profiling sample rate?

  • I wonder if the profiling sample rate will be definitly fixed or if it will be able to change in the futur.


    IMO they use a low sample rate like 48khz and the AA low pass filter kill the tone especially for clean tones.
    It avoids to get the right tone, it miss something and sounds steril.


    I wonder if they could increase this sample rate to 88 or 96khz in order to avoid the low pass filter limitation.

  • At a sample rate of 48 kHz, the low pass filter will be at a frequency of 24 kHz. Are you telling us you are missing frequencies above 24 kHz in your clean tones? Are you even sure that the Kemper profiles at 48 kHz?

  • Yes i miss frequencies, you can hear the low pass filter action on almost all sound cards.
    That's why DSD has been created, there is no low pass filter. It's a huge advantage.


    I don't know what converters they use but my advice is to sample at 88khz then downsample for the processing if requested. They will have a better image like that.

  • So, you’re really telling me that you can not only hear, but also miss frequencies above 24 kHz in a guitar tone? I’m sorry, but I don’t believe you. Or rather, I think you’re missing something else, but it surely isn’t high frequencies above 24 kHz! In any case, I’m not sure any of us on this forum have enough knowledge on the way the Profiler actually works in order to be able to advise the inventors and manufacturers on any better ways to profile.

  • Oh, wait a minute - to the OP, do you mean being able to run into your DAW via spdif at a higher sample rate? If yes, then it’s already possible with the latest beta firmware (which should be made a stable release once enough feedback has been compiled by the team).


    I won’t open the can of worms of high sample rates...

  • I can see it now:


    A top-flight engineer wants to tweak the tone of a mic'd guitar amp, be it clean or dirty, and therefore reaches for the 24kHz LPF, figuring it needs to be opened up. Never gonna happen!


    I mean, c'mon, almost all engineers low-pass guitar tracks at less than half that frequency anyway, so what are they doing? Are they all killing the tone? No, they're ensuring that unnecessary frequencies (including noise interference) aren't there to fight or interfere with cymbals and fairy dust.


    To the OP, regarding the range concerned with tone:


    Unnecessary low end is high-passed almost-universally by default.
    Tone tweaks for body are made in the low-mid to mid-range areas (in the hundreds of Hz).
    Tweaks for cut-through (poking through a mix) are made to the mid, high-mid and high-frequency areas, but nothing in terms of clarity or "Imaging", as you put it, will be gained above the highest frequency that the cabinet's speaker/s is / are capable of producing. That's not very-high, as you'd know, 'cause you're not gonna get cream from a cabinet that has studio-monitoring-style drivers in it. By definition, cabinet drivers severely low-pass the amp's signal.


    The only exception I can think of is if you require a little room interaction (reflections) through your mic, but even then, that sense of space doesn't require the ultra-high response you speak of. In fact, the famous "Abbey Road" trick is to both low and high-pass the reverb itself, band-limiting it drastically. Plus, critically, the Kemper's Profiling process does away with room reflections as discrete entities.


    The imaging thing you speak of AFAIK relates to entire mixes at super-high sample rates and not individual mono sources. IOW, you could pan a 1kHz-limited mono source within a mix, and at 96kHz that mix may well portray the instrument's location in the stereo field slightly-more accurately than if you mixed at 44.1kHz. Many question this anyway, but many others swear that it's the case. Either way, it's irrelevant to the Kemper's response in rendering an unmixed, mono, single-instrument sound source.

  • Dude, imaging, by definition, cannot be improved on a mono source, and that's what the Kemper's Profiling makes - mono sources.


    "Imaging" relates to sources panned in a stereo (or surround) field. IOW, where and how-accurately does the image appear in the 3-dimensional soundstage?


    You talked about the convertors' killing the tone 'cause of the LPF cutoff, so I provided you with info on where tone can be tweaked frequency-wise.


    You talked about the imaging being affected, so I addressed that too.

  • If you don't know how converters work no need to argue.

    If you know how AD converters work, then you're familiar with noise shaping and well aware that the added top end frequency spectrum with high sample rates will carry plenty of noise that has been moved there to get an increased SNR in the "important" parts of the audible spectrum.



    That's why DSD has been created, there is no low pass filter. It's a huge advantage.

    DSD in reality is mostly snake oil because 99.9% of what you get when you e.g. buy a SACD had been produced (recorded, mixed, mastered) with 24bit (or higher) PCM anyways. I'm not aware of any "affordable" and "easy workflow" production tools in the digital domain that use DSD. Systems like Sony Sonoma or Pyramix with corresponding DSD interface hardware are pretty expensive and rare.


    I think it's safe to claim that Kemper surely has other headaches than these exotic technologies with questionable advantages and overwhelming disadvantages.

  • Many higher end studio guys believe tracking @ 96k sounds better within the audible 20-20k range. It's about detail and accuracy. Digital is always a snapshot then a gap like film rather than real life. Making a Kemper HD that does 96khz output for recording would be a killer product. I would buy one plus a hundred thousand other guys. There are other benefits to 96khz like half the latency on effects etc. It's great many guys are happy with what they have and don't want anything better however there is lot who would love 96K processing and digital HD output.

  • You can still output at 96kHz IIRC, Kanga.


    Honestly, guitar-and-bass-rig information above 20kHz is useless, if barely-existent. Your argument could conceivably apply to acoustic recordings where spacial information, "dimensionality", reverberant flutters and fairy dust play a role in believability, but the Kemper's amp->cab component is mono and Profiling doesn't capture room ambience, so positional information and "dimensionality" can only be added through the use of panning and time-based effects.


    Latency isn't an issue with the Kemper, other than for some users who can't adapt to the additional delay imposed by realtime pitch transposition.

  • Many higher end studio guys believe tracking @ 96k sounds better within the audible 20-20k range. It's about detail and accuracy. Digital is always a snapshot then a gap like film rather than real life. Making a Kemper HD that does 96khz output for recording would be a killer product. I would buy one plus a hundred thousand other guys. There are other benefits to 96khz like half the latency on effects etc. It's great many guys are happy with what they have and don't want anything better however there is lot who would love 96K processing and digital HD output.

    You are asking us to sell you snake oil, but we don't want to go into that business.


    Some details:


    It's true that many studio guys believe in 96k. But it's a believe only, and they are wrong.

    George Massenburg is one of the producers with the most understanding of audio technic, as he develops hardware as well.

    Still he failed to understand how digital audio works, and none of his experts have created a proof for better sound at 96kHz.


    Many musicians cite this old interview:


    http://www.mediaandmarketing.c…IX.George_Massenburg.html


    "We have timing cues that allow us to identify and separate images in space that let us determine where a sound is located in a room, for example. And these cues have very fine gradations - perhaps far finer than the approximately 20 microseconds available in current digital conversion techniques. Maybe we need finer resolutions - maybe down to 5 microseconds, maybe further. I can't really find any hard research numbers on this."


    The assumption that digital audio has a coarse resolution in time or spatial location, stair steps, or are only a train of snapshots like a movie, is simply and completely wrong.


    Digital audio provides a full signal response up to 20 kHz. The resolution in time is infinitely fine, independend even of the sampling rate. The assumption of Mr. Massenburg that there is a timing granularity of 20 microseconds (and thus a problem that needs to be solved by higher sampling rates) is wrong.



    I know we could make a Profiler that runs on 192 kHz for a reasonable higher price, tell everyone that Massenburg was right, and make some good additional money by selling it to you and many others. But I sleep better by not doing so.


    CK

  • Many higher end studio guys believe tracking @ 96k sounds better within the audible 20-20k range. It's about detail and accuracy. Digital is always a snapshot then a gap like film rather than real life. Making a Kemper HD that does 96khz output for recording would be a killer product. I would buy one plus a hundred thousand other guys. There are other benefits to 96khz like half the latency on effects etc. It's great many guys are happy with what they have and don't want anything better however there is lot who would love 96K processing and digital HD output.


    And one more:


    Switching to 96kHz for lowering the latency would truly work, but is quite a bad idea.

    If your computer takes your project on 96 kHz, you can run on 48kHz as well but cut the block size setting in half, to achieve the same low latency.

    As you didn't waste half of your computer for the double sampling rate, there is chances to cut the block size to a quarter even.


    In the end, higher sample rates don't induce lower latencies.

    On the contrary, reasonable sample rates guarantee same low latencies, with a good chance to bring down the block size even more.


    CK

  • Many aspects of the recording chain, including many A-D convertors and plug ins, simply sound BETTER at 96k


    with respect, some of us who make records for a living actually know what we hear.

    George included.


    No one is saying we are hearing things above 20k.

    what we are saying is we hear the artifacts created in the audible range.


    Early proponents of digital recording were emphatic that 8 bit sounded "perfect".

    We've heard it before.

  • Many higher end studio guys believe tracking @ 96k sounds better within the audible 20-20k range. It's about detail and accuracy. Digital is always a snapshot then a gap like film rather than real life.

    That's not exactly true. All frequencies below Nyquist are reproduced at the correct level, to within the resolution available by the available bit depth. Dithering (the intentional noise mentioned elsewhere in the thread) makes the reproduction even more accurate by removing bias caused by steady-state signals at levels too close to a quantization boundary.


    Once the D/A takes over, the resulting output analog signal has no "gaps", and will correlate to the original analog signal in terms of harmonic content. It's not at all like film vs video where the frame rate relies on persistence of vision to fill in between frames; the digital frames are "filled in" by having a harmonic content higher than anything that is audible (higher than transmitted by just about any analog equipment, and not reproduced by any consumer audio equipment).


    What information do you believe is not being captured by the digital stream, and why would it require a higher sampling rate to be represented in the digital stream? Higher bit depth can absolutely result in a more accurate reproduction, as will better analog components and intentional dithering...but not sampling rate.


    All of this is very well-developed science, and hasn't changed since I studied it in grad school back in the early 90s.


  • Todays AD converters do NOT sound better at 96k, since they run sample rates in the MHz range like an SACD. They do not adapt the sample frequency to 48, 96 or 192 kHz. No change in sound. They don't funktion the same way as they did 20 years ago.


    Some PlugIns sound better at higher sample rates, because they are not well done. The Profiler would not sound better at higher sample rates, because we use higher sample rates internally at different places (sometimes in MHz numbers as well) in the signal chain, whereever it's required. This is how good DSP software is made. Selling a "96 kHz" Profiler that utilizes several sample rates internally would be snake oil.


    If there were artifacts in 48kHz or 44.1 kHz signals, they could easily be revealed.

    F.e. by making a 96 kHz recording, downsamping it to 48 kHz, substracting the results and see what artifacts stay.

    I have not found such proof on the internet.


    No DSP expert has ever stated that 8 bit sound is perfect, or even usable for high fidelity audio.

    They have stayed with their same thesis about how digital audio is well done, since the beginning.

  • you're suggesting that plug ins "aren't well done" doesn't really change the FACT that many of the plug ins we (who professionally make records for a living) USE every day sound a lot better at 96k.


    likewise, I'm sorry but I, like George and every other audio professional I know, will continue to use my own ears and judgment in service of my clients... and that still tells us that, no, not all modern convertors sound the same at 44.1 as they do at 96k.

    they don't.



    I was making records for living when the first 8 bit digital started to be rolled out.

    they indeed touted its "perfect sound".

    at NO stage in the development of digital audio (which has unquestionably improved greatly over the decades) did anyone making the stuff say "well it's not perfect YET but..."

  • just to add... none of this is a knock on the Kemper Profiler, which I love and use everyday both in the studio and on the road, live.


    it's just my reason for using it as an analogue output device.

    but i have absolutely no problem with that.