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DVD-Audio Support in MC 11?

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jlange:
There are some external DACs that do both upsampling and interpolation (http://www.av123.com/products_product.php?section=processors&product=1.1).

I have never listened to one of these devices, but it does make sense to me that, if implemented properly, it would enhance the signal.  When you consider a 20kHz sine wave sampled at 44.1 kHz, the digital information is only about two samples per cycle.  When reconstructed, it starts out as a square wave and is then filtered during D-to-A conversion to round the edges into something that "resembles" the original waveform.  If the sample timing doesn't correspond to the maximum peak and valley of the original sine wave, the reconstructed wave won't look much like the original at all.  Lower frequencies fare much better since 44.1kHz will sample 44 times during one 1000Khz cycle.  Our ears are most sensitive to distortion in high-frequencies (audiophiles often describe CDs as "harsh"). Also, high-frequency harmonics contribute to our sense of "air", "transparency", and "three-dimensionality" of music, which is why people describe DVD-A and SACD in these terms.

So to get a "better" reconstructed waveform, a DSP must perform some kind of curve fitting on the digital sample points and then plot new points on that curve when upsampling to 96 or 192kHz.  Drawing straight lines between the points (linear interpolation) clearly wouldn't be enough.

Alex B:
Our ears are not sensitive to anything at 20 kHz. When listening to a sine wave very few people can hear it without significantly increasing volume. Many people don't hear it any volume.

When listening to standard music a lowpass of 16-18 kHz is usually good enough for everybody. Even with excellent hearing it is difficult to hear any difference between a straight CD audio sample and a 16 kHz lowpassed sample in a blind test.

The 2-channel stereo DVD-A and SACD recordings are often mastered differently without the current loudness and compression race that has happened to standard CDs. That makes most of the perceived difference.

Alex B:
By the way, MC's audio resampling uses high-quality Shibatch SSRC filters. If you like to try what it does to 44.1 kHz source material just set the output to 96 or 192 kHz. My Terratec DMX 6fire 24/96 soundcard can handle up to 96 kHz, but I can't hear the difference. I can only experience it in a form of increased CPU usage.

I use resampling from 44.1 to 48 kHz only on my laptop because it's internal soundcard resamples everything to 48 kHz with mediocre quality. I rather use MC's HQ software.

jlange:
I know that most people can't hear 20kHz (I think my own hearing rolls off at 15kHz), I was just using that as an extreme example of how sampling rate impacts waveform reconstruction.  A 10kHz wave is only sampled 4.4 times at 44kHz, which seems (to me) just barely adequate to rebuild the waveform.  Of course real music is layer upon layer of many different sounds, harmonics, transients, and so on that make for a very complex waveform.

My main point is simply that good algorithms CAN improve the sound when upsampling CD audio.

Barrie:
Why does Upsampling improve sound quality?

From the explanation above, there is apparently no extra information in the upsampled signal that was not present in the initial signal. With a 44.1 kS/s input, both the input data stream and the upsampled data stream will only contain a spectrum that must be between 0 and 22.05 kHz and is probably only between 0 and 20 kHz.

This conventional analysis starts from the viewpoint that the behavior of the ear can be described in mathematical terms using Fourier analysis. This assumption is probably pretty good – it means we are interested in frequency responses, for example, and these do provide good guides to the performance of equipment and to descriptions of what we hear. The analysis was right at the heart of the definition of the audio coding used on CDs.

For those working with audio, it is also apparent that theories based on these descriptions are not completely adequate, and that there can be significant differences in the performances of pieces of equipment with similar "conventional" specifications. It seems that two things are going on here – the ear may have more than one mechanism at work, and sine waves may not be the best function to use as the basis for analysis. On the mechanism front, it seems highly likely that the ear has a sound localization mechanism ("where is it") that is fast, and independent of the mechanism that says "it’s a violin", and that is related to transient response. There may also be a third mechanism at work. On the analysis front, it may be that some form of wavelet is the best basis for mathematical modeling. The problem here is that sine wave theory is relatively simple, and has been fully worked out by generations of mathematicians, following on from Fourier. Wavelet math is just plain hard work, and does not yet have anything like such a solid core of mathematical results to call upon. Our ears, however, are not waiting.

If one gets the frequency response of some equipment right, but the provision of transient information wrong, one or more of the ear’s mechanisms cannot work properly, and so we are unable to separate out echoes and cues about where a sound is coming from the rest of the "what is it anyway" signal. dCS’ upsampling filters are designed to help sort this problem out. They are best analyzed not in sine wave terms, but using wavelets.

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