However, most mastering these days (see "loudness wars") seems to aim for reduced dynamic range such that even the quietest portions of the music are only slightly quieter than the loudest. Anybody who actually set their amplifier as suggested here, playing back any commercially produced CD would destroy their equipment and/or hearing within seconds.

I think here you have logical flaw. Most modern CDs have reduced dynamic range due to compression ( we have post about it here, use search). So it won't be any problem for such setting.

Statement in the quote is mostly correct as whole loudness wars started (in mass) as answer to high background noise levels.

You have to realise that when playing back a CD, the amplifier is usually set so that the quietest sounds on the CD can just be heard above the noise floor of the listening environment (sitting room or cans). So if the average noise floor for a sitting room is say 50dB (or 30dB for cans) then the dynamic range of the CD starts at this point and is capable of 96dB (at least) above the room noise floor. If the full dynamic range of a CD was actually used (on top of the noise floor), the home listener (if they had the equipment) would almost certainly cause themselves severe pain and permanent hearing damage. If this is the case with CD, what about 24bit Hi-Rez. If we were to use the full dynamic range of 24bit and a listener had the equipment to reproduce it all, there is a fair chance, depending on age and general health, that the listener would die instantly. The most fit would probably just go into coma for a few weeks and wake up totally deaf. I'm not joking or exaggerating here, think about it, 144dB + say 50dB for the room's noise floor. But 180dB is the figure often quoted for sound pressure levels powerful enough to kill and some people have been killed by 160dB.

:-)

Yes, the article is excellent

Yes, I agree.

"The main reason to *store (record) in higher rates is to slow down audio in post, or to lessen the amount of 'work' required of the filter to go between 48 and 44.1. Otherwise, the higher frequency data can literally cause problems.."

A good note in the anti-aliasing article: "The purpose of oversampling is to relax the requirements on the anti-aliasing filter, or to further reduce the aliasing. Since the initial anti-aliasing filter is analog, oversampling allows for the filter to be cheaper because the requirements are not as stringent, and also allows the anti-aliasing filter to have a smoother frequency response, and thus a less complex phase response."

Though for film dialog, the recording standard is 24bit/48Khz, as slowdown and export to the Redbook standard is not expected

To fully capture a wave, two points must be recorded; the crest (peak) and the trough (lowpoint): http://9-4fordham.wikispaces.com/file/view/wave_crest.gif/243891203/wave_crest.gif

Thus, the sampling frequency, or how often a recording 'checks in' to take a sample, must be twice the highest frequency of the wave which is desired to be captured. [on another loosely related note, the bit depth describes the resolution of the height dimension of that sample, or amplitude]

But what happens if the rate of playback is fractionally different than the rate that the recording was sampled at? That is, not a clean division like a half (.5), but something like .918? The audio will alias. And thus, a filter is required to get clean results - and like a bad olpf, if the filter is not good, the higher frequencies (like high optical detail) will suffer http://en.wikipedia.org/wiki/Anti-aliasing_filter

http://en.wikipedia.org/wiki/Sample_rate_conversion

"Huh? Most speakers can't play "all frequencies". And moist ears can't hear them either."

The DACs you referred to (not the speakers). All ears cannot hear above 22 - except for audible distortion caused by higher harmonics interacting

"What this means? As it sounds like mumbo jumbo."

When the playback rate is out of sync, the wave peak points are out of sync, as the playback sampling regularly falls between the peaks at the highest frequencies.

A wave: http://us.123rf.com/400wm/400/400/rustyphil/rustyphil0807/rustyphil080700012/3259783-large-image-of-an-electronic-sine-sound-or-audio-wave.jpg

Indeed, all frequencies can be played back on the system; to preserve the full higher frequency detail, the finer peaks of the wave must be in phase with the playback speed, which is dependent on the quality of the filtration 'the magic'. On the fly syncing falls short o' the best-

http://thenoisegroup.com/audio-tech-blog/incredible-tool-comparing-sample-rate-converters/

There are indeed surprising double blind studies that show people cannot tell the difference between 44.1/16 dithered and higher rates. The 192khz rate does have value when playing back both 44.1khz and 48khz lossless material on the same system. If the driver is run at 192, there is an audible transparency and smoothness in the breathy high frequencies of 44.1 (as compared with a 96khz driver). As far as storage, there are good post filters, such as izotope, which allow high quality transfer between 48 & 44.1. The main reason to store in higher rates is to slow down audio in post, or to lessen the amount of 'work' required of the filter to go between 48 and 44.1. Otherwise, the higher frequency data can literally cause problems..