Hi res music retailers - who do you use?

Schpyder said:

Tell me you don't understand frequency analysis, time domain waveform analysis, or digital signal processing without telling me.

To make it simple for you: superposition of two sine waves at the same frequency is a sine wave at that same frequency, regardless of phase shift or amplitude difference between the two original waveforms. This is easily mathematically provable via the compound angle formula. But this betrays your misunderstanding. If you're sampling at the same frequency as the waveforms, neither will be accurately reproduced. You're getting one sample per cycle, at the same point on the waveform every time. You are, in effect, recording a flat line.

Now, I assume you meant to say that you are sampling at twice the frequency of source A, so that you would be perfectly sampling the peaks and valleys of said signal. Also, I assume you are aware that modern statements of the Nyquist limit state it as a strict inequality: that is that you must be sampling at greater than 2x the maximum frequency you wish to reproduce, to avoid aliasing errors at that frequency. What am I saying, of course you don't, because that's exactly the situation you've posed here. It's also why we low-pass filter at a frequency a good bit below the Nyquist cutoff: to avoid the reconstruction errors that occur as you get close to the critical frequency.

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Well, you don't want to answer the question either. Once again a bunch of trivial information. Thus I will not be able enlighten you step by step. It's sad to see people that will never now what Stevie Ray Vaughan sounds like at 192,000 vs 41,000 samples (using proper equipment), because they don't even entertain others attempts to explain. You seem set in your ways and I get it, you don't want to be wrong, you have spent a lifetime believing what you believe. To late to turn back now?

figatry said:

Well, you don't want to answer the question either. Once again a bunch of trivial information. Thus I will not be able enlighten you step by step. It's sad to see people that will never now what Stevie Ray Vaughan sounds like at 192,000 vs 41,000 samples (using proper equipment), because they don't even entertain others attempts to explain. You seem set in your ways and I get it, you don't want to be wrong, you have spent a lifetime believing what you believe. To late to turn back now?

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Well. Why don't you enlighten us, since you know the answer and stop throwing out personal attacks and dismissiveness. We're listeneng.

figatry said:

Well, you don't want to answer the question either. Once again a bunch of trivial information. Thus I will not be able enlighten you step by step. It's sad to see people that will never now what Stevie Ray Vaughan sounds like at 192,000 vs 41,000 samples (using proper equipment), because they don't even entertain others attempts to explain. You seem set in your ways and I get it, you don't want to be wrong, you have spent a lifetime believing what you believe. To late to turn back now?

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You won't be able to enlighten me step by step because you are simply incapable, being wrong. Claims to the contrary are simply bloviation until you follow through.

I suspected this was some sort of stealth gotcha about "beats" or something.

Wheels Of Confusion said:

I suspected this was some sort of stealth gotcha about "beats" or something.

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Well, he's discussing two waveforms at the same frequency, so beats aren't even a concern. You just get constructive or destructive interference depending on phase. He was clearly trying to post some sort of gotcha about reconstruction error at the critical Nyquist frequency, but no one gives a shit about that, because it's a.) a completely well-understood phenomenon, and b.) everything in recording and mastering is designed around avoiding it (hence 20kHz low-pass filtering on a 44.1kHz sampled signal). In fact, it's worth noting that 44.1 kHz was only ever intended to reproduce 20-20,000 Hz, with the extra frequency overhead to account for precisely this issue (and to make the transition band of real rather than theoretical low-pass filters easier and cheaper to implement). The exact 44.1 kHz number came from PCM adaptors that stored audio on video cassettes. Yes, it's weird, but it's a thing that happened in the early days of digital audio.

I do enjoy the insinuation that some extra information will magically appear from old SRV masters that were frequency-limited by the physics of reel-to-reel analog tape to 20-ish kHz. The best numbers I've seen from R2R at high speed (15 or 30 IPS) is significant (>6dB) dropoff in frequency response at about 23kHz. Or if it was mastered post-digital changeover, it was probably done on DAT, or possibly the afore-mentioned 44.1kHz videocassette, And guess what sampling rates DAT supported: 32, 44.1, and 48 kHz*. Whoops.

* (of note, there was in fact a 96 kHz/24 bit DAT standard, and an earlier 96 kHz/16 bit proprietary format by Pioneer, but both were released 6+ years after Vaughan's death.)