 
|
Audio Asylum Thread Printer Get a view of an entire thread on one page |
For Sale Ads |
|
|
Audio Asylum Thread Printer Get a view of an entire thread on one page |
For Sale Ads |
87.74.2.174
In Reply to: REVIEW: Behringer SRC 2496 DAC Processors posted by Dynaudio_Rules on January 13, 2007 at 13:29:52:
...from (presumably) 44.1KHz?
Follow Ups:
Most ASRCs when set to run at supposedly "synchronous" sample ratios still remain asynchronous in actual operation, unless explicitly defeatable by the user or stated so in the documentation.....And if that's the case, depending on the "timing" of the output relative to the input, you can conceivably get different sonic "flavors" each time you play the disc when an asynchronous converter is running at a selected "synchronous" frequency, depending on the proximity of the triggering between input and output clocks.
a higher rate would be better than a lower one...why upsample to 88.2?
HowdyWell the ASRC chip it uses to upsample will have less error the closer to an integral ratio you get, i.e. 88.2 (= 2 x 44.1) is a lot better than 96 (= 320/147 x 44.1) On the other hand having a fixed 96k output can have it's own advantages. In any case it's not as simple as "higher is better".
-Ted
"Well the ASRC chip it uses to upsample will have less error the closer to an integral ratio you get, i.e. 88.2 (= 2 x 44.1) is a lot better than 96 (= 320/147 x 44.1)."Although the DSD based converters run by this golden ratio, the far-more-common ASRC chips do not. According to Analog's ASRC data sheets, the chips first oversample at an extremely high rate, then the output clock triggers the output to grab samples off the intermediate oversampled stream.
If all ASRCs ran "320/147", I wouldn't be complaining about it. But as of now, I cannot recommend any player or DAC that uses ASRC.
I first heard of this golden ratio from the late Julian Dunn of Prism.
HowdyIf you read the specs closer you'll see that what I said is true for ASRC. I didn't mean to imply that ASRC uses simple integer ratios as you can see where I expounded slightly in http://www.audioasylum.com/audio/digital/messages/126043.html . But the errors in interpolation depend on the ratios of output clock rate to input clock rate and their relative phase. It's a secondary effect but there none the less.
-Ted
I'm assuming there's a whole shed-load of (unnecessary) number-crunching/re-quantizing in converting 44.1 to 96KHZ.Every single sample re-quantized?
As I understand it, this is not the same as 'over-sampling' (it's done prior to A/D conversion and digital filtering), so surely using factors of the 'native' sample-rate has got to be the way to go?
looks like there is only one more step *division*
96 (= 320/147 x 44.1)
HowdyNope, it's more complicated:
This is Asynchronous Sample Rate Conversion (ASRC) vs. Synchronous Sample Rate Conversion.
With SSRC there is only one clock. Or to be more accurate the output clock is derived from the input clock synchronously: usually in a rational manner, i.e. twice as often as the input or 320 times for each 147 times on the input. The math to derive the output samples is fairly simple in principle, e.g. for 44.1 to 96 put in 319 zeros between each input sample, filter the stream at 22.05 and output every 147 sample. If you are careful you can save a lot of math by not calculating the samples you don't need and skip over all of the multiplies by zero...
With ASRC there are two independent clocks, the input and the output. In a theoretical sense you upsample the input to a high rate and then use the output clock to pick samples at the right time perhaps using interpolation between the samples when the clocks aren't perfectly aligned (and they will rarely be aligned.) ASRC has the advantage that the output clock can be very controlled (low jitter) and the DAC can be optimized for that rate. It has the "feature" that any input jitter is encoded in the data. The results of this can be either better or worse than SSRC depending on the input jitter and the jitter sensitivity of the DAC proper...
-Ted
Wow, there is a lot that goes into this whole upsamping DAC, clocking, jitter thing....all this just to get good sound out a digital media.The more I try to tweek, wire, upsample, de-jitter, and eq the nasty digital sound out of CD's, the more frustrated I get.
BTW, I did listen the 88.2 and I noticed a greater length of decay and a slight sense of air....maybe because of my room though, that added a bit more lively sound to the music....which wasnt good in my already lively room.
Ok I will have a listen.....Good thing about this unit is I can go from 88.2 to 96 with a push of a button on the fly....
Why would you pick 88.2 instead of 96K? Neither would be synchronous conversion.
Wouldn't 24bit/88.2KHz "up-sampling" leave the original 16-bit words unmolested (padded to 24 bit), and simply interpose mean-values between each sample?You think a "free (96KHz) lunch" is to be had?
All data will be newly created to match that internal asynchronous clock rate. If you want to keep the same data, you can only have one master clock, usually from the source, in which case the other clocks are synced to that frequency, so you can have true integer rate upsampling like in almost all DACS and CD players that use a digital filter (8x oversampling). You can't have a DAC that uses its own clock crystal do integer upsampling since the clock crystals can never be exact integer multiples due to initial accuracy and temperature drift.
This post is made possible by the generous support of people like you and our sponsors:
Top of Page ]
[ Contact Us ]
[ Support/Wish List ]
[ Copyright © 2024, Audio Asylum, all rights reserved ]
