Spdif at 44100, or analog input at 96/192 k for recording

  • Hello, I just have bought a used Focusrite Sappire Pro 26 IO, and making figure of what better recording settings I must use. Previously I have a Lexicon Omega and becouse its mediocre AD/DA conversors, I always record guitars thru SPdif and make the cubase projects at 44100 for compatibility, but now I wonder if would be better to forget SPdif (and the 44100 limitation) and go thru analog and set higher samplerates (96k/192K). What do you think that will be better for the quality of the complete project? do you think that use analog inputs (and high samplerate) will compromise quality of guitar tracks with regard to using spdif?
    Thank you

  • Well, think that in the songs there is more things than guitars, and the benefits of more sampling rate is not only high max frecuency, but also more resolution for internal processing of the effects, mix, less aliasing...... If only the max frecuency were important, then no more of 44100 hz. would be used to record in any case.
    My question is if a D-A/A-D conversion from analog out kemper to analog in of the interface at higher sample rate will sound better at final mix than a SPdif direct digital recording at lower samplerate (with a lower samplerate cubase project)

  • It might not be common knowledge that Cubase uses a common sample rate for all tracks in a project...


    That said, I think that S/PDIF @ 44.1 should be a fine choice. Some really good music has been produced at this sample rate, and using S/PDIF, you‘ll be recording exactly what you rear straight from the Kemper.

    francisco jent - 2 powered toasters & 1 remote

  • I use 96KHz, over XLR. The reason for this is lower round-trip latency, for use with DAW plugins, and monitoring in real-time.


    If I set my interface and DAW at 44.1KHz, with 128 samples, I get an overall latency of 10-12ms. That's the time it takes for the guitar signal to enter the interface, process through the DAW/plugins, and come back out the monitors.


    When I set my interface and DAW to 96KHz, I get an overall latency of 3.65ms. It sounds great to my ears, and it feels correct, when monitoring in real-time.


    If you haven't tried it, give it a shot, it may surprise some of the doubters out there.

    Kemper Powerhead w/remote & Kabinet
    Focusrite 18i8 (2nd Gen) - Windows 10 - Ableton Live - Yamaha HS-8's - DT770 80 ohms

  • Yes, I've always recommended sticking with the analogue outs. Just one less source of future potential glitching / confusion / other issues.


    The Kemper's DA converters sound fantastic to me, and IMHO any perceived advantage gained by going digital is going to be minuscule, if not imperceptible, and that's in isolation, let alone in a mix.


    That said, for some folks, S/PDIF is the way to go for them. It's horses for courses, really.

  • Any problems reamping via analog? I would like to record a dry signal at line level and reamp it after editing my terrible guitar playing. The digital route seems like less of a hassle but I’d like a slightly higher sample rate too.

  • I was curious when I read your reply, because I also go to DAW through XLR and I set my projects to 44.1KHz and I get 10-12 ms latency.
    But I always thought that when you increase the sample rate the more latency you have. Well, to my surprise when I set the sample rate to 96KHz the latency went down to 6 ms and I don't use plugins and monitoring, just plain Kemper signal.
    Care to explain why it happens? Are the drivers optimised for 96KHz sampling rate?

  • There are some great technical explanations out there for the reason that higher sample rates, equal lower latency. The easiest way to think of it would be comparing a two-lane highway at rush hour, to a four lane highway under the same conditions: the wider the road, the more traffic that can pass through in a shorter amount of time.


    Borrowed from - https://www.reddit.com/r/audio…hz_has_less_latency_than/

    Quote

    It's best to think about it as each step in the process taking a fixed number of samples, including the plug-ins. If the encoding takes 32 samples, then it's still 32 samples regardless of sample rate, but it's going to happen just over twice as fast at 96kHz than 44.1kHz. Those slices are a lot "thinner" at 96kHz.
    So yeah latency is way lower all the way through from input to output, including through plugins because the samples are just processed way faster.
    The downside is that it puts double the load on the computer as there's now 96,000 samples per second instead of 44,100. And your file sizes are just over double as well, so it's eating up your HD space faster. If it wasn't for these expensive downsides, everyone would just be recording at 96kHz, or 192kHz and enjoying lower latencies all the time.


    Computer power used to cost quite a bit, these days any i7, or i5 CPU is up to 96/192KHz. Hard drive space isn't really an issue anymore either.

    Kemper Powerhead w/remote & Kabinet
    Focusrite 18i8 (2nd Gen) - Windows 10 - Ableton Live - Yamaha HS-8's - DT770 80 ohms

  • Here's another recent thread that discusses SPDIF and sample rates in detail.


    According to a survey by Ronan Chris Murphy, most pros use 44k/24bit for recording and mixing, higher sample rates are mostly (but not exclusively) used by amateurs.



    I don't want to repeat the discussion from the previous thread, I'd rather add another point that may be relevant:


    When you use the analog inputs of your audio interface, the preamps in the interface will colour the sound. Depending on the preamps, this can sound very pleasant (I very much like what the preamps in my Presonus interface add to the sound..), or it may not be what you're looking for. Try it out for yourself, and compare results.

  • Why can't you use 44.1k and a smaller buffer size? The lower the buffer size, the lower the latency is I've come to understand. Just like the higher sample rates, it all come down to computer power available. What would be the benefit of using a higher sample rate over a lower buffer size?

  • Why can't you use 44.1k and a smaller buffer size? The lower the buffer size, the lower the latency is I've come to understand. Just like the higher sample rates, it all come down to computer power available. What would be the benefit of using a higher sample rate over a lower buffer size?

    You certainly can use 44.1KHz, with 64, or 32 samples, but if your interface and PC/MAC aren't up to it, you'll get buffer collapse, which translates to clicks and pops, or random artifacts.


    Even with 44.1KHz set at the smallest buffer, it won't be faster, or smoother than 96KHz with a sustainable and larger buffer rate.

    Kemper Powerhead w/remote & Kabinet
    Focusrite 18i8 (2nd Gen) - Windows 10 - Ableton Live - Yamaha HS-8's - DT770 80 ohms

  • Interesting. I figured that 44.1 at a 64 sample buffer and 88.2 at a 128 sample buffer would attain the same processor load and, thus, get similar results. I admit, I’m not an expert but is how it makes sense to me.

  • The Kemper is 44.1 internally. So it shouldn't matter whether you use analog 44.1 or higher because everything above 20k is already trimmed. Also an analog 192k input operates at that clock regardless of whatever rate it actually sends to the computer is, which is what you configure. So it shouldn't have any affect on sampling quality except for the highest frequency produced, and again this won't matter because the kpa is 44.1 and already trimmed those highs.


    Best latency and sound will be a digital connection at 44.1. 192 is just going to eat HD space for no good reason AND add the ad/da conversions to the latency