Upsampling 16/44.1 collection a good idea?

[kemmler3D]

Wasting 30 seconds is still a bit different than wasting dozens of hours...

If I can accomplish the same thing by just resampling on the fly with foobar2000, that'd be great! I wonder if there's a plugin which allows you to switch to upsampling to a multiple of 44.1kHz when the source is 44.1 (or 88.2 etc), and a multiple of 48kHz if it's 48k (or above, up to but not including 768kHz).

I think in general resampling plugins or features tend to support a lot of different resolutions, I would definitely try that first.

 

[Hayabusa]

Wasting 30 seconds is still a bit different than wasting dozens of hours...

If I can accomplish the same thing by just resampling on the fly with foobar2000, that'd be great! I wonder if there's a plugin which allows you to switch to upsampling to a multiple of 44.1kHz when the source is 44.1 (or 88.2 etc), and a multiple of 48kHz if it's 48k (or above, up to but not including 768kHz).

I think the foobar filters are good enough:https://src.infinitewave.ca/
There are several filters that are artifact free down to -160dB.
for instance "foodsp best sync"

 

[mike7877]

Would you do it with your digital photos?

I'd rescan them and save them in a better format.

edit: Rescan = open and resample

 

[mike7877]

That depends on the filter used by the program that did the conversion.
I'm not sure what you mean exactly, but if it is applying low-pass filter on hi-res files before sending them to DAC then I think @dualazmak is doing something like that. IIRC he uses low-pass filter at 25 kHz.

I'm gonna check out his stuff. He talks about it on the forum here?

 

[Keith_W]

For some reason a lot of people here are big on justification "why you wanna do that?"

You are right. I hope that my post earlier in this thread did not come across that way.

I always applaud people who are curious about trying new things. But you did ask whether it's a good idea to upsample your files ... I provided an answer

I suggest that you try it. I don't use Foobar, so I don't know if it upsamples files. Both HQPlayer and JRiver are available as trial versions. JRiver's trial expires in 14 days, HQPlayer's trial can be used indefinitely but it stops playback at 30 mins requiring a restart of HQP. HQP also has more resampling options than any other player, including options to resample to DSD. For these reasons, I recommend HQP as a more suitable tool for experimentation.

The reduction in SINAD and jitter is DAC dependent - some DAC's show more of a benefit, some show less - so I also recommend you purchase an E1DA and analyse single tones, multitones, and perform a J-test with REW. Analyse the frequency response and expand the graph to include ultrasonics because that is where the differences usually are. Try it with different sampling rates. You will find that the differences between sampling rates / choice of PCM vs. DSD will be in the ultrasonics, and extremely low in level. That is, unless you happen to own a really poor DAC. And of course, listen for yourself for any audible benefit (if any). The only way to convince yourself if ASR has been giving you the right advice or not is to try it and see.

I'm gonna check out his stuff. He talks about it on the forum here?

He talks about it a lot. Just tag him like this @dualazmak and I am sure he will appear in this thread very shortly.

 

[mike7877]

The red curve has artifacts. Its filter only attenuates by +/20dB at 24KHz . This means a 20KHz signal also mirrors at 24Khz at the level.

I understand that part - I'm going to look into the best ways to resample, resample that way, and then play back at the highest sample rate possible. There's a lot of processing power in PCs these days which I'm sure can do a much better job than a tiny part of a $10 chip. Yes, there are questions about whether the result will be audible, and I'll have my answer then lol

 

[MaxwellsEq]

https://src.infinitewave.ca/ is an excellent place to dig into sample rate conversion.

My research convinced me that with extremely well implemented DeltaSigma DACs, any sample rate conversion would be not be necessary. For owners of poor quality or NOS DACs there will probably be some benefit. Furthermore, sample rate conversion can create significant artefacts if a poor solution is selected. My curious mind makes me wonder what percentage of people up-sampling know enough about SRC to make the right decision on conversion tools.

 

[mike7877]

You are right. I hope that my post earlier in this thread did not come across that way.

I always applaud people who are curious about trying new things. But you did ask whether it's a good idea to upsample your files ... I provided an answer

I suggest that you try it. I don't use Foobar, so I don't know if it upsamples files. Both HQPlayer and JRiver are available as trial versions. JRiver's trial expires in 14 days, HQPlayer's trial can be used indefinitely but it stops playback at 30 mins requiring a restart of HQP. HQP also has more resampling options than any other player, including options to resample to DSD. For these reasons, I recommend HQP as a more suitable tool for experimentation.

The reduction in SINAD and jitter is DAC dependent - some DAC's show more of a benefit, some show less - so I also recommend you purchase an E1DA and analyse single tones, multitones, and perform a J-test with REW. Analyse the frequency response and expand the graph to include ultrasonics because that is where the differences usually are. Try it with different sampling rates. You will find that the differences between sampling rates / choice of PCM vs. DSD will be in the ultrasonics, and extremely low in level. That is, unless you happen to own a really poor DAC. And of course, listen for yourself for any audible benefit (if any). The only way to convince yourself if ASR has been giving you the right advice or not is to try it and see.

It wasn't so much directed at you lol, you just said a couple words and then were there lol.

I get that their questioning comes from a good place - they don't want others to waste their time.

My favourite DAC right now is E70 Velvet - I've heard it's better DSD than PCM, so I can try resampling PCM to DSD as well

 

[mike7877]

https://src.infinitewave.ca/ is an excellent place to dig into sample rate conversion.

My research convinced me that with extremely well implemented DeltaSigma DACs, any sample rate conversion would be not be necessary. For owners of poor quality or NOS DACs there will probably be some benefit. Furthermore, sample rate conversion can create significant artefacts if a poor solution is selected. My curious mind makes me wonder what percentage of people up-sampling know enough about SRC to make the right decision on conversion tools.

I did a bit of reading a few years ago, and there's one "component" which supposedly [allegedly] does a good job. Sounded superior to me when I compared it to the built-in resampler in foobar2000 (shh!!)

I forget what it's called now... I'll have to look. I had it installed again 2 years ago, so I think I know where I can find it.

I'm going to check out that site - maybe when I find that resampler you can give me your opinion on it?

edit: interesting... I think I know which one I used, and it looks perfect there! Have to find it again and make sure

 

[JSmith]

For some reason a lot of people here are big on justification "why you wanna do that?"

That tends to follow when one asks "is it a good idea"... just sayin'.


JSmith

 

[MaxwellsEq]

My favourite DAC right now is E70 Velvet - I've heard it's better DSD than PCM, so I can try resampling PCM to DSD as well

This is the thing I didn't get an answer to. There are some threads on ASR discussion the internal building blocks of ESS and AKM DACs and arguing that the DSD route through the chipset measures better than the PCM route. The problem I have is a) is this audible? b) do these benefits actually happen if the source file is native PCM (with the fs/2) brickwall filter) do any of these better routes actually make a difference? I'm not querying pure native DSD which has not seen a brickwall filter.

 

[melomane13]

That is always my justification, and it should be every hi-fi enthusiast's. I don't like getting into arguments with people saying "you can't hear that!" A lot of times, I can, in fact, hear differences. Quite easily, actually. But nobody believes what I say and demand double-blind testing, which, if I did, the results would then be questioned...

the sun revolves around the earth or does the earth revolve around the sun?
by making a parallel, you affirm that the sun revolves around the earth. we tell you that your feeling, having the appearance of truth, is illusory and give you the means to experience it for yourself.

you just have to have the courage to verify it, scientifically and in due form.

the difficulty is to question one's beliefs

try an ABX or stay in Plato's cave

 

[boxerfan88]

If I can accomplish the same thing by just resampling on the fly with foobar2000, that'd be great! I wonder if there's a plugin which allows you to switch to upsampling to a multiple of 44.1kHz when the source is 44.1 (or 88.2 etc), and a multiple of 48kHz if it's 48k (or above, up to but not including 768kHz).

Check out SoX, SoX mod components. One of them would fit your needs.

You’ll probably need to setup the SoX/SoX-mod component twice in your DSP chain, one to handle 44k1 and the other to handle 48k.

 

[Vincent Kars]

Because in the PC you can chooses what ever algorithm you want and you have infinity more computation memory and time available.

I fully agree, a PC can be configured to do a better job.
However, can you bypass the re-sampler of the DAC?
Can you defeat the filter of the DAC?
If not, where's the beef?

 

[pos]

Offline upsampling should in theory be better than realtime, but seeing that even that won't give any audible upgrades why even bother with any kind of upsampling?
Personally I just set my Windows machine to 96khz since that's what my MiniDSP is working at, but I doubt it matters anyway.

Your minidsp will resample to “his” 96kHz anyway, this cannot be avoided.

 

[Basic Channel]

I want to keep every step in the chain working as perfectly as possible, so that at the end of the chain, all these "barely perceptible" and "questionable if perceptible" effects are minimized to the maximum degree, so that when summed their total is as close to imperceptible as possible. That is always my justification, and it should be every hi-fi enthusiast's. I don't like getting into arguments with people saying "you can't hear that!" A lot of times, I can, in fact, hear differences. Quite easily, actually. But nobody believes what I say and demand double-blind testing, which, if I did, the results would then be questioned...

You do realise that we can all hear differences, that’s a fact. I can hear differences between two identical files that would sum to nothing in a null test. I can hear an EQ making a difference when it isn’t on.

Obviously the difference is that some of us are indefatigably infallible.

 

[PuX]

I choose 48khz because it offers some more headroom compared to 44.1 and movies/games/web are often in 48khz. I use SoX resampler in foobar2000 to upsample during playback.

most music is in 44.1 so by resampling to 48 you are only losing information, not gaining anything.
it's a bit like upscaling photos by 5%, they won't be any sharper, and it's not an even mathematical operation, involves float numbers.

 

[melomane13]

most music is in 44.1 so by resampling to 48 you are only losing information, not gaining anything.

no gain, no losing. i make the same only because i use convolver , and this require fixed samplerate.

 

[dualazmak]

I'm not sure what you mean exactly, but if it is applying low-pass filter on hi-res files before sending them to DAC then I think @dualazmak is doing something like that. IIRC he uses low-pass filter at 25 kHz.

Am I called? If yes, to begin with, let me share my post here;
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532

 

[LGD_]

I've been thinking of upsampling my 44.1kHz collection to 88.2, 176.4kHz, 352.8, or even 705.6kHz for playback...

I use foobar2000 as my player, with the SACD component, and it's DSD Processor,
I upsample my flacs to DSD 256 - while playing, ("on the fly").
This results in a very audible improvement in SQ , through my DAC (SMSL D6), compared to regular PCM playback with it.

This older post of mine explains it in greater detail:

SMSL D-6 DAC Review

D-6 'PrD' is DSD pass through (as per D-6 manual) and this activates DSDD mode of the DAC chip, shown below from AK4493S block diagram. DSD Direct mode passes the DAC chip DSP (volume control and modulator) and goes straight to switched capacitor filters and through to analogue output. I...

www.audiosciencereview.com

Give it a try, works great - and it's free (unlike HQPlayer).

Edit: The op asked ir there was a foobar plugin that could switch between all rates, both 44.1 and 48khz based:
The answer is YES:
With the DSD Processor,
I am upconverting everything from 44.1 to 192Khz (both 44,1 and 48Khz based) to DSD 256 seamlessly - no switching needed, on the fly. The original flac files are unchanged.