SPDIF out to interface quality...

Quote from MikeH

I am pretty sure this has been covered... But I just want it directly stated pure and simple....

Using kemper -> SPDIF out -> Audio Interface -> USB -> DAW

This is a COMPLETE digital chain with no analog to digital conversion AD/DA correct? Am I also correct in assuming, the "quality" of the audio interface, does not matter 1 bit as the signal is digital? Is there any reason for me to not use the absolute cheapest audio interface I can buy that supports SPDIF? Of course, there are going to be various features and things different interfaces may offer, but, if my only goal is to get kemper -> daw, why would I need anything more than the minimum/cheapest option?

Yes, yes and yes. I'm using an ESI U24XL to record through s/pdif. Sounds great. (Other not so good adjectives could be used to describe their drivers and customer support though).

Yes to your last question because it's always useful to have a usable mic pre for vocals or acoustic for example.

I tested the spdif out on the Kemper to Focusrite Saffire Pro24 to Firewire and compared it to running from Kemper main out into a Grace pre to Mytek DA converter to Spdif Toslink in. There was a small difference. The benefit to using the main out to a good AD converter is you are not limited to the clock rate in the Kemper. Higher bit rates generally have more "air" and a little more detail. I still noticed this switching between different bit rates when using the main out to AD though I am not sure why as the digital bit rate in the Kemper is set at the lower.

On a related topic. I can confirm that not all Spdif Toslink decoders are created equal. Running mic to pre to AD to Toslink on mac sounds significantly better that mic to pre to AD to toslink on Saffire Pro24 to Firewire.

The cheapest or the most expensive S/PDIF interface ... It MUST have an option/setting to sync to the external digital audio device.

Cheers,
Martin

True, or you will get static.

Quote from deadpan

I tested the spdif out on the Kemper to Focusrite Saffire Pro24 to Firewire and compared it to running from Kemper main out into a Grace pre to Mytek DA converter to Spdif Toslink in. There was a small difference. The benefit to using the main out to a good AD converter is you are not limited to the clock rate in the Kemper. Higher bit rates generally have more "air" and a little more detail. I still noticed this switching between different bit rates when using the main out to AD though I am not sure why as the digital bit rate in the Kemper is set at the lower.

You mean bit rate or sample rate? Since the Kemper is already set to 24 bit.

Did you try to see if you could still hear a difference between the tracks when you bounced to 16 bit 44.1k?

I know that higher sample rates give you a better sound during mixing and some plugins benefit from it, so I'm curious.

Kind of the same topic on SPIDF

How would you guys go from guitar>monster cable>Kemper>DIGI192HD>DAW? SPDIF, XLR, 1\4"? Where’s the best tonal quality? I could also go guitar>monster cable>Kemper>MackieD8B>DIGI192HD>DAW

If you're always using 44.1kHz in your DAW then there's not even the smallest doubt S/PDIF is your best option

Well I usually use 48k 24bit..Would it be in my best interest for guitars to change that? I use all midi drums with samples and kemper for bass and guitar btw.

Sample rate. Bit depth was always set to 24,

So does SPIDF sound better?

Do you guys run the spdif signal from the kemper at 0db? I only ask because if I do it hits logic too hot to record with good gain staging. I like my levels around -18db maybe peaking at -12db. Therefore I have to reduce the signal output to - 6 db from my kemper as I can't control this using the Scarlett's mix control software going into logic. It all sounds good to my ears but I wondered if I was losing out scientifically speak for not running at full strength from my kemper?

This whole gain staging topic is a pretty stupid myth. It should never be done at the start of a signal chain! You always want to capture the full dynamic range offered by a device. This way you capture max quality and you're free to tweak and bend your footage in the DAWas much as you like.

If you had a $2,000 camera with a $1,000 lens and you capthre the most beautiful landscape and sunset ever ... would you switch the camera down to 0.5Megapixels just because you plan to post on Facebook and Twitter? Or would you prefer the full 22 MP so you can print huge posters as well?

Never reduce quality at capture ( recording) time.

Just my 0dB,
Martin

Theoretically, 44.1 kHZ is exactly the same as 96 or 192 kHZ for the purposes of human hearing. All frequencies below 1/2 the sample rate can be perfectly represented digitally. 22 kHZ is the top of the human hearing range.

Bit depth represents the amplitude of each sample, and as you can imagine the higher this is, the more detail that can be heard.

In practice, none of that means anything unless your A/D and D/A conversions are accurate. Higher bit depth would likely result in greater accuracy, but this depends on the converters themselves. I'd prefer 99% accurate converters running 16 bit @ 44.1 kHZ over 95% accurate running 24 bit @96 kHZ. If accuracy is held constant, then of course the higher the better. But at 100% accuracy, the sample rate is moot (concerning human hearing).

I run SPDIF at 0 db from the Kemper, but I do find lots of rigs I download are pushing the Output into clipping. I also find I can get clipping when my interface monitors between -2 and -5 db, even though it appears to have more headroom. If anything, I might boost the SPDIF send, so that my interface clips at the same point that the KPA does, which will make it easier to level my rigs. For actual recording, I think I'd do as you said and reduce the SPDIF level so that it's exactly where I want to track my guitars.

I think SPDIF is the highest fidelity option (although the 44.1 kHZ clock may not work for some people) - you bypass a set of D/A - A/D conversions and a run of cable that can result in signal degredation and increased noise. If you have to reduce the volume in the Kemper, I don't think that translates to a loss of fidelity. 24 bits translates to 16,777,216 discrete volume levels per sample. My Pod HD's volume pedal moves in 0.1% increments, which means 1,000 discrete volume levels. You CAN hear the "stair-stepping" in volume if you listen real carefully for it at high volume, but it's very, very subtle. 24 bit resolution is 16x that. CD's only use 16 bits.

If the Kemper stores this data as a floating point number, reducing the volume is a moot point - there is no loss of resolution, even if you reduce the volume by 1,000,000%. If it's not, you would still have to reduce the volume exponentially before you started to notice a loss of quality. A -6 db cut is approximately 1/2 the volume. You'd need something like a < -30 db drop before it would even be possible to hear the difference, maybe more.

Quote from Nims8582

So does SPIDF sound better?

SPDIF has less da ad conversion so it is purer though it depends on your other equipment.

Yeah I can run at 0db and not clip on the kemper or logic but it's just if I'm recording through logic then the input level is too hot for recording purposes. There's still headroom just not as much as I'd like to fit into those typical gain staging ideals. I was under the impression that the focusrite mix control software would be able to control this but it doesn't seem to, unless I've missed something. My only real solution at this point is using a trim plugin in Logic

Recording at -18 is only to avoid clipping in your converters afaik. Since with s/pdif there's no conversion it doesn't matter what level you record with as long as you're not clipping?

After you're done recording you just simply turn your track down so that it hits -18 on the peaks, which is how I've been doing it.

Well actually I just look at individual tracks for possiple clipping and make sure on the master bus peaks hit -12dB (which equals to the nominal peak level in the analog console domain) by adjusting the individual track volumes (master vol sits at 0dB obviously).

Yeah that would work but altering individual tracks doesn't lower the gain staging on the individual tracks, just lowers the volume. However it does hit the master bus so yeah that would work. I just wish I had some way of controlling on a per channel basis. Although I got some stems back from a pro mixer and each stem hits the red on occasion, but the mix sounds sick. Who know what to think.

Quote from AlexanderDoyle

Yeah that would work but altering individual tracks doesn't lower the gain staging on the individual tracks, just lowers the volume.

In the case of recording the Kemper through s/pdif it's just a digital transfer of 1's and 0's so going over the -24dB of the analog domain doesn't matter, at least that's how I see it? Entirely a different matter than with analog sound sources, preamps and a/d converters? (Ie. gain staging is not a concern, it's relatable to transfering a file from an external harddrive to your computer)

I guess ck could chime in on this?