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[duesy]

How effective are adjustments to DSP Studio Output Format settings to sound quality (Source Bitdepth and Sample Rate)? My sound card supports up to 192 K HZ sample rate, but MC will stall and buffer at this setting. Thanks,

[Matt]

MC uses one of the highest quality resamplers available.  That high quality takes some CPU.

However, it's not necessarily better to overdrive a good DAC.  Let your ears be the judge.

[duesy]

I'm not sure your response addresses the question. Whatever the variables, I'd appreciate knowing the impact the forementioned MC settings have on sonic quality.

[GHammer]

What is the source material providing, 16 bit, 20, 24, 44.1K, 48, 96?
Hard to make a silk purse and all that...

[duesy]

High quality .mp3 files extracted with EAC, low compression ratio with Lame. Typically yields 250-320 Kbps VBR. RadioParadise.com now offers a 192 Kbps web stream (excellent, content by the way). All I'm trying to determine is whether upping MC's settings will yield better sound, and therefore justify potential upgrades to my hardware. I've hit the ceiling at around 92Hz Sample Rate, which may be due to network lag (wireless streaming .mp3 from local server), possibly requiring modification to a hardwired LAN connection. Plus my new sound card (M-Auido Audiophile 22) supports 64 bit rate and 192 KHz sampling so I'd like to realize its full potential. Thanks for any comments and suggestions.

[Myron]

High quality .mp3 files extracted with EAC, low compression ratio with Lame. Typically yields 250-320 Kbps VBR. RadioParadise.com now offers a 192 Kbps web stream (excellent, content by the way). All I'm trying to determine is whether upping MC's settings will yield better sound, and therefore justify potential upgrades to my hardware. I've hit the ceiling at around 92Hz Sample Rate, which may be due to network lag (wireless streaming .mp3 from local server), possibly requiring modification to a hardwired LAN connection. Plus my new sound card (M-Auido Audiophile 22) supports 64 bit rate and 192 KHz sampling so I'd like to realize its full potential. Thanks for any comments and suggestions.

If you're using lossy compression then I'd forget all about changing word width and sample rate.  You've already thrown away a bunch of information with the lossless compression and NOTHING will ever get it back.

If you're concerned with sound quality you should be using a lossless format.  Even then, NO sample rate converter can create information that's not in the original, so theoretically, sample rate conversion alone woll not improve anything.  There are some claimed benefits of lower reconstruction filter artifacts in using higher sample rates but sample rate converters will typically start with a lowpass filter tied to the original sample rate, so I'm not sure I buy this argument.

If you look at D/A specs, their performance typically degrades with higher sample rates, so this is working against you.

The one reason I do see to increasing word width is that it will minimize degradation cused by DSP algorithms, especially volume control.

[GHammer]

MC does a very nice job and is 32 bit through the internal signal path.
Matt pointed out that MC will handle rates to the limit of your sound card, but at a cost of CPU usage.

As for resampling, I'd likely not bother unless I had external equipment that requires a certain rate.
I haven't seen any reliable info on benefits of resampling.

If you are sure you do, I'd let my evil twin Skippy change settings while you're out of the room. See if you can tell what rate is being used. No elegant way to ABX it that I'm aware of.

[jgreen]

As Myron pointed out, you've already chopped the toes off your sound, when you ripped your CDs to lossy.  Reripping to lossless is the single best upgrade you can make to your digital path.

As Ham's last post implies, expensive stuff sounds better, as do convoluted processes.  The key is knowing which is which.  The A/B blindfold test really isn't fair, unless you're told ahead of time which one is supposed to sound better.

In terms of getting the most out of your hardware, I would suggest pointing your speakers at your ears.  The sonic difference between optimal and poor speaker placement trumps all the other stuff combined. 

[duesy]

I regret the decision to compress. I started "way back" in 2000 building my music library when compression made sense and disk space was still pricey. Now, I would not hesitate to use lossless though the constraining factor would seem to be lack of sufficient iPod capacity and performance. Various rooms at our home enjoy local control (via used laptops equipped with sound cards attached to audio systems) from a central server containing .mp3 clips at about 5:1 compression. I decided early on to standardize on one jukebox player (MC), one type of file structure, one distribution network. iPods can be sync'd either from the server or client. Anyway, I'm off the topic. I think my question has been well addressed and I appreciate everyone's time in responding. The original source material used to create my compressed music library largely scattered to the wind, not to mention the considerable time spent cleaning the file names for consistency and syntax. Now, if someone could figure a way to "rebuild" compressed .mp3's back to their original WAV state, the world would be a beautiful place!