Up, over, and resampling

It hasn't helped anything in 20 years.  It may have a modest effect on the top octave of the DAC's frequency response.

Imagine two data points on a musical signal on one channel.  The first point is 40, the next point is 48.  The raw data:

40, 48

4x Oversampling:

40, 40, 40, 40, 48, 48, 48, 48....

Upsampling to 176.4 kHz

40, 42, 44, 46, 48....

Asynchronous Sample Rate Conversion is like upsampling to any given target.  It does the MOST manipulation of data:

39.5, 42, 44, 47, 49.5 .....

All these techniques were really good before 2000 as DAC's performed so much better with high rez (88.1kHz or higher) data.  In the last few years DACs perform so much better with Redbook (CD, 44.1kHz/16 bit) than the previous generations.

This is a really good overview on oversampling / upsampling from Analog Devices:

https://www.analog.com/media/en/training-seminars/tutorials/mt-017.pdf

If you need more information, I suggest using the keywords "multirate digital signal processing" when you do your online search.

I’m not sure I’m understanding where you’re going here- I had a quick look at the Aries G2 manual (couldn’t find for G1) and the only mention of “upsampling” had to do with streaming MQA files. Apparently there is a setting that will employ this action during the “unfold”.

As for passing files through I don’t believe this unit “upsamples” PCM native 16 bit to 32 but rather “supports 32 bit files” when output via USB. The standard SPDIF formats (coax,Toslink and AES) support up to the standard 24/192.

Either way- not sure how your planning on configuring your NAS storage drives or if you’re streaming across a network from another ripper/playback unit but I would always rip the music in its native format.
 

If I remember correctly the Soekris will not upsample and will accept files up to 24/192. I’ve no experience with Chord but I believe some of their units will upsample- just not sure if the Qutest will or not. 

@yage thanks for the link, I’ll need to carve out some time to focus on it properly

The current process of ripping was just what prompted me to look at the manual and discover new things. Ripping is done at native redbook to FLAC.

I find the three terms upsampling, oversampling, and resampling confusing because while I’m fairly certain they have distinct technical meanings, they are often used interchangeably. If I use one wrong then please correct me. I’m trying to get a better handle of this.

The Aries G1.1 upsamples all files, adding null bits to make bigger words. When sending S/PIDF 16 bit becomes 24 bit. For USB all samples become 32bit. This is independent of source, library files or streaming services, all are upsampled. This would be like adding trailing zeros after a decimal point. The number is unchanged but the available precision is greater for subsequent manipulation, like volume control or DSP.

The oversampling is usually a multiple of the native rate  44.1kHz can become 88.2, 176.4, or even 352.8kHz which is 8x over USB. (Similarly 48kHz can become 96,192, or a max of 384kHz.) At its crudest existing value repetition could be used, but I think the more sophisticated approaches interpolate intermediate values or try to recreate more of the original waveform by complex iterative algorithms. No idea what method the Aries uses or the computing power required. I think some DACs perform a similar manipulation allowing the use of less harsh antialiasing filters that keep artifacts well beyond human hearing

Resampling I think is both simultaneously and not necessarily in multiples and can have more issues. Changing sampling rates by non multiple factors seems like it could be especially problematic. That might be why so many devices have multiple clocks, one for 44.1kHz multiples and one for 48kHz multiples

i haven’t tried the USB with the Soekris  The Qutest is supposed to be galvanicly isolated and have a great USB input. It might be the change in streamer or going to USB or a magic cable, but there are a lot of cliched impressions I could share about first trying the Aries Qutest combo after using a Node2i for almost 4 years, 3 of them with the Qutest. Node2i will find use somewhere. Too good not to use, but not worth selling.

Ultimately I want to pick the best set of parameters for me based on theory and verified by listening (even though I’m not sure what to listen for). Hoping that if I understand it better then I’ll be less likely to do something incredibly boneheaded in the future if there are changes in affordable tech  

Upsampling and oversampling with respect to audio playback is essentially the same function. In the DSP world you can usually find it explained under the topic of multirate interpolation. The incoming data is zero-padded to the desired sampling rate and the digital filter interpolates the values. The interpolation step is not a ’guess’, it’s more akin to a ’recovery’ of the sample values as if you had originally sampled the signal at the higher frequency. When the interpolation filter’s impulse response is convolved with the audio data, the filter coefficients multiplied with the other samples in the signal along the length of the filter will essentially fill in those zero-padded positions.

Resampling by a non-integer factor doesn’t involve the system clock. It is upsampling (zero-padding) by an integer factor followed by an anti-aliasing filter and downsampling by another integer factor. The resampling ratio is given by the upsampling factor divided by the downsampling factor. Here is a link to a good overview - https://www.eetimes.com/multirate-dsp-part-2-noninteger-sampling-factors/

The computational load is based on how many coefficients the digital filter has (i.e. the length). The ideal filter for audio purposes is the sinc function. However, it is impossible to implement because it would have an infinite number of coefficients. The digital filter is always an approximation and there are many approaches and algorithms to construct these filters. Also note that the longer the filter, the more delay it introduces into the reproduction process.

Here is a free online book on DSP that you’ll probably find useful:

https://www.dspguide.com/

Here are some informative videos that explain some of the fundamental concepts of digital video / audio:

Digital Media Primer for Geeks

Digital Show and Tell

The oversampling is usually a multiple of the native rate 44.1kHz can become 88.2, 176.4, or even 352.8kHz which is 8x over USB. (Similarly 48kHz can become 96,192, or a max of 384kHz.) At its crudest existing value repetition could be used,

OP: Well wish you had read my post. This is the ONLY definition of OVERsampling there is. The original values are repeated. This requires no processing but improves the behavior of the output filters. If it isn’t doing exactly this it is something else.

The crudest, best and average oversampling systems all do this. The only variable is the number of oversamples per original.

@cat_doorman 

I would do a serious listening test of the same CD with different sample rates. Don't think there will much difference if any.

Depending on the size of your collection (and other CDs you may borrow) you will end up with a sizable amount of data.  I understand storage is cheap and getting cheaper but I would also start out with two copies (anticipating failure!).  I'm kind of a belt and suspenders guy.

Regards,

barts

@erik_squires

Value repetition cannot be used. Think about it. If you simply copy forward values to the new sample positions, you’ve stretched the signal out in time. Zero stuffing the audio data is how oversampling / upsampling is performed on playback because we do not have the information on what value the signal is in that moment in time. The digital interpolation filter will calculate the values in the zero stuffed positions and output a valid audio signal.

Please read the following on multirate interpolation - https://www.staff.ncl.ac.uk/oliver.hinton/eee305/Chapter9.pdf

https://www.stereophile.com/asweseeit/344/index.html

Whoops my bad. I only got Oversampling half way correct. It does NOT interpolate, that part I was correct about, what I missed is that the intermediate samples are all at 0.

Also, I agree with the following statement, that the best thing these filters do is slightly tweak the top octaves:

So while I strongly suspected that the improvement I heard with the dCS 972 was simply due to its using a different oversampling filter, along with the benefit of better downstream DAC behavior when fed a 24-bit rather than a 16-bit signal—as I described in my January 1996 review of the Meridian 518—I wasn't sufficiently sure to spill ink on the subject.

Non oversampling ( Audio Mirror) is sooooooo Sweeeeeeeeeet.

@yage I appreciate the extra homework. A reading list is what I was ideally looking for, There’s too much about the subject (even if you exclude disinformation) to cover it properly on one of these threads. I’ve skimmed through some talks by Rob Watts, but I assume having a better background prior will make those flow a little better.

@barts the CDs are being ripped to lossless FLAC. Any manipulation beyond redbook isn’t saved to disc. I had done ALAC before when I was still using a lot of iTunes, but that disc died a tragic death so I only have AAC copies of most of what was there. Those are fine for my phone, but not for my main rig. The first time I ripped was to fit all my CDs to an iPod, the old kind with a HD. Then I tried local streaming, lossless files over Airplay, but could never get the Airport Express to behave. Reboot at least once a day. Now after a few years of streaming lossless from Qobuz, it’s time to get everything on the streamer SSD. And I will back it up. I also need to decide which of the albums I mistakenly bought off iTunes as mp3s are worth buying again either on CD or digitally.