling features but had no idea if these were a gimmick or might add value. As usual your videos give enough information for one to make a decision oneself, rather than spoon feeding a one-size-fits-all answer. Kudos, and love your channel all the way from Australia :)

That's a very interesting one, thanks. I only recently learned about how and why upsampling works (from your channel, Hans, when you were talking about DAC filtering in a previous episode), and when I saw an upsampling Cambridge Audio DacMagic Plus going for a good price, I got it. My existing DAC was a DacMagic 100, which sounds good to me, but even with my modest system the DacMagic Plus really does sound better. I haven't decided which filter option I prefer, but they do sound different. I haven't tried software upsampling on my computer yet, but I'll give that a try too.

Thanks for the explanation. I upsample on a computer running Roon and find red book becomes smoother and more expansive when upsampled to DSD128 which is the max my Ultrarendu can handle in combination with a T+A dac8.

Great video... Are there any plans on reviewing the new 129$ ifi mqa capable DAC?

Thank you Hans. Today I was attacked in one of the Facebook groups for sharing my joy in realizing that upsampling made things sound smoother, more relaxed and better. I'm a snake oil follower, they say. Well, that won't change my impressions on how good it sounds.

Yes wholeheartedly agreed, upsampling with right modulator and noise shaper for given music genre and adequate combination of the above for the final output sampling rate can work wonders, but it can also hurt the music or the staging if misused, also pulse phase manipulation is important, and I like minimum phase for studio and acoustic music and linear phase for orchestra or classical recordings, Now, how about a video for "bits are bits" crowd? I've worked in IT and Medical environment long enough to know 0&1 are not universally perfect from A to B, and in this case, people look into wrong areas like strictly transports/protocol details to back their claim, entirely omitting the fact that EMI / RFI polluted environment or poorly screened cables or device topology might cause bit flipping and ECC mechanism will not be able to replace corrupted bits as distortion occurs outside its area of operation... Maybe this is something you would consider talking about, as I would rather spend my energy enjoying great-sounding music than evangelise stubborn individuals who know better and can be resistant to broadening the knowledge 😉

Hey Hans, it's your Scorpio brother again haha, so I'm very lucky to have found your channel. I've made some updates to my system based on your channel. Latest we spoke about was my sms200 neo and sBooster. They are breaking in now. With this latest addition to my setup I've never adjusted my filter settings in Audirvana. Last night I did some searching but it's way above my understanding. For now I've set my Nyquist to 88 and phase filter down to 30, some say set it to 0. Am I heading in the generally right direction based on theory?

"I prefer minimum phase with smooth rolloff" can someone tell me if there are any issues with linear phase filters? Without having listned to a range, it makes more sense to me...

can you make an example of a good upsamling setup.? one with roon and one without. :)

Does it make sense to upsample with DSD in roon (my dac supports DSD resampling to 256) and then upsample from 48 to x 2 or x 4 in the DAC aswell?

What are your thoughts on this subject if there is a miniDSP SHD Studio in the chain between my computer and DAC? I'm thinking of buying one. The miniDSP's internal sampling rate is 96Khz so I'm not sure what happens to an upsampled digital signal it receives and what it does upon output to the DAC.

You are a knowledgeable and smart guy for sure!

Nice review and I have one question, I use the Devialet 440 Pro Ci with Wilson Sasha2, Nordost Tyr2 cables, dCS bridge and Metronome T5 transport. The Devialet is already very resolving but I’m questioning myself if I would get significant improvements adding a dCS Upsampling to my system?

When using the iZotope 64bit SRC converter in Audirvana you can choose for forced upsampling, but under Advanced parameters you also have the possibility to manipulate the Cutoff frequency from the default 1,00 NYQUIST up to a level of 1,50 or even 2,00 x Nyquist. The effects of playing around with these optional Nyquist parameter are clearly obvious, but I have no idea, no understanding at all, about what is actually happening. Is this variable Nyquist parameter in any way related to the subject of upsampling as well?

Great video. Would upsampling Tidal 44.1 to say 176.4 (from Windows 10) be better or to say DSD 256 ( through say Audirvana SW) be better sounding theoretically? Thanks

Excellent and logical communication of concepts as always - great explanation!

Does up-sampling help with a DAC like the Schiit Multibit DAC or R2R ladder dacs?

Are you using a audiophile fuse in your external dac power supply?

Have you heard NOS DACs and what are the advantages? Many audiophiles prefer these non-oversampling DACs and claim more relaxed, natural tonality and spaciousness without the tiring gritty sound of over sampling ones.

You did not mention the issue of high res audio and the output bandwidth of the DAC. If the DAC supports so called high-res audio, it will be designed to accept high (>44.1khz) sample rates as native audio, not assuming it is simply upsampled audio to help the analog filter. If you upsample a 44.1khz file to, say, 88.2Khz, and send that to an 88.2khz capable DAC, the DAC will surely assume that this is native 88.2Khz audio, and that it should start filtering at just above 44khz, in order to output an analog signal with a 40khz bandwidth, rather than filtering at just above 22khz in order to output a signal with a 20khz bandwidth. In this case, all the audio between 20-40khz will be output to your amplifier, causing the amplifier to produce substantial intermodulation distortion within the proper 20-20Khz audio band. This is unlikely to help the sound quality. Amplifiers nearly always produce more distortion at high frequencies than at low ones. It seems to me that the technique you are espusing only works if you use a DAC which can be controlled in such a way as to accept an upsampled high sampling rate signal (such as 88.2Khz or 176Khz) but to filter using a filter starting at 44.1Khz. Is there such a DAC?