QUOTE(Dawnrazor-age @ Oct 14 2006, 13:03) 
< Nelson Pass's commitment to single ended MOSFET designs is strange, but his amps end up sounding very good - not necessarily transparent, but nice.>
Pass doesn't only make SE designs, the one I have is the X series. What Double blind test are you refering to that concludes his amps are audibly not transparent?
My point was more than many of Pass's designs measure poorly (and, based on those measurements, one would expect a certain ABX result) but sound nice. I don't use that headphone amp any more, but could do some testing if anybody was interested.
QUOTE(Dawnrazor-age @ Oct 14 2006, 13:03)
I made some 192 vbr mp3s using lame and have done a comparison with the original .wavs. I scored 14/15 in the Foobar ABX test you recommended. That is with crappy onboard sound into some decent head phones. It was fairly close, but once you hear the differences, it is easy to figure it out with consistency. The bass was not as full and rich on the MP3s, and there was a bit more slam on the original.
I am assuming you want me to post the results and the files. Where should I do that, in the file upload section?
QUOTE(Dawnrazor-age @ Oct 14 2006, 13:03)
I was able to easily ABX files at different rates 44.1 vs. 96khz. maybe it was the distortion you are talking about, but they did sound different.
I'm still busy reading that article, but for most hardware upsampling a signal to a higher rate before the DAC (not counting oversampling DACs, which do it for different reasons) will tend to be worse, rather than better in quality. Maybe you could try something like ABC/HR to rank the results.
QUOTE(Dawnrazor-age @ Oct 14 2006, 13:03)
Also, I remember a gentleman who had a dac that lit up when HDCD files were played, and when the Kmixer was involved, the dac would not light up, but when using ASIO, it lit up. HIs conclusion was that the bits were not identical through the kmixer.
QUOTE(Dawnrazor-age @ Oct 14 2006, 13:03)
Thanks again for getting me to do the ABX test. It really solidified what I thought I knew...and that was on just average equipment! I can only image how much easier it would have been on my main rig where imaging could be part of the test.
From http://www.positive-feedback.com/Issue22/nugent.htm
QUOTE
Upsampling is one of the BIG advantages of using a computer to drive your music. Digital music is recorded at a particular sample-rate, generally 44.1 kilosamples per second on CD's. Upsampling or resampling adds more samples that were not part of the original recording process, but try to approximate the samples that would have been recorded had the music been recorded at the higher sample rate initially. The added samples are computed with various mathematical algorithms that examine the music waveform prior to and after the time-slots where the new samples will be inserted. These new samples not only add more detail to the music, but improve the dynamics as well.
This isn't an accurate representation of what is going on with resampling. I suspect Steve Nugent knows that, but is writing for a non-technical audience.Ok, in general, resampling adds no more information into a signal. The new samples that are inserted are inserted in positions predicted by interpolation algorithms which add nothing to the signal - the original sampling rate sets the bandwith and the original bit depth sets the SNR. Adding more samples to the signal with some algorithm that is not an oracle will add one of two things:
- Nothing, or
- Distortion
Of course when you take digital to analog conversion into account the picture is more complex than that, but in general resampling is just a waste of cycles (if your DAC supports the orginal source rate).