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96kHz/24 bit v 192kHz/24 bit from Mac Mini?

IME, all files sound best played at their native resolution. Itunes on Mac OS is pretty useless if you want to start playing around with HD audio as it has digital EQ you can't bypass and the output bitrate will be fixed at that set in the Mac OS midi utility.

BitPerfect, Audirvana & Kodi (aka XBMC) can be configured to automatically spit out unmediated audio at native bit rates and are preferable to itunes for this reason.

If you must up or down sample this should always be done in multiples of the native bit rate (eg. 192 down to 96, 44.1 up to 88.2) otherwise it will sound wrong as the processor will be called on to do some dirty interpolation (fnar, fnar :)).
 
But... but... doesn't upsampling give you smaller staircase steps - so a less digital sound? :)
No. See above. It just fills that additional sample block with a clone of the previous sample block. It's like saying 2/10 is bigger than 1/5.
 
If you wish to upsample I suggest you choose a high performance music player programme like HQPlayer or Audirvana (which uses iZotope) ; Mac OS's Core Audio upsampling and filtering performance is mediocre:

http://src.infinitewave.ca

Even Foobar is nothing special...

R

I use Audirvana on my Mac Pro and dislike the sound that comes with up-sampling. Too smooth for my liking.
 
I use Audirvana on my Mac Pro and dislike the sound that comes with up-sampling. Too smooth for my liking.

What do you mean by "smooth"?
For me" smooth" is the opposite of "harsh" and is a good thing.

There are many filtering options that you can choose from when upsampling, didn't you like any of them?

Many DACs ASRC upsample to 176.4 and 192 KHz internally before feeding the D/A chip.
In my opinion and experience doing this in a computer produces better results but this depends on the DAC and on the music player software being used.

R
 
No. See above. It just fills that additional sample block with a clone of the previous sample block. It's like saying 2/10 is bigger than 1/5.

But 2 is twice as big as 1, and 10 is twice as big as 5, so 2/10 is *four* times as big as 1/5! (notice the smiley in my original posting) bigger
 
If you must up or down sample this should always be done in multiples of the native bit rate (eg. 192 down to 96, 44.1 up to 88.2) otherwise it will sound wrong as the processor will be called on to do some dirty interpolation (fnar, fnar :)).

And what's so dirty about interpolation (as long as it is digital and not physical)?
 
Just a few observations from someone who can, seemingly uniquely (and I don't believe that for a moment) tell the difference between 16/44 and 24/192..

I spent some time with a Meridan Explorer 2 DAC listening to the differences between the above quality levels. These were evident and obvious.

I have a PC with a sound card capable of playing 24/192 and have the setting set to 24/192.

The PC doesn't render 24/192 "properly". By which I mean that it has certain audible artefacts most notable at the top end. A certain "fizzy" quality even when playing 24/192. But then it's probably £20 of DAC tech at best.

I could hear a stack of differences between resolutions, but one of the most obvious was in the handling of HF detail and how that interacts with the rest of the frequency range.

At 192 I found the top end to be quite precise and flowing in a way that nothing below that resolution could manage, even 24/96 which I didn't think was that much better than 16/44. You can hear the 24 bits versus 16, but the 96 wasn't much of a step up - the 192 was.

I wonder if it's a "flaw" or whether the music is being flattered by low res (16/44, 24/96) playback. I wait for someone to express incredulity at describing 24/96 as low-res. Anyway..

The DAC chip in this PC is dreadful. OK, it's better than the one in my iPhone but it's not great. It's OK blasted out of the PC speakers (Creative Pro ones) but would be hilarious played through decent speakers.

My impression is that Apple kit tends towards brightness, unpleasantly so.

Playing anything through our Apple TV box sounds unpleasantly bright and over-enthusiastic and often unlistenable and that can only manage 16/48, I think. Apart from up-sampling from 44 to 48 it's doing something else. Not sure what but the result isn't nice.

Any flaws at all in the rendition coupled with that brightness is going to hurt the ears.
 
And what's so dirty about interpolation (as long as it is digital and not physical)?
Hey, It's a pretty arcane argument and I sincerely doubt many here could tell the difference, but I just like to keep unnecessary faff out of the digital signal path and anything other than direct multiplier up or down sampling falls into that category. As does avoidable up or down sampling.
 
No. See above. It just fills that additional sample block with a clone of the previous sample block. It's like saying 2/10 is bigger than 1/5.

That would be a very crude and basic way to 'upsample'. Particularly when the ratio of the sample rates isn't a simple N:1.

Upsampling can be done in many ways. Some better than others. So the real problem is how well it is being done, and the details of the process used.

Yes, some upsampling in software can work very well. The methods used by the 'sox' utility are an example. This can give good results. (And the same code is used elsewhere.) Similarly some computer mixers can do this well *if* the system is set up accordingly and can do it in a timely fashion.

However if you've bought a *good* DAC then it should itself carry out high-quality rate conversions when needed, and render the audio well *without* the computer host having to fiddle with the samples first.

The main advantage of having computer software do the re/up-sampling is that - if you know what you're doing - it gives you the ability to define, check, and choose how the process is done, and get the reconstruction filtering *you* prefer. I did do some demo programs on this a while ago using x2 upsampling for simplicity.

Anyone interested can have a look at

http://www.audiomisc.co.uk/software/ARMiniX/Upsampling.html

The programs with source code are provided. But note that these demos were written to run on the RISC OS operating system. So as they stand, they won't work on other OSs. A compentent programmer could port them if they wanted, but I assume similar ones are available. Indeed IIRC 'sox' lets the user define the resampling process as they choose. So anyone who wants to experiment with the effects of different ways to upsample might find sox a good place to start.

Jim
 
No. See above. It just fills that additional sample block with a clone of the previous sample block. It's like saying 2/10 is bigger than 1/5.

You are assuming that all upsampling/filtering software uses linear interpolation which as far as I know is not the case.

R
 
Simply repeating sample values isn't even linear interpolation. In effect it is the ancient (in engineering terms) "sample and hold" process which still can show up in basic 'NOS' designs. To remove HF products it requires a good following *analogue* filter.

Linear interpolation means that a new sample created in between original ones sits on a straight line between them. e.g. one mid-way between them has a value that is their average. In effect joins the dots with straight lines.

Pioneer used to call this 'Legato Link' IIRC. My old Pioneer CDR decks use it. Not perfect in engineering terms, but better than a poke in the eye with a sharp stick. :)

Jim
 
Simply repeating sample values isn't even linear interpolation. In effect it is the ancient (in engineering terms) "sample and hold" process which still can show up in basic 'NOS' designs. To remove HF products it requires a good following *analogue* filter.

Linear interpolation means that a new sample created in between original ones sits on a straight line between them. e.g. one mid-way between them has a value that is their average. In effect joins the dots with straight lines.

Pioneer used to call this 'Legato Link' IIRC. My old Pioneer CDR decks use it. Not perfect in engineering terms, but better than a poke in the eye with a sharp stick. :)

Jim

Thanks.
By the way, I have been enjoying your website for years.

Best,
Ricardo
 
Also
https://www.st-andrews.ac.uk/~www_pa/Scots_Guide/intro/electron.htm
although I ceased adding much to that when I 'retired' about ten years ago. :)

Plus, now, of course the ukhhsoc site
http://www.torrens.org.uk/ukhhsoc/
:)

FWIW I did in the past create some other sites. But handed those over when I retired.

Edit: Occurs to add for the sake of clarity. My 'Armstrong' website in practical terms preceeded the full AudioMisc one. Although it is now a section at

http://www.audiomisc.co.uk/Armstrong/armstrong.html

the Armstrong pages are largely self contained.


Jim
 
I have tried Audirvana and another audio output "improver" (Soundflower with Audio Essentials) and did not like the results. It made the music far too warm and rich sounding for my personal taste. The other point is that my more recent tracks are all studio grade downloads at 96/24, so other than the option of auto setting the output with Audirvana or similar, I should not get any negative effects from up sampling the 44kHz/16 bit CD files to 96/24. I do seem to get adverse effects by up sampling to 192/24 and as I only have a couple of downloads at this bit rate, there is little to be gained. What I would like to know is if others have experienced this problem, which I am fairly sure, originates in the Mac Mini not in the DAC-2L.


Audirvana is not an audio output improver.
 
Are you sure about this? Mac OS X has a Core Audio subsystem which provides several layers of hardware abstraction. Most of the specialist players like Amarra and Audirvana can be configured to bypass some if not all of these layers, but I am fairly certain that iTunes relies on Core Audio quite heavily.

There is an inexpensive app available on the Mac app store called BitPerfect, which can be configured to automatically switch the Mac's output to match the content's native sample rate. This effectively sidesteps the Core Audio resampling. It is discrete (a toolbar icon is all you see during normal operation) and it plays nicely with iTunes.

That is all correct.
 
If you wish to upsample I suggest you choose a high performance music player programme like HQPlayer or Audirvana (which uses iZotope) ; Mac OS's Core Audio upsampling and filtering performance is mediocre:

http://src.infinitewave.ca

Even Foobar is nothing special...

R


100% agreed. I use Audirvana and HQPlayer and both are superior using a MacMini with SD card OS X, external PSU and either an external drive or an internal SSD drive only for the music archive.

In fact, a MacMini 2014+software+FLAC uncompressed files from 44,1 to 192 blew my Naim HDX out of the room at a fraction of the cost.

Upsampling is a question of taste and CPU power. Sometimes I like the result and sometimes I don't. My DAC uses the XMOS USB interface, which is one of the best if not the best interfaces for USB.

Of course good software low level surpases all the OS X level drivers and iTunes is a joke in the audiophile arena, in fact most tools are focussed on usability and database management and only few put sound first and maybe 1 or 2 combine usability and sound.

Chris

A lot of information can be found on the Computer Audiophiles Forum, thousands of threads and posts on this topic
 
100% agreed. I use Audirvana and HQPlayer and both are superior using a MacMini with SD card OS X, external PSU and either an external drive or an internal SSD drive only for the music archive.

And what specific aspects of the sound are affected by those components?

A lot of information can be found on the Computer Audiophiles Forum, thousands of threads and posts on this topic

I think "misinformation" is a better description for most of the stuff on CA. :)

There are a couple of knowledgeable posters there, but they usually get shouted down.
 


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