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ALK ES500T extreme-slope network design ready


Al Klappenberger

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31 minutes ago, Marvel said:

I've been wanting to get this to add delays and eq for a pc based crossover. I could do a two way with the interface I already have. A three way if I figure out part of the cofiguration

 

Kinda seems like a bit of overkill for a crossover.

 

Filters and EQ are about the easiest things to do in audio. (They're also the easiest things to do wrong.) I wish that someone like miniDSP would offer an API (application programming interface). I'd be very interested in working with something like that to provide custom DSP crossovers. The systems that I make for myself work fine for me, but they're not friendly enough for general consumption.

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20 hours ago, Edgar said:

 

Believe it or not, I've found that it sounds best (to my aged ears) at 48K.

 

Of course, for direct digital connections I use the sampling rate of the source, usually CD at 44100.

 

After I submitted this post, I got to thinking -- and realized that almost all of my listening lately has been to streaming sources. Since they sample at 48K or less, I guess it's no wonder that 48K sounds OK on the Steinberg system. Sampling something at 96K or 192K after it's already been sampled at 48K won't recover any lost information.

 

I'm going to listen to some vinyl sampled at higher rates, and see if things change.

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I would be interested to hear your impressions.

 

While the blue cat plugin allows for some complex uses, it would make it much easier to set up a 2, 3, 4 way crossover. Add delays and eq to fine tune. It's not an outrageous price (although it would be nice if there was an LV2 version). My DAW works better in linux and I could run a lightweight distro with less overhead than Windows. Unless I put in on a Mac mini I have, but I would rather not.

 

Bruce

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On 7/11/2018 at 5:39 PM, PrestonTom said:

I simply use a 4 channel stepped attenuator. 

 

Could you tell use where you purchased yours? Some of us do get it about the Gain Game that plays heavily into the S/N equation, cause by super efficient horns drivers.

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16 minutes ago, ClaudeJ1 said:

Could you tell use where you purchased yours? Some of us do get it about the Gain Game that plays heavily into the S/N equation, cause by super efficient horns drivers.

It is not something you purchase, it is something you build.

 

Check eBay for four channel rotary switches with about 23 steps. Be sure to get "make before break" switches. Find one of the calculators online and then purchase the appropriate resistors. Since it is before the amplifier, they can be 1/4 watt. Then prepare for a evening or two with the soldering iron. The design needs to account for the source's output impedance and the load's input impedance (with all the usual compromises/goals of a "passive pre-amp" and, yes, know that is a misnomer). The rotary switch does not need to be expensive ("boutique") nor do the resistors need to be fancy.

 

BTW, it is not just the S/N ratio you need to worry about with gain structure. It is also whether you are "using" all the bits in the ADC

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1 minute ago, PrestonTom said:

BTW, it is not just the S/N ratio you need to worry about with gain structure. It is also whether you are "using" all the bits in the ADC

Yes, that is what I meant. Getting all the high bits before it goes back to Analog to the power amps.

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15 minutes ago, ClaudeJ1 said:

Yes, that is what I meant. Getting all the high bits before it goes back to Analog to the power amps.

 

We are on the same track. The problem frequently goes un-noticed since some of these DSP units are designed for pro use (about anywhere from 5 to 10 times the nominal voltage compared to home audio). So the user can be losing 3 or 4 bits without knowing it, and that is just the beginning.

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30 minutes ago, PrestonTom said:

 

We are on the same track. The problem frequently goes un-noticed since some of these DSP units are designed for pro use (about anywhere from 5 to 10 times the nominal voltage compared to home audio). So the user can be losing 3 or 4 bits without knowing it, and that is just the beginning.

Exactly. It's pretty basic stuff that sometimes gets ignored. That being said, even if we don't operate in the high bit area by keeping the signal high and attenuating back down when going to analog, we still have more signal to noise ratio than necessary when it comes to music recording/compression/limiting on all but the purest of program material..........full symphonies, which PWK claimed only neede 17 db of headroom. We have 135 db to work with on Blue Ray Audio standards, UNCOMPRESSED. Yet, music producers have gone insane with this "loudness wars" going the other direction. Unbelievable.

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On 7/13/2018 at 4:22 PM, Edgar said:

 

I'll often use Matlab to do the arithmetic for traditional filters (Butterworth, Bessel, etc.) if I happen to be using them, and/or to do bilinear transforms, FFTs, etc. But I consider it a tool rather than a design method. Signal processing is my specialty, so, as they say, "It's what I do."

 

I run my audio under various flavors of Windows, just because it's what I'm most familiar with. I've done it in MacOS CoreAudio and Linux ALSA at various times, but I find ASIO to be the most user-friendly and ALSA to be abominable. CoreAudio and Wasapi are supposed to be very good, but I have very little experience with them.

 

Can you explain how you connect the PC to your audio equipment?  I assume you have an active multi-channel system.

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42 minutes ago, mark1101 said:

 

Can you explain how you connect the PC to your audio equipment?  I assume you have an active multi-channel system.

 

Glad to.

 

First a little background: The Steinberg UR824 ADC/DAC supports ASIO, which stands for "Audio Streaming Input Output", a bit-perfect audio transfer standard that was created by Steinberg back in 1997. ASIO is used extensively in Pro Audio, and has been kept up-to-date as new audio formats have emerged. It is not limited to Steinberg equipment; many manufacturers support ASIO in their products and usually mention it in their advertising copy.

 

Analog sources go to the analog inputs of the UR824. There are input gain controls so I can adjust each input according to the voltage level of its source.

 

Digital sources go to the digital inputs of the UR824. It can handle up to 192kHz, but my digital sources are all 44.1k or 48k.

 

The ASIO driver in the UR824 transfers the audio to the PC via a dedicated USB 2.0 connection. The ASIO SDK (software development kit) provides C++ language source code for all of the audio data handling; all I have to do is program the signal processing that I want to perform and plug my subroutines into ASIO in the right places. In my case I implement crossovers, a little EQ if needed, and also a proprietary multichannel upmix algorithm that I developed. I compile and link the resulting program, and it executes as an application on the PC.

 

The outputs from my subroutines are transferred by ASIO back to the UR824 through the same USB connection. There they are passed through DACs in the UR824 and sent to the amplifiers.

 

It's not as easy it sounds!

 

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22 hours ago, Edgar said:

 

Glad to.

 

First a little background: The Steinberg UR824 ADC/DAC supports ASIO, which stands for "Audio Streaming Input Output", a bit-perfect audio transfer standard that was created by Steinberg back in 1997. ASIO is used extensively in Pro Audio, and has been kept up-to-date as new audio formats have emerged. It is not limited to Steinberg equipment; many manufacturers support ASIO in their products and usually mention it in their advertising copy.

 

Analog sources go to the analog inputs of the UR824. There are input gain controls so I can adjust each input according to the voltage level of its source.

 

Digital sources go to the digital inputs of the UR824. It can handle up to 192kHz, but my digital sources are all 44.1k or 48k.

 

The ASIO driver in the UR824 transfers the audio to the PC via a dedicated USB 2.0 connection. The ASIO SDK (software development kit) provides C++ language source code for all of the audio data handling; all I have to do is program the signal processing that I want to perform and plug my subroutines into ASIO in the right places. In my case I implement crossovers, a little EQ if needed, and also a proprietary multichannel upmix algorithm that I developed. I compile and link the resulting program, and it executes as an application on the PC.

 

The outputs from my subroutines are transferred by ASIO back to the UR824 through the same USB connection. There they are passed through DACs in the UR824 and sent to the amplifiers.

 

It's not as easy it sounds!

 

 

Thanks for the explanation.  I am trying to understand this.  So are you using the loop back function of that unit to get the sources into the PC?  Your PC application processing actually creates the output channels, yes?  They are then returned to the UR824?  I assume the UR824 has programmable routing of its inputs to its outputs like a speaker processor?

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35 minutes ago, mark1101 said:

 

Thanks for the explanation.  I am trying to understand this.  So are you using the loop back function of that unit to get the sources into the PC?  Your PC application processing actually creates the output channels, yes?  They are then returned to the UR824?  I assume the UR824 has programmable routing of its inputs to its outputs like a speaker processor?

 

I need to be careful in my answer, because you may be using "loopback" in a generic sense while the UR824 has a separate "loopback" function that is entirely separate from the ASIO function that I described.

 

ASIO completely separates the UR824 inputs from its outputs. The inputs are the physical analog inputs (after passing through ADCs) plus the physical digital inputs. The outputs are the physical analog outputs (after passing through DACs) plus the physical digital outputs (which I am not presently using). In-between the ASIO inputs and the ASIO outputs are:

  1. A USB connection that sends digital audio from the UR824 to the PC
  2. A computer program that I write (well, they give me a template and I configure it as I please) that processes the digital audio
  3. A USB connection that sends digital audio from the PC back to the UR824

So my PC application uses the input data from the UR824 to create the output data that are returned to the UR824.

 

And yes, the UR824 has programmable routing of its inputs to its outputs like a speaker processor. It also has quite a variety of canned effects processing (reverb, EQ, etc.). I simply don't use any of them, except for the basic "mix" that selects inputs to ASIO and routes outputs from ASIO.

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