This is a bit of a complicated thing to explain. The conversion from analog to digital actually induces noise in the digital part of the conversion. It also does not completely capture the whole analog signal. So, it is well known that an analog signal sounds better than the Digital it was converted to. This diminishment of quality may be mitigated by increasing or maximizing the digital size of the sample, ie. increasing the number of bits per sample from three to 16 or to 64 or to 128 (a very big number of digits) would dramatically increase the accuracy of the digital rendering and also reduce noise. It would also increase file size.
So, 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. This has the effect of restoring the quality of the original analog signal. This restoration is not an exact copy of the original analog sound but is close enough to improve the quality of the sound significantly (you can hear it).
If I could figure out how to paste a picture in this blog I could show you how this all works.
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. The larger the sampling size, the smaller the sampling error will be and the lower the noise level will be. Noise and losses are inherent in the A to D conversion.
I have two pictures that will show all of this very well but I don't know how to put them in here.
Usually digital audio in the form of .mpg files are already in what is referred to a compressed state. 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.
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.
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Bystander to the conversation, but I appreciate it. It would be great to see a discussion about the tech between two people who know what they are talking about.
I’m not a techie - but I wouldn’t have thought there was much issue in principle in converting analog to digital and back to analog losslessly. (Other considerations like file size aside). But I didn’t think this is what the Trammel patents propose. I thought (and I may be completely out to sea) that the Trammel process allows for data to be saved in transmission by adding value at the receiving end that provokes the brain in to thinking there is a harmonic (or something) though no data was actually transmited.
I also thought the Trammel process is analog - i.e it’s mechanism of action works on the physical apparatus within the device. It is not itself digital compression.
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