Replies to post #28308 on Max Sound Corporation (MAXD)

[grantastic]

Thanks for the reply. Did you ever read the web pages I linked you to previously? Somehow I think you did not.

You have never given any links or quotes from MAXD itself which explains how you came to have the "understandings" you discuss below. This confirms for me that these claims can NOT be explained, so it's up to naive consumers to make uninformed guesses about what this software is supposedly doing.

If you want actual INVESTIGATIVE data, here's another page where the myths you appear to believe are essentially disproven: https://www.androidauthority.com/why-you-dont-want-that-32-bit-dac-667621/

The conversion from analog to digital actually induces noise in the digital part of the conversion.

Untrue. Noise DOES get introduced during sound PROCESSING when it is dithered. The amount of noise required to dither a standard 16-bit deep sample is below human perception.

It also does not completely capture the whole analog signal.

It DOES capture the entire analog signal ... within the ability of human ears to hear. The CD standard sample rate can capture frequencies up to 22khz. Children can typically hear up to 20khz, and adults generally can hear only up to 17khz. So already, digital sampling is capturing sound no normal human could ever hear.

increasing the number of bits per sample from three to 16 or to 64 or to 128

Human ears cannot discern improvements in depth beyond 16b deep samples. 24b-32b deep sampling is useful because it essentially eliminates culmulative noise introduced during many iterations of post-processing.

64b or 128b deep samples are completely pointless and not used in music production obviously.

it is my understanding that the MAXD audio makes an algorithmic calculation on any existing digital audio and artificially fills in the parts the analog to digital conversion did not capture.

"Uncaptured Parts"... as in the frequencies higher than 22khz? What would be the point of "filling in" such data if it were even possible? Only a non-human could hear those sounds, which is why they are routinely removed from music during post-processing.

...

If your idea of "filling in the parts" is to interpolate what the signal voltage would be in between samples.... every DAC already does that to some degree or another; that's simply basic hardware included with even the most basic of DAC chips.

(you can hear it).

If you think you are hearing these frequencies above 22khz, please do tell us which device you are playing it on. I will get one and test your claim by playing a 25khz test tone for my dog.

Each digital sample of any part of the analog signal is not always on the money. It is sometimes high and sometimes low but on average it produces a pretty good rendition of the original analog signal.

Actually at 16 bit depth, the samples are quite precise, certainly more precise than is necessary to exceed the limits of most humans' perception.

The MAXD algorithm, as I understand it, simply restores an approximation of what was not captured in the original A to D conversion of any analog file.

So you've said, yet you don't explain how adding sounds which humans cannot possibly hear "improves quality"

Knowing that the A to D conversion may be restored, whether it has a low or high sampling size, will allow transmitters to minimize their file size by a coarse conversion(eg. 3 bits per sample, small size) and restore it on the receiving end to near perfect analog quality.

Intriguing concept. By "sampling size" it appears you mean "bit depth". Have you tested your theory?

Go ahead... get your 16b deep music file. Convert it to a 3b or 4b deep music file. Now it sounds a little worse, right?

Take that 3b/4b deep file and play it in your MAXD sound app. According to you it should automagically sound EXACTLY like the 16b file you started with.

Here's a random youtube video which compares 16b deep vs. 4b deep songs:

Good luck!