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Is there a big difference between digital (SPDIF) distributors / amplifiers?
Old 16th February 2017
  #1
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Is there a big difference between digital (SPDIF) distributors / amplifiers?

Hello,

I'm looking at purchasing a digital distributor to duplicate a single SPDIF signal multiple times.

I found two distributors, one goes for $59.00:

VDA1 | Inday

...while the other goes for $660.00:

Sonifex RB-DDA6S S/PDIF Digital Distribution Amplifier

Other than the more expensive distributor having two more outputs (which I won't need) and the ability to switch between US and UK voltages, I'm not sure why there's an 11 times price difference between the two distributors.

Is there something important I'm missing or are both distributors pretty much the same thing?

I was thinking that maybe during the duplication process, one signal might be cleaner than the other, but both companies are claiming that the duplicate signals exactly match the input signal (so I assume there can't be any added noise / artifacts - though who knows if that's really the case).

Any help would be greatly appreciated.

Thanks,
Nelson
Old 16th February 2017
  #2
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Get the $60 one.
No difference whatsoever except that the $600 one is a rip off in a pretty box with a built in power supply.
Old 16th February 2017
  #3
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Originally Posted by cavern View Post
Get the $60 one.
No difference whatsoever except that the $600 one is a rip off in a pretty box with a built in power supply.
Hello Cavern,

I sort of thought that might be the case, but was unsure.

Thank you,
Nelson
Old 17th February 2017
  #4
A few years ago I was looking for a coaxial s/pdif distribution system, but I also needed the ability to have any input be the one sending the s/pdif digital signal to any output(s). I use Steinbergs VST System Link among several DAW PC's, to send digital audio, transport commands, clock info a and midi.

What I ended up buying was a used Shinybow SB-5588 matrix switcher, which is also a distribution amplifier, I think for around $100. This is actually a device made for the video market, but also has analog audio and digital audio I/O. I only use the Shinybow for it's digital audio s/pdif.



Being true matrix switcher, I can switch any one of my DAW's as the acting master, sending s/pdif digital audio, to any or all of it's outputs, to all other DAW's simultaneously. Or any input can send to any output individually at the same time.
Old 17th February 2017
  #5
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Originally Posted by Steve Fogal View Post
A few years ago I was looking for a coaxial s/pdif distribution system, but I also needed the ability to have any input be the one sending the s/pdif digital signal to any output(s). I use Steinbergs VST System Link among several DAW PC's, to send digital audio, transport commands, clock info a and midi.

What I ended up buying was a used Shinybow SB-5588 matrix switcher, which is also a distribution amplifier, I think for around $100. This is actually a device made for the video market, but also has analog audio and digital audio I/O. I only use the Shinybow for it's digital audio s/pdif.



Being true matrix switcher, I can switch any one of my DAW's as the acting master, sending s/pdif digital audio, to any or all of it's outputs, to all other DAW's simultaneously. Or any input can send to any output individually at the same time.
Hello Steve,

Thanks for the heads up on the Shinybow.

Out of curiosity, did you notice any signal degradation when using the Shinybow?

In other words, if you were to record two signals:

Signal 1 = SPDIF Source --> Interface

Signal 2 = SPDIF Source --> Shinybow --> Interface

...did the two SPDIF signals cancel out completely, or was there some difference in one of the SPDIF signals?

Given that SPDIF is a digital protocol, there shouldn't be any conversion going on during the duplication process, but there could be slight phase / gain differences in the signal during the duplication process.

Thanks,
Nelson
Old 17th February 2017
  #6
Quote:
Originally Posted by nelsona View Post
Hello Steve,

Thanks for the heads up on the Shinybow.

Out of curiosity, did you notice any signal degradation when using the Shinybow?

In other words, if you were to record two signals:

Signal 1 = SPDIF Source --> Interface

Signal 2 = SPDIF Source --> Shinybow --> Interface

...did the two SPDIF signals cancel out completely, or was there some difference in one of the SPDIF signals?

Given that SPDIF is a digital protocol, there shouldn't be any conversion going on during the duplication process, but there could be slight phase / gain differences in the signal during the duplication process.

Thanks,
Nelson
Hi Nelson,

I did not notice a difference of any kind in the signal, when comparing going through the Shinybow SB-5588 s/pdif I/O's, VS going directly from among my PC's pro audio interface's s/pdif I/O's. The manufacturer rep communicated to me that what go's into any input, is what comes exactly out of any & all outputs. I was told the same from other manufacturers & models of these types.

By the way, I had also looked at the units you 1st had mentioned, among some others, even very expensive professional studio designed units, but decided this was the most cost effective and appropriate type of device for my VST System Link set up, due the 'matrix switching' capability, as I needed the ability to choose which PC was going to be sending my VST System Link data, to any other PC(s) at the quick pressing of some buttons.

Just to be clear, the Shinybow, and most all other s/pdif devices do NOT 'mix' or 'combine' digital audio s/pdif signals (as like an analog audio mixer does), but I don't think you were asking for that. (if anyone want that ability, look at the Roland M-1000). Or, if you need a computer to receive multiple s/pdif signals simultaneously, simply install multiples of your capable audio interfaces for the added digital I/O.

I did a lot of experimentation sending s/pdif signal among PC's. Regarding phase differences, in my own experience, this can happen when transferring digital audio signals among any devices, and seems do with a slight latency, because when I manually realign the transferred audio, the phase sound can mostly go away. Also when comparing the transferred audio's wave forms throughout an entire standard song length, the waves peaks can be ever so slightly off. But this is such a small degree that it's impossible to hear. It can only be seen by fully expanding/stretching the wave form in my DAW's audio editor. All this is WITH or WITHOUT using the Shinybow switcher/distribution amplifier.

As part of my experimentation with sending s/pdif among my DAW PC's (with and without the Shinybow), much of that was with sending & receiving identical audio parts (say a guitar or drum track etc), or a mix of parts (a song). I muted either the Left or Right side of the digital stereo audio field when transferring, and send one at a time, and I did this from different PC's. That is, if I had a song mix available, I 1st 'exported' that audio, then imported that to say 2 DAW PC's so they had an IDENTICAL wave form each. I then transferred that song from each of the 2 PC's, to a 3rd receiving PC, #1 PC sent just the LEFT side, and #2 PC sent just the RIGHT side of the same song mix to the #3 PC. Listening to the 3rd PC's Left & Right transferred audio sounded perfect.

Another clarity is that the ONLY time I could even hear any 'phase' sound, was when comparing IDENTICAL audio parts! In at least my real world, that would never happen. I came to the conclusion that when I'm sending digital audio from any one PC to any other, they're going to be totally different parts, therefore I will not be able to distinguish any oddities. For example, I was using 4 PC's with my VST System Link set up. PC #1 was sending a mix of all my VSTi's (drums, bass, key's etc, etc), PC #2 was sending a mix of everything guitar (standard guitar audio tracks, and guitar direct tracks that were processing with guitar soft-sims ), the 3rd PC was sending all my vocal tracks. The 4th PC was the 'mix PC' receiving digital stereo audio from all PC's, but sending/receiving only ONE PC's audio at a time, as I don't the ability for any one PC to receive 3 coaxial s/pdif signals simultaneously...though I certainly COULD if I installed 3 identical audio interfaces in the mix PC, giving me 3 sets of coaxial s/pdif I/O's. I actually HAVE the spare interfaces to do this, but I just haven't bothered yet.

With my particular set up, using a digital audio distribution amp (in the form of a matrix switcher/amp in my case) I do NOT have to compensate for any latency, as any signals sent into the distribution/matrix/amp, get sent OUT of ALL the matrix's outputs perfectly/simultaneously, where the receiving DAW's get sent that signal at the exact same time. I also use a monitor mixer, so when I'm monitoring each of my 4 PC's audio (from their analog outputs), my VSTi's PC, my guitar PC, and my vocal PC are all in perfect synchronization.
With just above, I'm getting a little off topic. As I mentioned previously, I'm using my s/pdif to not just send digital audio, but also digital clock, and transport commands, and I can send midi down the same line at the same time. This gives me sample accurate synchronization among PC's.

For your purposes, as far as I understand it, YOU need far less complexity than me, and just want a simple digital audio distribution amp...ie; a distribution amp that can receive s/pdif from one device, and send identical signals from each of it's outputs accurately. There are many devices available that can do this, including the Shinybow. I chose my unit because I got it at a great price used. These things were something like $1,200-$1,700 when brand new?" Check eBay, and you'll see. Oh, and by the way I actually have THREE of these types of units, the 8X8 Shinybow, an 8X8 Atlona (which looks almost identical, and I wonder if they're manufactured by the same people, but branded differently?), I also have a unit by AVocation, this one is an 8X16 unit. I bought them ALL at great prices! In theory, I could use the 8X16 AVocation to send out 16 simultaneous s/pdif signals. But that's just crazy. You may want to give some thought, as to whether you want the true matrix capability, it's nice to have, and I for one didn't pay any more for it.

An added beauty for me is that these unit's also have RCA analog audio, and RCA video (if I ever want or need it). I'm sure that some would balk that I'm using a matrix designed for the video market, but as I said, I can tell no difference between what I can send directly from my pro audio interfaces VS my matrix switchers. In FACT, with my VST System Link set up, it's actually been MORE stable. In a typical System Link set up, Steinberg recommends using a 'ring' network (daisy-chaining s/pdif among computers) which adds a very small latency with each added computer, with a distribution amp, each of the receiving PC's get the signal at the exact same time, from the sending PC. If I was totally insane, I could use my AVocation 8X16 matrix, which I also got for around $100 (or even buy a 16X16 matrix), to use 16 computers.

So do I think that using an s/pdif distribution amp degrades the signals in any way? Not in my own experience. If I haven't been clear on something, let me know and I'll try to clarify.

Last edited by Steve Fogal; 17th February 2017 at 06:34 PM..
Old 18th February 2017
  #7
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Originally Posted by Steve Fogal View Post
Hi Nelson,

I did not notice a difference of any kind in the signal, when comparing going through the Shinybow SB-5588 s/pdif I/O's, VS going directly from among my PC's pro audio interface's s/pdif I/O's. The manufacturer rep communicated to me that what go's into any input, is what comes exactly out of any & all outputs. I was told the same from other manufacturers & models of these types.

By the way, I had also looked at the units you 1st had mentioned, among some others, even very expensive professional studio designed units, but decided this was the most cost effective and appropriate type of device for my VST System Link set up, due the 'matrix switching' capability, as I needed the ability to choose which PC was going to be sending my VST System Link data, to any other PC(s) at the quick pressing of some buttons.

Just to be clear, the Shinybow, and most all other s/pdif devices do NOT 'mix' or 'combine' digital audio s/pdif signals (as like an analog audio mixer does), but I don't think you were asking for that. (if anyone want that ability, look at the Roland M-1000). Or, if you need a computer to receive multiple s/pdif signals simultaneously, simply install multiples of your capable audio interfaces for the added digital I/O.

I did a lot of experimentation sending s/pdif signal among PC's. Regarding phase differences, in my own experience, this can happen when transferring digital audio signals among any devices, and seems do with a slight latency, because when I manually realign the transferred audio, the phase sound can mostly go away. Also when comparing the transferred audio's wave forms throughout an entire standard song length, the waves peaks can be ever so slightly off. But this is such a small degree that it's impossible to hear. It can only be seen by fully expanding/stretching the wave form in my DAW's audio editor. All this is WITH or WITHOUT using the Shinybow switcher/distribution amplifier.

As part of my experimentation with sending s/pdif among my DAW PC's (with and without the Shinybow), much of that was with sending & receiving identical audio parts (say a guitar or drum track etc), or a mix of parts (a song). I muted either the Left or Right side of the digital stereo audio field when transferring, and send one at a time, and I did this from different PC's. That is, if I had a song mix available, I 1st 'exported' that audio, then imported that to say 2 DAW PC's so they had an IDENTICAL wave form each. I then transferred that song from each of the 2 PC's, to a 3rd receiving PC, #1 PC sent just the LEFT side, and #2 PC sent just the RIGHT side of the same song mix to the #3 PC. Listening to the 3rd PC's Left & Right transferred audio sounded perfect.

Another clarity is that the ONLY time I could even hear any 'phase' sound, was when comparing IDENTICAL audio parts! In at least my real world, that would never happen. I came to the conclusion that when I'm sending digital audio from any one PC to any other, they're going to be totally different parts, therefore I will not be able to distinguish any oddities. For example, I was using 4 PC's with my VST System Link set up. PC #1 was sending a mix of all my VSTi's (drums, bass, key's etc, etc), PC #2 was sending a mix of everything guitar (standard guitar audio tracks, and guitar direct tracks that were processing with guitar soft-sims ), the 3rd PC was sending all my vocal tracks. The 4th PC was the 'mix PC' receiving digital stereo audio from all PC's, but sending/receiving only ONE PC's audio at a time, as I don't the ability for any one PC to receive 3 coaxial s/pdif signals simultaneously...though I certainly COULD if I installed 3 identical audio interfaces in the mix PC, giving me 3 sets of coaxial s/pdif I/O's. I actually HAVE the spare interfaces to do this, but I just haven't bothered yet.

With my particular set up, using a digital audio distribution amp (in the form of a matrix switcher/amp in my case) I do NOT have to compensate for any latency, as any signals sent into the distribution/matrix/amp, get sent OUT of ALL the matrix's outputs perfectly/simultaneously, where the receiving DAW's get sent that signal at the exact same time. I also use a monitor mixer, so when I'm monitoring each of my 4 PC's audio (from their analog outputs), my VSTi's PC, my guitar PC, and my vocal PC are all in perfect synchronization.
With just above, I'm getting a little off topic. As I mentioned previously, I'm using my s/pdif to not just send digital audio, but also digital clock, and transport commands, and I can send midi down the same line at the same time. This gives me sample accurate synchronization among PC's.

For your purposes, as far as I understand it, YOU need far less complexity than me, and just want a simple digital audio distribution amp...ie; a distribution amp that can receive s/pdif from one device, and send identical signals from each of it's outputs accurately. There are many devices available that can do this, including the Shinybow. I chose my unit because I got it at a great price used. These things were something like $1,200-$1,700 when brand new?" Check eBay, and you'll see. Oh, and by the way I actually have THREE of these types of units, the 8X8 Shinybow, an 8X8 Atlona (which looks almost identical, and I wonder if they're manufactured by the same people, but branded differently?), I also have a unit by AVocation, this one is an 8X16 unit. I bought them ALL at great prices! In theory, I could use the 8X16 AVocation to send out 16 simultaneous s/pdif signals. But that's just crazy. You may want to give some thought, as to whether you want the true matrix capability, it's nice to have, and I for one didn't pay any more for it.

An added beauty for me is that these unit's also have RCA analog audio, and RCA video (if I ever want or need it). I'm sure that some would balk that I'm using a matrix designed for the video market, but as I said, I can tell no difference between what I can send directly from my pro audio interfaces VS my matrix switchers. In FACT, with my VST System Link set up, it's actually been MORE stable. In a typical System Link set up, Steinberg recommends using a 'ring' network (daisy-chaining s/pdif among computers) which adds a very small latency with each added computer, with a distribution amp, each of the receiving PC's get the signal at the exact same time, from the sending PC. If I was totally insane, I could use my AVocation 8X16 matrix, which I also got for around $100 (or even buy a 16X16 matrix), to use 16 computers.

So do I think that using an s/pdif distribution amp degrades the signals in any way? Not in my own experience. If I haven't been clear on something, let me know and I'll try to clarify.
Hello Steve,

Thank you for the detailed and insightful response.

Nelson
Old 19th February 2017
  #8
Quote:
Originally Posted by nelsona View Post
Hello Steve,

Thank you for the detailed and insightful response.

Nelson
Sure but NOTE; that 'not all' these matrix switchers may work with 'Digital Audio'. If you are considering any of the brands I've mentioned above, or others, make sure they 'specify' that they have a digital audio coaxial RCA. Some of the units may also specify that you can use coaxial digital audio via their video RCA connections, but this is NOT A GIVEN.

There are also models that are 'audio only', and there are models with various INS/OUTS ... like 8 IN/8 OUT, 4 IN/4 OUT, 2 IN/4 OUT etc, etc.


If you'd rather consider units designed for professional recording studio use, there was a few companies like Z-Systems (ZSys). These are discontinued and can be found on eBay, I believe they are matrix switchers as well as distribution amps.

A company called Kramer (not the guitar company I believe) also had a distribution amp only unit, model #6505 for digital audio, 1 IN/5 OUT.
Old 19th February 2017
  #9
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Originally Posted by Steve Fogal View Post
Sure but NOTE; that 'not all' these matrix switchers may work with 'Digital Audio'. If you are considering any of the brands I've mentioned above, or others, make sure they 'specify' that they have a digital audio coaxial RCA. Some of the units may also specify that you can use coaxial digital audio via their video RCA connections, but this is NOT A GIVEN.

There are also models that are 'audio only', and there are models with various INS/OUTS ... like 8 IN/8 OUT, 4 IN/4 OUT, 2 IN/4 OUT etc, etc.


If you'd rather consider units designed for professional recording studio use, there was a few companies like Z-Systems (ZSys). These are discontinued and can be found on eBay, I believe they are matrix switchers as well as distribution amps.

A company called Kramer (not the guitar company I believe) also had a distribution amp only unit, model #6505 for digital audio, 1 IN/5 OUT.
Hello Steve,

Thanks for the heads up on Kramer.

I actually don't think the VDA-1 will work for my needs, and will instead need to pick up something aimed more at a studio environment (as you previously mentioned).

I've pretty much narrowed things down to the Kramer 6505:

https://www.kramerav.com/Product/6505

...or the Henry Engineering Digimatch 2x6:

http://www.henryeng.com/hedigimatch.html

I'm leaning towards the Henry, as it can also convert SPDIF to AES (which might be useful down the road).

Thanks once again for all your help,
Nelson
Old 20th February 2017
  #10
Great, whatever suites your own particular needs now, and potentially down the road is what's best

I almost bought one the the above mentions (of yours or mine) within this thread. In the end, for me it was the full matrix switching capability that was the most important feature, beyond simple distribution of coaxial s/pdif signals. But then I'm sending digital audio between multiple full DAW PC's only, and needed anyone of them to act as the master, and slaves at anytime....without having to keep unplugging/replugging my digital coax cables in and out.

By the way, out of curiosity, I don't think you've stated what you're needing s/pdif distribution for ... ?
Old 20th February 2017
  #11
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Originally Posted by Steve Fogal View Post
Great, whatever suites your own particular needs now, and potentially down the road is what's best

I almost bought one the the above mentions (of yours or mine) within this thread. In the end, for me it was the full matrix switching capability that was the most important feature, beyond simple distribution of coaxial s/pdif signals. But then I'm sending digital audio between multiple full DAW PC's only, and needed anyone of them to act as the master, and slaves at anytime....without having to keep unplugging/replugging my digital coax cables in and out.

By the way, out of curiosity, I don't think you've stated what you're needing s/pdif distribution for ... ?
Hello Steve,

The distributor is for a somewhat complex part of my future studio setup.



Note: Click on the following link for an enlarged version:http://pasteboard.co/AJoRiMPn8.png

In my studio I will be utilizing a Kemper profiling amp.

I've heard from multiple source that bypassing the Kemper's DA results in the best signal (which is why I'm going SPDIF In / Out to the Kemper).

However, I also want to record at 96 Khz.

The problem with recording at 96 Khz is that the Kemper must be the master clock when used digitally (and the Kemper only supports 44.1 Khz @ 24 bit).

That said, there aren't any issues when the Kemper is sending a SPDIF signal to my interface (as my interface has a built-in sample rate converter to up-convert the 44.1 Khz signal to 96 Khz).

It's when my interface sends the Kemper a SPDIF signal that we have the following two problems:

1. I need to down-convert the 96 Khz signal being sent by my interface to 44.1 Khz (as my interface only has a sample-rate converter on its SPDIF input).

2. The down-converted 44.1 Khz signal needs to be in-sync with the Kemper's internal clock.

Most external sample-rate converters either up or down-convert using their own internal clocks or from the embedded clock signal of the incoming signal (meaning that we'd be clocked to the interface's internal clock - as it's the source of the signal being sent to the external sample-rate converter).

To get around this, I purchased a Sonifex RB-SC1 (used on ebay), which can clock itself to:

a. It's internal clock.

b. The clock signal from the incoming SPDIF signal (which the SC-1 will either up or down-convert).

c. An external SPDIF input.

It's the third option (c) that make the RB-SC1 unique. As I can take an incoming signal from my interface and down-convert the 96 Khz signal to 44.1 Khz, while at the same time clocking the down-converted signal to the Kemper's internal clock.

To do this, I will duplicate the Kemper's outgoing signal twice. One duplicate goes to my interface (to be up-converted to 96 Khz) while the other goes to the RB-SC1, so that the RB-SC1 is locked to the Kemper's internal clock.

To do this, I need a digital distributor.

I understand that it's a convoluted way of doing things, but there's absolutely no way around it (unless I was using a software sample-rate converter to both up and down-convert signals - which would just add a bunch of manual work on my end).

Nelson
Old 20th February 2017
  #12
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Nelson, have you tried just recording the Kemper with analog outs? IMHO, with this sort of thing, it's always better to go for a simpler solution because you'll have less problems. I'm not sure what you are trying to do here is worth the effort to be honest.
Old 21st February 2017
  #13
@ Yutaka, nah that would be too easy

@ Nelson, thanks for satisfying my curiosity, yes that's quite a set up. I was going to say that 'someone else' here on this forum was mentioning the Kemper.recently ... then I realized it was YOU If you simply must do s/pdif with the kemper, that's fine. When I get an idea myself, I usually will follow through with it...as long as it makes sense. I went through great lengths to put my System Link system together with s/pdif's up the ying-yang
Old 21st February 2017
  #14
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Originally Posted by Steve Fogal View Post
@ Yutaka, nah that would be too easy

@ Nelson, thanks for satisfying my curiosity, yes that's quite a set up. I was going to say that 'someone else' here on this forum was mentioning the Kemper.recently ... then I realized it was YOU If you simply must do s/pdif with the kemper, that's fine. When I get an idea myself, I usually will follow through with it...as long as it makes sense. I went through great lengths to put my System Link system together with s/pdif's up the ying-yang
Hello Steve,

Yeah, I've been working this out for quite a while.

The thing about going SPDIF is that I'm purchasing the RME ADI-2 Pro as my interface.

It's only a 2 in 2 out analog interface.

It also has 4 more channels of digital input (2 SPDIF and 2 ADAT).

So if I plan on using monitors with the ADI-2 Pro, I'll need to use the Kemper via digital ins and outs.

Someone may say, why not just buy an interface with more inputs like an RME UFX II?

The whole reason for going with the ADI-2 Pro is that it utilizes the highest-end converters in RME's arsenal (nearly equal to Benchmark's newest high-end DA the Brooklyn and I'd assume the Crane Song's Solaris).

The added bonus is that you're getting RME driver stability; the very product I've been waiting for years.

Nothing in RME's line-up comes anywhere close to the ADI-2 Pro in quality. So yes, to get around its shortcomings I have to go about things in a somewhat convoluted manner, but when it comes to audio, I guess that's why people spend thousands upon thousands for high-end gear rather than just buying a simple $200.00 Focusrite interface and calling it a day with free plugins.

Thanks,
Nelson
Old 21st February 2017
  #15
Quote:
Originally Posted by nelsona View Post
Hello Steve,

Yeah, I've been working this out for quite a while.

The thing about going SPDIF is that I'm purchasing the RME ADI-2 Pro as my interface.

It's only a 2 in 2 out analog interface.

It also has 4 more channels of digital input (2 SPDIF and 2 ADAT).

So if I plan on using monitors with the ADI-2 Pro, I'll need to use the Kemper via digital ins and outs.

Someone may say, why not just buy an interface with more inputs like an RME UFX II?

The whole reason for going with the ADI-2 Pro is that it utilizes the highest-end converters in RME's arsenal (nearly equal to Benchmark's newest high-end DA the Brooklyn and I'd assume the Crane Song's Solaris).

The added bonus is that you're getting RME driver stability; the very product I've been waiting for years.

Nothing in RME's line-up comes anywhere close to the ADI-2 Pro in quality. So yes, to get around its shortcomings I have to go about things in a somewhat convoluted manner, but when it comes to audio, I guess that's why people spend thousands upon thousands for high-end gear rather than just buying a simple $200.00 Focusrite interface and calling it a day with free plugins.

Thanks,
Nelson
Nelson, it's hard to wrap ones head around your particular configuration, but I think I've got it. Having never used this stuff, I had a few thoughts...

Do you actually HAVE to return the s/pdif signal back into the Kemper? Or can you just run s/pdif out only? I ask because, even though Steinberg recommends returning s/pdif back into the 1st computers interface in a loop, I really do NOT have to. But I don't know much about the Kemper, and how it should be configured, or what would happen or not, if not returning s/pdif back into the Kemper. As far as my own stuff, I don't ever have a need to return my s/pdif's back into anything...as long as I choose which ONE of my devices are sending the master clock, the clock slaves receive that clock.

Your diagram shows the Kemper's 44.1 Khz MASTER CLOCK signal going into clock source of the Sonifex (after split in two by the Henry), which gets converted up to 96 Khz, going into the RME. I understand the reasoning of converting, as you want to record at 96 Khz.
If I understand what they Kemper is, it's an amp/cab guitar simulator right? Unless your intention is to plug a guitar into your RME interface, and use the Kemper in a effects return configuration? ... can't you just plug a guitar straight into the Kemper, bypassing a lot of extra steps? Although I guess the main idea is to tweak the sounds 'after' recording dry/direct. In that case, a return back into the kemper is required. Am I understanding this Kemper correctly?

Also, can the Sonifex make use of the AES/EBU outs in a way that splits up the digital signal, and possibly use an AES/BU adapter? Thus eliminating the need for an s/pdif distribution amp in the chain? Once again, I don't know what the Sonifex's capabilities really are here. But I do see unused I/O's on the AES/EBU.

Will all this up/down sampling going on, cause any added latency, and/or signal degration/distortion?

Is it possible to gang 2 or more of these RME's in one system for more I/O's if needed? (just curious). I see that the RME has analog 1/4" & cannon plug outputs I see.

The Kemper has USB and an Ethernet type jack, can they be used for you at all in your situation?
Old 22nd February 2017
  #16
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Originally Posted by Steve Fogal View Post
Nelson, it's hard to wrap ones head around your particular configuration, but I think I've got it. Having never used this stuff, I had a few thoughts...

Do you actually HAVE to return the s/pdif signal back into the Kemper? Or can you just run s/pdif out only? I ask because, even though Steinberg recommends returning s/pdif back into the 1st computers interface in a loop, I really do NOT have to. But I don't know much about the Kemper, and how it should be configured, or what would happen or not, if not returning s/pdif back into the Kemper. As far as my own stuff, I don't ever have a need to return my s/pdif's back into anything...as long as I choose which ONE of my devices are sending the master clock, the clock slaves receive that clock.

Your diagram shows the Kemper's 44.1 Khz MASTER CLOCK signal going into clock source of the Sonifex (after split in two by the Henry), which gets converted up to 96 Khz, going into the RME. I understand the reasoning of converting, as you want to record at 96 Khz.
If I understand what they Kemper is, it's an amp/cab guitar simulator right? Unless your intention is to plug a guitar into your RME interface, and use the Kemper in a effects return configuration? ... can't you just plug a guitar straight into the Kemper, bypassing a lot of extra steps? Although I guess the main idea is to tweak the sounds 'after' recording dry/direct. In that case, a return back into the kemper is required. Am I understanding this Kemper correctly?

Also, can the Sonifex make use of the AES/EBU outs in a way that splits up the digital signal, and possibly use an AES/BU adapter? Thus eliminating the need for an s/pdif distribution amp in the chain? Once again, I don't know what the Sonifex's capabilities really are here. But I do see unused I/O's on the AES/EBU.

Will all this up/down sampling going on, cause any added latency, and/or signal degration/distortion?

Is it possible to gang 2 or more of these RME's in one system for more I/O's if needed? (just curious). I see that the RME has analog 1/4" & cannon plug outputs I see.

The Kemper has USB and an Ethernet type jack, can they be used for you at all in your situation?
Hello Steve,

Regarding your questions:

1. Yes, I have to return a SPDIF signal to the Kemper.

The reason for this is that the Kemper is used to re-amp a dry signal.

A simple way of looking at the Kemper is that it uses amp profiles (which it either generates from an actual amp or are uploaded to the unit itself) to distort an incoming dry signal.

When re-amping (i.e. taking a dry signal and distorting it through the Kemper), using SPDIF is regarded as the easiest / most accurate way of re-amping by many in the Kemper community.

Furthermore, given that my audio interface only has two analog outs (which will be used for monitoring), I must instead send a dry stereo signal to the Kemper via SPDIF (unless I want to spend a lot more money on a quality patch bay - which I'd rather not).

2. Yes, I can plug a guitar straight into the Kemper, but I prefer not to.

After many discussions with multiple experienced Kemper users, it was determined that recording a guitars signal via a dedicated DI is superior to using the Kemper's own instrument input.

This is likely due to the quality of the internal components in the Kemper (as it's primarily an amp modeler rather than a high-end DI).

Also, I wish to avoid as many AD / DA conversion steps as possible.

The reason for this is that the Kemper's AD / DA converters aren't the best.

Furthermore, the AD / DA converters on my interface are actually some of the best / newest converters you can buy.

So I'm looking at avoiding steps which end up degrading my signal (such as unnecessary AD / DA conversions), resulting in an extremely high-end setup, while also controlling costs as much as possible.

3. As for the Sonifex, it cannot duplicate a signal.

The multiple outputs are "either or".

So you can either receive an incoming SPDIF signal or AES signal (but not both). You can then output either an outgoing SPDIF signal or AES signal (but not both).

There are switches on the backside of the unit to determine whether you're inputting / outputting a SPDIF or AES/EBU signal.

4. As for latency involved in sample rate conversion.

I suppose that's inevitable with any sort of processing on a signal.

That said, the SRC (sample rate converter) on my interface - and especially the Sonifex's - are some of the best hardware SRC's you can buy.

Furthermore, since I will only be recording a dry signal and then re-amping that same dry signal multiple times afterwards (not while playing), latency won't affect my guitar playing whatsoever.

The reason for this is that the DI I've purchased not only outputs a line level signal (which will be fed to my interface), but also an instrument level signal.

This instrument level signal will then be routed to a practice amp in my room (as playing guitar with no distortion can be difficult for some - including myself).

So I get no perceivable latency (when going through my DI), as the signal path from my guitar --> DI --> practice amp, remains analog without any conversions.

5. I understand this seems convoluted, but the signal path I've chosen took years to figure out (after many discussions with a multitude of forum users).

Furthermore, for around 6 - 7K, I'm able to build an extremely high-end studio using the best DI (MW1 Studio Tool), some of the best AD / DA converters (on the RME ADI-2 Pro) and using the best amp simulator (Kemper).

I know it may seem like a lot of money, but in the end, I'm achieving a quality level on par with some of the most high-end studios (obviously with a far lower channel count), while not having to pay the high-end studio prices.

As those high-end studios can easily charge $100 - $500 an hour for studio time; easily exceeding the price of my studio build over the course of a single album.

Nelson
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