Convert CD-rips to 24/88.2 better SQ?

Posted by: okli on 12 October 2011

what's your opinion / experience - does this bring sound quality benefits? I've started to experiment with my CD rips and I think I like the "hires" versions better...

Posted on: 12 October 2011 by Rob van den Brand

Converting 16 bit 44 khz material to 24 bit 88.2 khz won't make a song sound better, just as converting a low bitrate mp3 to 16/44 won't. It won't add any information to a track because it just isn't there. If you're comparing the same track at different bit depths however, for instance a rbcd (16/44) with a sacd (at 24/88.2), the sacd could sound better (depending on the mastering) because there's more information to be heard. 

Posted on: 13 October 2011 by okli

Rob, did you make any tests - I think that upscaling the bit depth leads to higher dynamic range and the upscaling algos are smart enough to use it - I didn't check the foobar's documentation what are they doing in this case but I don't think they simply insert 1 byte of zeroes and I can clear hear that the converted files sound better to my ears with my equipment. OTOH, why all the CD players are using internal 24/192 upscaling DACs, if this won't cause any SQ improvements. Even Naim states for UQ that they pass the iRadio signal (which is at max 320 kbps) through their internal DAC and I must say the sound is really great for this material. As for upscaling / oversampling 320 kbps mp3 to 16/44.1 I think this is another conversion, because the input has been already compressed (lossy vs. lossless). Any further opinions / experiences / tests are highly appreciated.

Posted on: 13 October 2011 by Andy McGhee
Up sampling can make a difference in some hardware depending on how it's done. If extra samples are inserted using interpolation rather than just padding then it allows the convertor and hence the anti-aliasing filter to operate further away from the upper limit of human hearing, or use simpler filters, and this can improve quality in some systems. Of course this isn't adding sonic information that isn't there (well it is, but maybe not what was intended) but it can help the DAC reproduce it more faithfully.
Posted on: 13 October 2011 by Mr Underhill

 

I did a post about CMP2 here:

 

https://forums.naimaudio.com/di...077#1566878605731077

 

In using this I set cPlay to do upsampling. My comments:

 

Using 'Diana Krall - A Night in Paris' and 'Led Zepplin - IV' as examples:
I am NOT a fan of Diana Krall, but I do enjoy this album as she sticks to the great American songbook, and does it with aplomb.
I had selected an output of 192KHz, and the rip was of my CD, and so at 44.1KHz 16bit.
I select the VHQ upsampling.
This will then have been handled within the Benchmark DAC1 I use.

I really was VERY taken with the analogue sound that was produced - excellent, at least in some ways.

I then played the Led Zep, and could here what the upsampling was doing, but it was removing BITE. It all sounded a bit, well, BORING.

I moved back to Foobar, which was very good, in the ways described above.

I then moved back to cPlay, but set the output to 44.1 - and so avoided upsampling, and matched Foobar2000.

Upsampling definitely is interesting, and I can hear why it could be excellent when applied to the right music in the right way - but I am wary that the effect may need to be managed on a case-by-case basis.

 

Basically I turned it off, on the whole I found it more detrimental than helpful.

 

M

Posted on: 14 October 2011 by okli

M, very interesting and informative - I have to make some more tests, when the time permits. By coincidence I wanted next to convert some heavier stuff and Led Zeppelin was my first thought - so I'll try to convert some of my favorite tracks on the weekend and let's see (actually hear) what will happen. OTOH I find the 24/96 Metallica from hdtracks really better than my CD, but this is another story... (hopefully nothing to do with upscaling and upsampling of 16/44.1 source :-)...) 

 

 

 

Posted on: 14 October 2011 by Andy McGhee

Sorry, in my previous post I stated anti-aliasing filters rather than low pass filters. I'm sure some of you had noticed. The principal still stands but hopefully I have prevented myself looking too foolish.

 

Andy

Posted on: 14 October 2011 by Mr Underhill

okli,

 

I think the great thing about this is that the software is available for us all to do tests, listen and form an opinion that works for us.

 

If you have a spare laptop using the CMP principles I think it is incredible what you can achieve for zero money - and then play with things like upsampling within a software client, i.e. Not having to create a new set of files.

 

....mind you, what if you then found that the files sound better as no 'on-the-fly' transformation is being done?

 

...or, NO - I've made my bed for the moment, and happily listening to music as I type.

 

M

Posted on: 14 October 2011 by Simon-in-Suffolk
Okli, further to Andy's point, oversampling or sampling reduces the recorded jitter noise that was created when digitally mixed or converted ( by averaging it out over the new frequency spectrum) Therefore it can make the converted analogue audio sound more natural.
Most high end DACs over sample  to achieve this and benefits in the filtering DSP and analogue filtering. However if your DACs DSP is not very high quality,you may well get a better effect oversampling at the WAV encoding end.
Simon
Posted on: 17 October 2011 by okli

Hi Simon - I've compared both versions on the same system - first from USB stick directly on my nDAC and after that UPnP to UQ, which is connected digitally to the nDAC, so it should be the source file causing the difference.

 

Anyway, as M very clever points out - we can play with these things, but at the moment I'll leave everything as it is and enjoy the music...