Watts Up...?
Sep 6, 2017 at 1:28 PM Post #331 of 4,672
The audioquest seems to be a splitter, i.e. if you want to run 1 output to two inputs e.g. biamping.

But are you sure you need any special cable, most sub-woofers has a stereo input and a internal conversion to mono, hence a standard stereo cable is all that is needed?
Yes that's my understanding of sub-woofers, so I misunderstood - using two Y splitters (one output to two inputs) will obviously be fine.
 
Sep 6, 2017 at 5:55 PM Post #332 of 4,672
Yes that's my understanding of sub-woofers, so I misunderstood - using two Y splitters (one output to two inputs) will obviously be fine.

Hmm, my Thiel subwoofer only has 1 mono input , probably just designed for 5.1 home Theater.

So that means 2 L/R output from Dave to 1 Input on subwoofer.

I'll probably just go for the Switchcraft to be safe.

http://www.switchcraft.com/Drawings/SC700CT_CD.pdf
 
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Sep 7, 2017 at 12:59 AM Post #333 of 4,672
Sep 7, 2017 at 2:39 AM Post #334 of 4,672
Another option would to add some room correction box for the subwoofer like:
http://www.dspeaker.com/en/products/anti-mode-8033.shtml

Sounds perfect for my needs as the Thiel also has another input that bypasses its internal filter.

Now it's time to think about running the other output set from Dave directly into a high sensitivity horn like one of the smaller and cheaper (limited bass) Voxativ and save on a power amp!
 
Sep 21, 2017 at 8:10 AM Post #335 of 4,672
Rob
How is Davina progressing? Have you had time to gauge the SQ improvement yet?
 
Oct 8, 2017 at 7:25 AM Post #337 of 4,672
Rob was your “Beyond Off-The-Shelf DAC Chips” lecture filmed and will it be available to watch?

Equally did you get the chance to listen to the new Revel Performa3 F208be speakers with their Beryllium tweeter by any chance? If so would value your impression.
 
Oct 8, 2017 at 9:28 AM Post #338 of 4,672
Yes it was filmed (coughing fits and all) I will ask them about the video availabilty when I do the next part today. No chance to look around the show unfortunately...

I will be posting both presentations on this thread later.

Yesterday was about timing, today is about DAC topologies and Pulse Array.
 
Oct 9, 2017 at 10:52 AM Post #339 of 4,672
RMAF2017 Seminar: Beyond off the shelf DAC chips - timing and filters

Here is the first presentation that I gave:

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Oops hit the file limit! Next post will cover the last slides....
 

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Oct 9, 2017 at 12:15 PM Post #343 of 4,672
bring on more fir filter taps and wta fpga enhancements eventually into TT2 please and thank you.( no rush)
 
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Oct 9, 2017 at 3:56 PM Post #344 of 4,672
Thanks for sharing Rob, very interesting read. I find it interesting your note that timbre depends on the transients. Surely it does (especially for the piano), however it should depend primarily on the harmonics of the notes which vary from each instrument. It is not so difficult to tell apart a trumpet from a violin sustaining a long note since their harmonics are very different.

I don't actually find it surprising that you found out that with the accuracy of the the M-Scaler it sounds better (I'm curious to listen to it now). That is logical if the implementation is able to satisfy the condition of the Nyquist-Shannon-Kotelnikov theorem. It seems to me this has a lot to do with the low pass filter used on CD audio, wasn't for it you would not be able to mathematically reconstruct the original signal and the end result would not be as good. Don't forget that if using Whittaker-Shannon interpolation to reconstruct the signal, the Nyquist criterion is not anymore a sufficient condition - it is a must.
 
Oct 9, 2017 at 7:12 PM Post #345 of 4,672
I was at both RMAF talks -- let me say they were the best talks on audio I've heard in a long time. I feel so fortunate to have chosen this year to drop in on RMAF and have it coincide with your talks (I believe you said this was your first talk in the U.S.)!

I did have a philosophical question I wanted to ask at your first talk but we ran out of time.

I get your premise. We need a pretty good (many taps) approximation of perfect Shannon interpolation in order to construct an analog signal that captures sufficient transient details to satisfy the ear.

However, there is an underlying assumption -- that the PCM we are converting originally came from an ADC sampling of an analog signal.

But how valid is this assumption in all cases? Here's what I mean.

Yes, clearly, if you take the master tape from a Beatles album, run it through ADC, eventually turn it into a PCM file for streaming on Tidal, etc.... then this is all good.

But what if instead, we consider a more recent song... something off the Billboard charts... something that almost certainly underwent digital post-production. The engineers doing the mixing and mastering -- well, they will be judging their work based on a more run-of-the-mill DAC process, with all its imperfections. And further, if they detect something muddy with a particular track or a particular time, they may have applied various processing to compensate for the detection that was heard through an average DAC process (perhaps that problem may not even really be there if they had a 1MM tap filter in their DAC, but they don't...).

i.e. the PCM source now no longer represents a sampling of an analog source, but rather a work of art tuned by sound engineers, approved by the artist & the record company, etc -- with all of the monitoring performed on your average DAC.

Is it still a valid goal, then, to recreate the analog signal based on a different technique as opposed to just using an average DAC as that's what the artists approved as their work product?

Even if mixing and mastering were done in analog, DACS can still sneak in. Of the high end studio monitors likely to be used in the studio, Barefoot, Generlec and PMC all do ADC/DAC within the monitor. Focal and ATC are the only two monitors I can think of with pure analog paths.

-Ray
 

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