Is it really useless Upscaling 16/44.1 music to 24

My experience is limited, but I recently purchased a Meridian Explorer USB dac, cheap and it upsamples. Coming between my computer on WMP and my main rig it sounds fantastic. By comparison I think it's better than my Marantz SA 8004 sacd player.

It all depends on why it is done. To do it just to have more bits representing pretty much the same mix is probably a waste objectively.

However I would think it useful if remixing or remastering to achieve a different sonic result is the goal. The results might be considered an artificial artifact of the original but might be able to better emphasize certain aspects of the music in the recording most likely at the expense of others. If done well the results could be significantly different sounding and might well even be deemed better by some. A better recipe per se still using the same ingredients but rearranged in some way that might be tastier in some ways.

Fwiw I use two different non oversampling mhdt dacs and the sound is some of the best digital I have heard.

Oversampling dacs can do well also. I think it' s all about the implementation details and non oversampling dacs seem to have an easier task in order to get things "right".

I've been exploring PCM up-conversion and filter options on an Esoteric K-01X. The unit is far from broken in, but so far is proving that up-converting 16/44.1 can be a good thing. Of the many possible combinations of up-conversion and PCM filtering, selecting PCM-to-DSD or 8x PCM up-sampling sound best and are very close to each other. Up-sampling PCM sounds worse if increased from 4x to 8x in combination with any of the four selectable PCM filters-- but noticeably improved if used without a PCM digital filter.

The choice often comes done to theory versus practical. If a 16/44 DAC does it job perfectly and the brick wall filter is irrelevant, then 16/24 should perfectly reproduce the original sound. However, DACs are not perfect and brick wall filters do matter. The issue is how much they matter. Some DACs sound better with higher sample rates, some do not. It depends on the implementation of the DAC. In fact, 16/44 files externally converted to 24/192 can sound different than 16/44 files fed to the same DAC that internally converts to 24/192.

As to the "adding zeros' comment. I see that a lot. Converting for 16 bit to 24 does add a bunch of zeros to the end of each sample. However, converting 44 to 176 or 192 definitely does more than add zeros. It does an interpolation between the original data points to come up with the additional points. It is not a normal linear interpolation, but rather the surrounding data is fit to a mathematical function that takes into account the local pattern and determines the intermediate points from that function. Sounds complicated but the various algorithms are pretty well known. It just depends on which ones the particular converter uses.

As to JRiver, there are people on their forum asking for different upsampling converters, because they believe there is a difference in them. It can be a lively discussion, because there is far from a consensus on this. The JRIver principles seem to believe that upsampling should only be used if their upsampling is better than what the DAC forces on you. They are of the mind that upsampling for any other reason adds nothing. I would not start this discussion over there unless you want to get lectured.

The idea of upsampling from 44 to 176 rather than 192 is that 176 is an integral multiple of 44 and 2 out of 4 of the original data points can be used. When you convert from 44 to 192 then then all of the data points will be calculated ones. Of course, the result also depends on how you DAC handles 176 versus 192.Some DACs have separate clocks for the two sample rates (44 versus 48) and some have just one clock and generate the other rate mathematically.

The answer here has to be that implementation matters and that you should try it in your system and see if you hear a difference. The results are dependent on the software converter used and the DAC. That said, unless there is really flawed component, the differences will be more subtle than dramatic.


I will say, that there are people that will adamantly state that there can be no difference between 16/44 and the upsampled version. However, I am afraid these people live in a theoretical world, not a practical world where DACs are not perfect and brick wall filters do exist.

As to DSD, there is endless discussion about whether DSD sounds better than PCM. Once again, in my mind, at this point, it comes down to implementation of the DAC rather than whether one format is intrinsically better than the other.

Sorry for being so verbose. This is a topic that I have spent a lot of time thinking about. And, believe me, there is no consensus on it. It trite, but let your ears be your guide.

I listen to tidal and jriver alot and upsample with an esoteric G25 u from 44.1 to 192 so that I give my dac a nice robust 192 signal. It likes that. It plays music better being fed upsampled music. Just the way my K03 is. I have experienced the same with other fine dacs.

Cerrot - I believe you are reclocking as well as upsampling. Have you been able to determne which makes the most difference?

Hi Dtc. I believe there is a reclocking oscillator on the ports. I dont use the clock function of the G25U any longer as the clock in my dac is more accurate. I would love to be able to tap a clock output into my squeezebox (nice project to figure out). I cant separate the two functions but the unit itself makes a world of difference.

Thanks Cerrot. I am always trying to find new information on this complicated topic. The G25U is an interesting option, especially for older DACs with less good jitter protection.

I think outboard upsmpling/reclocking is key to PC audio. It needs all the help it can get. The G25U brght my PC to levels of my K03.

Thanks Cerrot. I am always trying to find new information on this complicated topic. The G25U is an interesting option, especially for older DACs with less good jitter protection.

Sorry for the double post. I was cleaning up old windows and must have reposted by mistake.

May not exactly be relevant but I'm using JRiver to convert everything to DSD and output to a Schiit Loki and, honestly, I have no major complaints, though one minor one. If I'm using the PC to browse the web while I'm listening, I occasionally get a click/pop in the music. In these cases it's not like I'm listening critically so I just look past it. When it's showtime though, it really works a treat, $150 entry fee notwithstanding (not counting JRiver license, of course).

Using Jriver to convert from PCM to DSD takes a lot of computer power, so even on a pretty powerful computer occasional glitches are to be expected if running anything else. The Loki is a steal for DSD playback.

Got it - that's what I figured and it's not that big of a deal really. When I am listening critically, I don't mess with the computer anyway :)

Trcns,

You've gotten very good feedback. I certainly agree with Dtc, and also adding this as another example.

My PC audio goes from a highly optimized PC running Windows Server 2012 in core mode, with Audio Optimizer, JRiver, into Audiophilleo with PurePower, into Metrum Octave. The Metrum is non-oversampling so any upconversion needs to be done at the computer. In this setup I like it better to keep things native. So no upconversion.

Recently I tried Acourate (software for digital room correction, amongst other things) and to convolute it needs to turn everything to 24 bit within JRiver. The combo of convolution/correction plus upconversion I like better than the above.

A few months ago I purchased an exaSound e22 which gets rave comments from the femto clocks and its ability to play DSD256 natively, and all the fuss about DSD. This unit retails for 3.5k, while my AP+Metrum retailed for combined 2k two years ago.

- Feeding the DAC at the native rates I liked the AP+Metrum combo better.
- Upconverting to DSD in JRiver and feeding that to the e22 vs natively feeding into the Metrum, I like the Metrum better.
- Upconverting at the PC with HQPlayer (instead of JRiver) to DSD and feeding the e22 sounded as good as the Metrum. Maybe...maybe a touch better. But not enough to justify the price difference. So I sold the e22.

Goes to show that implementation is the key. Not ditching exaSound. I think it's a very good product, targeted for people who don't have such an optimized PC, not running WS2012 with Optimizer, etc.

And regarding my preference to keep things native with the Metrum, I often wonder if my speakers aren't the bottleneck to hearing the difference (B&W 804S). Another example of implementation being the key!

I would much rather have the hard wear upsample and not the softwear. JRiver (SOFTWARE) upsampling added noise and distortion, in my rig. I upsample 44.1 TO 196K with an esoteric G25 U and it sounds much better than anything Jriver could upsample in my system. I cannot use JRiver for upampling or dsp. i feel the dsp just tries to compensate for other issues further down the audio chain.

I didn't care for JRiver upsampling either. That is why I feed the DAC at the file's native resolution. However, I realize there are many factors to consider. Some DACs might sound better at higher sampling frequencies, some computers might be noisier, and some might not want to spend in a G25 - for example. But generalizations are tough to get right.

I am running a 10 meter spdif cable from my PC (separate room) into my sound room so the G25U is kind of like a signal amp for me. I guess today, its an upsampler/reclocker in the digital world. Its worked splendidly. I think if I had the puter in the same room (i would blow my brains out) I may feed the hi res signals natively and bypass the g25u but it does a good job with intrnal bypass circuit, stepping put of the way when anything higher than 44.1 comes through it. 44.1 sounds absolutely amazing. Of course there is a variance in quality amongst 44.1 recordings but for the most part 44.1 is imensely enjoyable.

I think if I had the puter in the same room (i would blow my brains out)

I assume you mean because of the noise the computer makes? Fair point, with standard computers. My computer is designed to be in the listening room, only playing music. It uses very little power, so generates very little heat, so passive cooling is good enough (with a purpose-designed case, of course). The PC has no motors, fans or HDD. Makes absolutely no noise.

And a linear power supply helps in not injecting electrical noise, despite being plugged into a different AC circuit just in case.

I am streaming TIDAL with the Beta APP from my Oppo, bistream via HDMI into an ESSENSE DAC ($499) which converts to 24/192. What is really happening? Can anyone, with a real expertise, describe the process including whether or not re-clocking is involved. I have a truly high end system and this sound is truly glorious. The "air" and the soundstage fill the entire room. I have never heard anything like it before at THE Show or CES. BillinOC

Lewinskih01 - I do mean noise and my puter is very quiet as well. Its also plugged into a power conditioner and APS and sits on a 4 inch maple shade base with the brass points and an isolation brick on top - and I would never let the nasty bugger in my sound room necause of all the emf and other nasties is gives off (not just noise you can hear but noise you cant as well). I have a bathroom right behind my sound room on the other side of a wall - I may drill throyg and put the server in that bathroom (I guess like Hillary) to get it closer and get away from the 10 meter cable but it does suound awesome.

Lewinskih01, There is no such thing as "Linear Power Supply". All power supplies are switchers and so called linear one is pretty bad one switching at 120Hz - frequency very difficult to filter out. Switching happens at the peak of the voltage while current is taken from the mains in short spikes of very high amplitude (polluting). For these reasons Rowland amplifiers use high frequency (1MHz) SMPS switching at zero voltage/ zero current even in his class AB amps. SMPS get their bad name from crude computers applications.

Steakster, Linear power supplies pollute more then well executed switching power supplies, that got bad reputation from crude computer applications. Well executed SMPS is extremely quiet and that's why Jeff Rowland uses them now in all amplifiers (also class AB), not to mention preamps (like Capri) where, because of low power, efficiency is not a factor. I'm not sure where "linear" came from, but probably refers to linear output vs. input voltage change - meaning unregulated. You can add linear regulator, but only for small loads like CD, preamp, etc - not for the power amp. They require huge capacitors to keep voltage ripple down. Transformers are huge as well. The reason you had better results with linear supply was the fact that your computer switcher wasn't designed for audio application.