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Romy the Cat
Boston, MA
Posts 10,156
Joined on 05-28-2004
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Post #:
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14
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Post ID:
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19891
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Reply to:
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19890
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fiogf49gjkf0d
 cv wrote: | | I bet Yamamura is using non-causal filtering... this is all running off a music server, so could process everything offline. |
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Chris, there is no such a thing as “non-causal filtering”. Filtering is filtering, would it digital or analog, on digital domain of cause it is much easier to do 20th order. The fact that Yamamura uses music server does not say anything. He might have the filtration splitting before DAC (the way how it shall be) but then he needs to have 6 DACs. From the Stefano original article on his blog I understand that he does not use 6 DACS, so it is most likely configured like this: DAW-> I/O interface - > DAC -> crossover (with own D/A and A/D) -> amps. It might be different configuration and he might use own software based filtering of the DAW’s stream but then how the software interacts with analog power amps?
| cv wrote: | | This makes it possible to do forward-backward filtering - you run the filtered signal "backwards" through the original filter (ie reverse the FIR coefficients) and get zero phase shift....and doubled up amplitude response... Scale everything appropriately to make best use of 24bit dac range (easily done offline), add some proper dithering, and Romy's objections to digital filtering might well disappear... |
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Well, this is exactly the propose of this mental exercise – to assess if it possible to have the Romy's objections to digital filtering disappeared. I think when you think about what Mr. Yamamura you mentally employ forward-backward digital filtering-correction. For sure it might be done but there are two major conflicts in this direction:
1) The ringing the take please with brick-wall filtration is too short to be handled by conventional digital processors. Any processors run at some kind of sampling rate, let pretend that it is 192kHz PCM. Let do not look into the DSD or Double-Rate DSD as it is all crap as they do PCM editing at 352,800 Hz. If you look at the scoop then you will see the ringing at brick-wall filtration is too fact to be processed even at 192kHz. The ringing distortion are just sit between two individuals samples and therefore then can’t be corrected.
2) This is the biggest problem of all. What you think about backward reconstruction through the original filter then think what would be the reference of reconstruction. For sure it would be a complimentary slope that has to be matched, right? Well, the Mr. Yamamura configuration has no such a slope. If you split a digital steam on 6 channels, crossover them and then combine them together into one cable then you can do analyses of the complimentary slopes, recognize what does not match and do the corrections (still within the limitation of the current sampling rate). However, in case of the inhalation above there is no cable into which the 6 signals shall flow but instead it is 6 individual drivers, dispersed in space and surrounded by air. All of it has distortion rate way less comparable with what backward reconstruction might understand.
OK, let to fantasize a bit. Let pretend that Mr. Yamamura has a microphone at the listening position that has 6x120dB per octave partitions and do analyses belong the same bandwidth as the original crossover do. Let pretend that during calibration phase they run squares across the system and the microphone “get” the ARRIVED discrepancies. We can presume that manipulating the crossover and delay we could tweak something to make the arrived pulses to look more acceptable. That would be cool solution, it is possible to do and… this is exactly what I would like to hear as I do not know if it has any practical sonic merit. It needs to be heard…
Rgs, the Cat
"I wish I could score everything for horns." - Richard Wagner. "Our writing equipment takes part in the forming of our thoughts." - Friedrich Nietzsche
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