Upsampling and oversampling 

Excellent explanation, probably the best I found on youtube.

Great video! Very clear, concise and detailed. As always, I learn a lot from you. Good work, Hans!

thanks hans. it's been a long journey for me. I use a Lenovo thinkpad x230 laptop for my source. the shielding in these units is remarkable and the internal filtering of the USB is of a superior standard. my dac is a audiolab mdac + and my headphones are hd700's. the sound I get is more than adequate but I totally agree with you in saying the best equipment to evaluate other equipment are your ears. We all buy equipment made the same on speakers made the same connected by wires made the same. so should the sound be the same. no our ears are all different shapes and sizes with varying degrees of audibility. one of the best upgrades I ever bought was a bottle of otex ear wax cleaner. Used in the normal manner it gave me a better insight into my music collection Anyway thanks for a great channel and thank you for being impartial in your reviews. It's so refreshing to have such a opinion of the equipment you test and yes I'm enjoying the music. thanks

Funny how the energy of the universe swirls about, I just bought a Cambridge DAC magic 100, and I'm very satisfied overall, my Sangean HD tuner sounds best through my IGA for reasons you stated in the video I would say, great video and I am indeed enjoying the music!

Thanks for this explanation. I admit to having been confused about upsampling vs. oversampling. The mathematical smoothing of the data points by interpolation and curve fitting seems logical. We used to do that with other data in experiments on chemical processes at one point in my career. On my DAC, it seems that there is an initial upsampling , and than an averaging of four of these upsampling calculations in the conversion. It does sound very smooth. Just starting to dive in and try to understand this more closely. Of course, MQA has my head buzzing right now and looks like a breakthrough, but I'm still in elementary school on this one. Keep up the good work. We appreciate you sharing your knowledge.

I have to say that setting sampling rate from 44,1kHz to 88,2kHz on my Dragonfly Black really improved sound quality. Apart from improved sound quality I found out that on 44,1k first moments of sound get cut off - for example in Windows testing sound is something like "tu tu tuuu" , but with 44,1k I hear something like "tr tuuu". To me 88,2k sounds better(more clear) than 96k - I guess that's because most of my music is in 44,1k and 96/44,1 doesn't go as well as 88,2/44,1.

Its better from now recording at 96KHz/24bit and you dont need upsampling or oversampling,,Record labels must starting transfer the 60s-90s musics from Analog Recorder (Reel to Reel) and recording digital at 96KHz/24bit. SACD still have very good quality !!

Dank je wel hans!

Cool video, sympathic and wise teacher here. I wonder HOW upsampling is exactly implemented. May be a shifting sinc filter would be great.

thank you for a good video

Brilliant xx

Have you an opinion on whether or not changing the format during upsampling is advantageous or detrimental to SQ? IE PCM>DSD upsampling done in software, such as Audirvana+, provided your computer can handle it. I realize that when upsampling PCM>PCM, you want to stay in multiples of 2. I also realize that DSD can be divided by both 44 and 48, so rounding errors shouldn't be a problem. But, what if anything is lost in the conversion of PCM>DSD? There certainly is a lot of hype about this in the forums. So much so, I'm about to upgrade my converter to one that supports DSD so that I can join in the fun.

What do you think of Computer Upsampling VS a DEDICATED UPSAMPLER DEVICE (NOT the DAC)/An Inline Microcomputer e.g. Raspberi Pi or Jetson Nano? Thanks Very interesting & helpfull video.

thank you