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Decoding AAC at a Higher Bit Depth and Sample Rate - Page 2

post #16 of 19
Quote:
Originally Posted by vhobhstr View Post

jaddie - I didn't know that the DAC did filtering. Thanks for explaining that. However, in your example of making a 44.1KHz file into an 88.2KHz file, if the interpolation is skewed from a straight line interpolation, wouldn't the distortion component be at half the new sampling frequency, since only every other sample is skewed? Also, if a 44.1KHz AAC were decoded at 88.2KHz, not every interpolated sample would be skewed the same amount, or in the same direction. For example, in one cycle of an 11.025KHz cosine wave, the first sample would be an original*, the second would be slightly higher than a straight line interpolation, the third: an original, the fourth: slightly lower than a SLI, the fifth: an original, the sixth: slightly lower than a SLI, the seventh: an original, the eighth: slightly higher than a SLI. The interpolated samples would fit right on the curve of the cosine wave and so no distortion component would be added. With a straight line interpolation, however, the interpolated samples would not fit on the curve of the cosine wave. In fact they would add a low amplitude, odd shaped distortion component. This distortion component would be repeated at a rate of 11.025KHz.

 
*I'm using the term "original" assuming that there aren't any imperfections in the encoding and decoding process.

Well, at least you're getting the point: resampling doesn't improve anything other than changing the sampling frequency that must be filtered by the anti-aliasing filter. Your comment about not realizing the DAC has a filter is interesting.  I wonder how many others out there don't realize that the jagged waveforms they see when they zoom in on a waveform in a DAW never actually get out of the DAC.  Could explain a lot of interest in high rates!

post #17 of 19
Quote:
Originally Posted by jaddie View Post

For example, lets take two samples at 44.1KHz, any two samples.  Now we want to make it into an 88.2KHz file.  That means there will be an additional sample added between the original two.  And where would you place the value of that sample?  Right smack between the originals.  Where else could it be?

Proper sample rate conversion shouldn't be doing linear interpolation between samples, shouldn't always produce output right between the originals when going from 44.1 kHz to 88.2 kHz.

See pictures in this post for SRC to 4x sampling rate:
http://www.head-fi.org/t/640476/resampling-explained#post_8945984

If you do linear interpolation between the original values in the 44.1 kHz top graph (the nonzero values, which are one every four samples), you end up with points far from what is shown in the bottom graph.

Problem is exacerbated with higher frequency content in the signal relative to the sampling frequency. If the sampling frequency is way higher than the frequency content of the signal, then linear interpolation is not so bad.
post #18 of 19
Quote:
Originally Posted by mikeaj View Post


Proper sample rate conversion shouldn't be doing linear interpolation between samples, shouldn't always produce output right between the originals when going from 44.1 kHz to 88.2 kHz.
See pictures in this post for SRC to 4x sampling rate:
http://www.head-fi.org/t/640476/resampling-explained#post_8945984
If you do linear interpolation between the original values in the 44.1 kHz top graph (the nonzero values, which are one every four samples), you end up with points far from what is shown in the bottom graph.
Problem is exacerbated with higher frequency content in the signal relative to the sampling frequency. If the sampling frequency is way higher than the frequency content of the signal, then linear interpolation is not so bad.

Correct of course, but how do you explain that simply?  I just used linear because I though it would be easier to understand.  Perhaps not.

 

It's a fine point anyway.  The idea is that interpolation doesn't add any useful content or detail, it just connects the existing dots, hopefully in an intelligent way, but it never adds anything that wasn't in the original.

post #19 of 19
Yeah, it's hard to say.

The main point about jaggies that show up in an audio editor not actually existing in real life, is the important part. I agree; if the DACs were just outputting a straight line between samples (or sample and hold), certainly that would motivate using a higher sample rate...
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