4 ms·

Probably correct. 24/88.2 is probably the perfect compromise between too much and too little. Then there is the issue of the Nyquist-Shannon sampling theorem l

by ipspam 5y ago

Probably correct. 24/88.2 is probably the perfect compromise between too much and too little. Then there is the issue of the Nyquist-Shannon sampling theorem living up to its FULL POTENTIAL. "It establishes a sufficient condition for a sample rate that permits a discrete sequence of samples to capture all the information from a continuous-time signal of finite bandwidth. Strictly speaking, the theorem only applies to a class of mathematical functions having a Fourier transform that is zero outside of a finite region of frequencies. Intuitively we expect that when one reduces a continuous function to a discrete sequence and interpolates back to a continuous function, the fidelity of the result depends on the density of the original samples. The sampling theorem introduces the concept of a sample rate that is sufficient for perfect fidelity for the class of functions that are band-limited to a given bandwidth, such that no actual information is lost in the sampling " (1) Did you get the significance of that? "Such that no actual information is lost in the sampling" If a proper filter is used, and infinite computing power is used, the analogue waveform can be PERFECTLY recreated. This simply doesn't happen, and designers don't even try. Hard to believe, right? The issue is that for full information retrieval, a filter of infinite tap length is needed. Think infinite ripples or ringing in both directions of the sample point. And that requires computing power getting close to infinite. Which brings me to a crazy dude exploiting this theory to its fullest, Rob Watts, designer of Chord Qutest and M-scaler. "What’s the significance of the ultra-long tap-length digital filter and of the WTA algorithms? In seminars given around the world, Watts has suggested that few designers recognize the full implications of digital audio sampling theory, which according to Watts points to an astonishing conclusion. Specifically, Watts contends that garden-variety 44.1kHz digital audio files could, if processed through a digital filter of near infinite tap length, yield analog waveforms every bit as accurate and complete as those produced from higher-res audio files, albeit with slightly higher noise floors. Stop a moment and let that claim sink in. Watts is saying, in essence, that a DAC with a properly designed filter system can deliver ultra-high-res sonic results from conventional CD-quality files. " (2) All design choices are a trade off. Size Vs processing power to "unpack" and recreate data points between samples. Which brings me to my main point, it may be unnecessary to have a sample rate higher than 44100x per second, but it is practical to do so. By doubling the sample rate and storage requirements, it takes far less computational power to play the song as it would to mathematically calculate that extra sample between each sample. (1) https://en.m.wikipedia.org/wiki/Nyquist-Shannon_sampling_theorem https://en.m.wikipedia.org/wiki/Nyquist-Shannon_sampling_the... (2) https://www.theabsolutesound.com/articles/chord-electronics-hugo-tt-2-dacpreampheadphone-amp-and-hugo-m-scaler https://www.theabsolutesound.com/articles/chord-electronics-...

BizarroLand 5y ago
That's interesting and it makes sense. Early and mid-stage audio courses focuses so much on the numerical differences between levels of digital integrity, 44.1 vs 48 vs 88.2 vs 96, etc., and they mention that if you encode an audio sample and compare it with a waveform analyzer that you can see the differences between the audio waveforms. Which is true, but the differences are incredibly minor and mostly in the inaudible range, and those differences are really nothing once you consider how the audio is going to be altered and mangled to some degree by the DAC, amplification, and speakers they are played back through, the rooms they are played in, and by the listener themselves. Fidelity is always lost, so the smart thing would be, unless it is business critical with real money on the line, to find a balance. 96k is half the size of 192k but twice DVD audio rate. I would say that's a good place to be unless your paycheck depends on something better, and if they aren't played side-by-side on a tens of thousand dollars audio system within an expertly acoustically treated room you would never be able to detect the difference.

by ipspam 5y ago

BizarroLand 5y ago

That's interesting and it makes sense. Early and mid-stage audio courses focuses so much on the numerical differences between levels of digital integrity, 44.1 vs 48 vs 88.2 vs 96, etc., and they mention that if you encode an audio sample and compare it with a waveform analyzer that you can see the differences between the audio waveforms. Which is true, but the differences are incredibly minor and mostly in the inaudible range, and those differences are really nothing once you consider how the audio is going to be altered and mangled to some degree by the DAC, amplification, and speakers they are played back through, the rooms they are played in, and by the listener themselves. Fidelity is always lost, so the smart thing would be, unless it is business critical with real money on the line, to find a balance. 96k is half the size of 192k but twice DVD audio rate. I would say that's a good place to be unless your paycheck depends on something better, and if they aren't played side-by-side on a tens of thousand dollars audio system within an expertly acoustically treated room you would never be able to detect the difference.