Posts by Michael_dk

    From what I see, Kemper is using the usual versioning. So no one knows how many releases of OS 9.x we will get.

    So, just for the understanding. A usual software versioning (it's called semantic versioning) looks something like this:

    2. │ │ │ └────── Build number (optional)
    3. │ │ └───────── Revision number
    4. │ └─────────── minor version number
    5. └───────────── major version number

    For example: If everything is fine after 9.1.1, we could already get 10.0 as the next version and no 9.2, 9.3 or more...

    So it could be something like

    Major version: Major new features

    Minor version: Minor new features or changes to behaviour

    Revision number: bug fixes

    (The above obviously doesn't take into account stuff that is not customer-oriented, like refactoring of code etc)

    The funny thing to me is that the same people arguing that "you can't hear the difference between sample rates" are the ones arguing that "you can hear how much better it sounds being taken in SPDIF digitally"

    Funny. Never heard that from the same people.

    It's one thing if the unit isn't capable of 48k. However, there are a number of valid reasons for the higher sampling rate, including the construction of the low pass filter necessary on the back end. For one thing, 44.1 requires a much steeper filter, where 48 can use a gentler slope. You can rely on the SRC in your particular DAW to handle the conversion, but not all SRCs are created equal and the conversion can introduce artifacts. (That's the best reason for setting everything to the same sample rate rather than relying on conversion after the fact.) Now, if Kemper simply relied on SRC to produce a 48k signal then that puts the responsibility on them to deploy a high-quality SRC - so again, if the unit is actually designed to operate at 44.1 only then it's debatable as to whether that's really the best thing for them to do.

    Good point on the slope of the low pass filter.

    Isn't SRC (in the digital domain) a non-issue this days, though?

    I tried to understand the difference between 44.1khz and 48khz and its just not sinking in. So my question is: Is there a difference between both of them as far as quality goes? Or is it a compatibility thing.

    From what I have read 44.1khz is what most people use at home and 48khz is what most studios use. If this is not correct please feel free to educate me because I really don't understand sample rate.

    It's not that 44.1 kHz is used at home and 48 kHz is used by studios. Plenty of studios use 44.1, and plenty use higher sample rates.

    48 kHz is the standard sample rate if you're doing stuff for movies or TV.

    Sound quality is debatable - as to whether the human ear can detect a difference or not. In any case, if you want to use the DAWs monitoring, then higher sample rates generally have lower latency. This can be relevant if you want to monitor through plugins in the session, not just use the plugins after recording (say, you want to monitor the vocals through compressor and reverb in the DAW while recording.

    For the vast majority of us I suspect that sample rate is the least of our worries insofar as the quality of the end product is concerned :-)

    For those who rely on buying profiles to get our amp sounds though, it's a big deal and time saver for me to get the amp into my Kemper and then reliably work the gain and EQ for all the snapshots I need. That's if it works well. TBD.

    My dream scenario would be if sellers will focus on capturing the amp as a direct amp profile just at one setting, but with multiple cabs / mics / mic positions (especially with a systematic approach to this, i.e. one-inch increments of mic movement etc). Provided the liquid profile magic works as advertised.

    At the 6 minute mark he starts to explain that bandwidth limiting the input signal resolves the sine wave from a reduced set of samples. But he never quite explains how. Common sense would say a matching output filter? Oversampling, etc?

    I don't know. Magic, I think :D

    Random google search and click found me this page, which seems to explain some of it. Too technical for me to understand, at least by one quick read-through.

    This is a common and understandable error in understanding (that is absolutely not meant as a dig at you, even if it may sound like that).

    If you're interested and have the time, I suggest to watch this video. He is really great at explaining and showing what actually happens :-)

    Here’s my take, sample rate determines how frequently a sample is taken, a sample being a measure of voltage. If frequencies higher than half the sample rate are processed, aliasing occurs. The aim is to capture frequencies to 20khz with an anti aliasing filter after that point hence sample rate of 44.1khz. The bit rate determines how closely to the measured voltage a sample might be represented in the digital domain. 16 bit is accurate to 1/65536 24 bit to 1/16777216. This gives theoretical dynamic ranges of 96 and 124 dB respectively.

    This is exactly my take as well :-)

    With the dynamic range being loudest signal to noise floor from conversion process

    I think many use something like 83 dB (I believe), which is where the Fletcher-Munson curve is most balanced.

    But many mix at even lower levels much of the time, because there are benefits to that as well.

    And check periodically (and shortly) at higher levels to see if bass overwhelms.

    I'm wondering what a more "scientific" phrasing of the issue would be.

    Is the "blanket thing" actually just general EQ (it doesn't sound like that's it, exactly)?

    Is it lack of high end?

    Is it constant, or is it in the decay of the notes? Is it lack of attack/transient?

    Thanks! Much appreciated! :-)

    In digital audio the kHz stamp is describing hos fast the cycle updates, not the frequencies included in the audio content.
    So together with the bitrate it's setting the granulation of the audio. Higher resolution just means your audio is smoother, closer to analog audio if you will.

    Not true - if I understand you correctly.

    In digtal audio, the sample rate in effect ONLY describes the highest frequency included in the audio content. I.e. the Nyquist frequency (half the sample rate) determines the highest audio frequency that can be unambiguously derived from the digital information.

    (I'm not too sure about "sample rate" as used for describing MP3s, but that's not what we're discussing).

    As I understand it, the main reason some plugins work internally with higher sample rates is in order to reliably handle harmonics introduced by the plugin itself, which might lie above the nyquist frequency for the given DAW session.

    Similarly, Bit depth determines the noise floor only.

    In TJ vid, you can see that the few parameters we see, some turns greyed when he changes amp sims....

    I assume an amp with only two buttons wil be replicate with only two parameters allowed...

    What?? I didn't notice any footage of the profiler display when talking about the liquid profiles, was this in the original 20minute video?

    Good point.

    The mic choice, yes. But as for mic placement, I see that more like an eq, than something that is worth the effort of capturing many mic positions and putting them into one "liquid cab" ... like in the new cabs from Line6 or fractal.

    Man, I'd LOVE to see Liquid Cabs!

    Or even just incorporating a number of cab profile choices within a single profile

    I think the mic placement for me would be much more relevant that mic choice, due to comb filtering etc. Can't EQ that stuff out!