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In Reply to: Re: personally, I'm waiting for a Fourier based upsampler to 88.2kHz posted by Rodney gold on February 21, 2001 at 23:46:54:
Thus, the original data isn't seen at the output. 88.2 would. That's all I'm saying. How can you say data that goes in every 22.7ms & go out every 10.4ms? The original points don't go out. The original point's interpretation leaves but not the actual data.....just my 2¢
» Mart £ «
Planar Asylum
where the speakers are thin but the music is anything but
Follow Ups:
If you upsample 44.1 to 88.2 , it does NOT add in a intermediate sample between 2 succesive 44.1 samples that has a value it sort of guestimates between the 2 , to make it more clear , if sample 1 has a value of 10 and sample 2 has a value of 20 , upsampling 2x doesnt add a value of 15 inbetween. A lot of folk are implying this and it just isnt so.
Rodney Gold
"upsampling" or more accurately said "data rate conversion" NEVER adds data nor does data word length increase the actual s/n ratio of the original word length.That is not to say there cannot be sonic improvements by the use of different filters allowed for through "upsampling" or oversampling.
Rodney its because people have a very simplistic view of sampling.But, yes a zero sample is inserted to increase the data rate but a filter is required.
The simple 2x filter is like this(this is off the cuff so cutme some slack on the coeff):
Given samples S1,S2,S3,S4
to derive the new sample value between S3 and S4
= ((-.21)* S1) + ((.64)* S2) + ((.64)* S3) + ((-.21)* S4)You can go further if you like but simple connecting the dots isn't the way.
-CAL
It does. It has to, to avoid the aliases of the original
stream staying in the pass band. Of course it won't be '15',
the result depends entirely on the filter style used.
TNT Audio Netzine
The general perception amongst some is that upsampling to 88.2khz is similar to sampling at 88.2khz , perhaps I'm not clear in what I'm trying to convey here , but essentially it is that whatever you have at 44.1 is what you have and the 88.2 upsampling from that source doesnt add more accurate info. If I am wrong , I have laboured under a misconception.
Rodney Gold
It adds a lot, though never information.
TNT Audio Netzine
Werner sagely wrote about up/oversampling> It adds a lot, though never information.
Exactly the tradeoff.
Cheers,
-Steve
However, I don't trust any algorythm so much as to exclude the original data. I'd rank 96kHz upsampling slightly above 96/24 sampling of a HDCD analog output because the DAC had no HDCD decoder chip because it stays in the digital domain but the results are pathologicly similar. That is all I'm saying.
....just my 2¢
» Mart £ «
Planar Asylum
where the speakers are thin but the music is anything but
unless the prorammers are complete morons these original numbers SHOULD be passed through which should preclude too much artistry in processing the extra points. That's my concern anyway.I agree the 15 number is completely erronious. Usually, the algorythm is Guassian & requires lots of data with a digital time delay. Then, is feedback with digital data to preclude frequencies above &half Nyquist.
However, I think a naturally decaying HF treble note in a busy song has too few points to reconstruct it properly. The answer is less than unique. I think an intelligent Fourier scheme could be employed to determine if its on any other channels. If it is, more points can be used to determine the true nature of the wave. Plus, one could use a naturally decaying curves as the spline routine instead of some mindless parametric schtick. I could be full of it, but would be worth a look.
....just my 2¢
» Mart £ «
Planar Asylum
where the speakers are thin but the music is anything but
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