Level into Logic is sooo low

  • First of all, apologies to the OP for having completely hi-jacked this thread! ...At least hope this discussion is useful/interesting for some :rolleyes:

    It is strange to me that people believe to hear differences in the sound, but never mention noise or quantization noise.

    I hope you don't believe that Italian electro-technical engineers (specially those who have been revolutionizing Italian audio and hi-fi culture in the last 30+ years) don't know the theory and technique of A/D conversion... LOL


    Quantization noise is of course a well-known issue even here at South. But it's not what's discussed in the article I mentioned, which refers to recording (and reproducing) a low-level signal (for example late reverberation reflections in a hall).
    Quantization values (i.e. the number of bits used when recording) are not fixed, but related to signal levels; in fact, the signal level is expressed by the number of bits used, which corresponds to its amplitude.
    The so called quantization noise, which is a reflection of the quantization distortion, occurs anyway, because it responds to the difference between the analogic (continuous) signal and the sampled&held stepped one. It depends both on the sampling frequency (for example, 44,1 kHz) and the quantization (for example, 16-bit). Which is intuitive, because the closer the steps, the less the difference with the original soundshape.


    But the point here is that the A/D converter uses the full sampling range (let's say 16-bit, theoretically) for a 0 dB signal, but progressively less bits for expressing the level of a weaker signal. And this is what Eng. Nuti writes.
    Now, when this low-level signal varies its amplitude (for example an echo fading away), the resolution available to faithfully reproduce this fading is exactly the number of bits the converter was able to use for representing it. If the signal amplitude corresponded to 4 bits, this fading may just assume the following values: 16, 8, 4, 2, 0. In other words, you just have 5 fixed stepped levels to reproduce the fading.
    In a more robust A/D system, with 24-bit or higher quantization resolution, you find that the number of bits available at that same level is much higher (for example, 7), giving a series of 128, 64, 32, 16, 8, 4, 2, 0: almost double the "resolution", and much closer that the previous digitalization to the real signal. The perceived difference is a more solid perception of the room and its volumes, more stable stage reconstruction, firmer instruments positioning in the 3-D space. If the playback system allows it.


    As you can see, the issue discussed has nothing to do with quantization noise, which is present nevertheless.


    Generally speaking, a 16-bit sampling allows for 65,535 different levels, while a 24-bit sampling allows 16,777,215 different levels: even keeping the same sampling rate, it's 256 times denser. Doe someone really think that this huge amount of space is just ... empty?


    Of course, if all you record is a 0 dB signal (like some punk song LOL) the improvement with a higher resolution source is less meaningful than recording a classical concert live, where the ambience is a living part of the program and the dynamics may be outstanding.

    Quote

    Where can I find an A/B comparison on the web?

    The number of sites that make HD audio samples available for technical comparison is steadily increasing. You might try googling for "HD audio samples". Often you find lower and higher samples in the same site for comparison purposes.


    OTHO, many stores are now selling HD files along with the 16/44,1 CD, and you can buy both and compare them on your reference system. One of them is http://www.hdmusicstore.it (Italian only, I'm afraid).
    Neil Young has recently acquired 5 (!) companies with the goal of spreading the HD music all over the world. His word is "down with the mp3!", where the density of information in an mp3 is as low as 5% (!!) of the original recording (studio tape).



    The point here is discussing the practical limits of a technology, not a single person's ability to tell differences. We know some people have a better discriminating hearing than others, and this is a fact; it's a matter of genetic, health, education, listening pleasure and other reasons.
    Eric Johnson can tell the brand of the batteries in his stomps from the way power fades off; I know people able to tell the brand of power cables (the ones between power amp and cabs that is) in a hi-fi system (between a limited number) when you switch them. They are exceptions for sure, but those exceptions are a huge help when it comes to fine-tune real devices and understanding their limits.


    As for me, I've always thought that you have to expose yourself to the best possible experiences, in order to get better every day: most possibly, a person accustomed to listening to mp3's won't wholly catch all the improvements in better-sounding material, while the opposite is quite likely to happen. I believe we also become our experiences.


    OTOH, many people are perfectly happy with mp3: I absolutely see no bad in this. It's exactly like people happily playing any guitar they came across, and other people always looking for a better pu impedance, a better volume capacitor, a better tube set, or a better come sensitivity... or a better modeller!
    What matters, IMO, is happily express ourselves and enjoy our senses :)

  • Haha- no worries on changing the theme of the thread... still working on my original issue, but I can say I learned a little something about sound quality and file formats!

  • Gianfranco, if you like, open a new thread for this topic. Maybe in "Other Gear" ?
    I love (and hate) these esotheric topics, and there is so many!


    To give a quick reply here:
    I assume that your brief explanation of the sampling theory and signal quantization was not addressed to me personally. I am generating audio processing code on a daily basis since 16 years :)


    Let me clarify my previous claim:
    Sounds at that low level will not just disappear or change their stereo image, without other side effects worth mentioning.
    They will either get heavily modulated by quantization errors, that is clearly noticable at 4 Bit.
    Or the sound will drown in analog dither noise, but still preserve its level and stereo image, as on an analog tape.


    If somebody talks about digital audio being degraded at the last few bits of the resolution, and stating that room ambiences sound different on low resolutions, but not noticing or mentioning these unavoidable side effects, is about to create another urban myth. And there is so many.


    And I may correct you: Quantization errors are fixed at an absolute level. But of course they become more significant when the signal level decreases. But you have to turn up the volume to notice them.

  • Eric Johnson can tell the brand of the batteries in his stomps from the way power fades off; I know people able to tell the brand of power cables in a hi-fi system (between a limited number) when you switch them.


    OT
    Sorry Gianfranco, but in my experience, when I met some people with this "superman's ear" they never replicate their super-power in front of me... not even with audio-files through internet...
    Recently I tested (just for fun) one of these "super-heroes" even with the KPA (real amp vs. profile) and he failed :S


    Actually, we live relatively close (I'm from Tuscany)... if you know one of these people just let me know... I'm very curious :rolleyes:

  • Eric Johnson can tell the brand of the batteries in his stomps from the way power fades off; I know people able to tell the brand of power cables in a hi-fi system (between a limited number) when you switch them.


    Frankly, I doubt that Eric Johnson really can tell or the power-cable people. They all state that they can but there is not a single double blindfold test which actually proofs that. When they know what they hear they can tell anything for sure. But make it double-blindfold and they all fail. The problem is that nobody dares to ask Mr. Johnson to do a blindfold test. And he wouldn't agree anyway as that could hurt his reputation.


    I've worked for a Keyboards and Recording magazine in Germany for 10 years and we always wanted to do such tests (the cable thing is much discussed in our sector as well) but the people who had the strongest opinions would never show up unfortunately.


    I know of a vendor of High-End Speakers who once did a blindfold test with speaker cables (as he didn't believe in cable-myths but wanted to get to the ground of it). And as opposed to the power cables, a speaker cable could really matter (if only in theory). Some cable manufacturers (very promninent names) sent their experts with their quite expensive products and nobody could pick his/her own cable let alone tell a 1.000 € cable from bell wire ;) I have the deepest respect for those experts as they did at least show up. And I don't mean to be sarcastic about it. Most Hifi-people have strong opinions but would never show up to proof them. The guys at the Hifi-Magazines being by far the worst - at least over here in Germany. Whatever I read about CD's, how they work and why it's required to use 1.000 € cinch cables, when I read about Jitter, pits and lands and how they alledgedly represent digital 0 and 1 - it all shows only one thing: Those people don't have a single clue about how that CD-thing actually works technically.


    Anyway, to this very date there is not a single double blindfold test that proved at least somebody had succeeded in telling cables apart. Not a single test! Considering that HiFi-discussion is at least 50 years old I would have expected proof of that - if it really existed ;)

  • Thank you.


    And I believe the same applies to double sampling rate (88.2 kHz, 96 kHz). Has anybody heard of a scientific blind test yet? Does anybody really hear the difference? And if so, have they determined the reason behind it?
    It's crazy to make people waste half of their calculation power and disk space.


    Even more esotheric: Analog summing!
    Here's a review of a test:


    http://www.soundonsound.com/sos/jun04/articles/qa0604-5.htm


  • Short and to the point! Also dated back to July 2004, since then digital technology has taken even more numerous steps forward.


    Still, there always be those who "hear" the differences...Possible? Maybe, I don't know . Do I care? Not as much.


    I don't even remember how much money I had spent, back in my stubborn days, to buy really expensive amps, loudspeakes and cables.
    The result was that I wasn't listening to music anymore, like when I was a teenager from a shitty mono record player, instead I was after the next big upgrade to come closer to the "live" feeling. When I realised it, I sold everything, bought pretty basic things and listened to the music and only music again.
    Bottom line, I re-enjoy listening to old and new staff, and got rid of the "hm..the mids are not naturally produced, I should buy that cable, or that reference speaker".
    I listen to mp3 on my iPhone going to work and wav and flacs when at home. And you know what? I enjoy listening to any the same.


    As for digital recording, I have very poor ears. I can live with 44, 48, or whatever....as long as my recordings sound as I want and the mixes are as good as my ability to mix.


    :)

  • There's an on-going discussion though on the effect that inaudible, ultrasonic frequencies have on our brains and how this influences the way audible sounds...well, sound. Profoundly deaf people were found respond to respond to ultrasonic frequencies, so it follows that 'hearing' people would too, without knowing it. These HFC's (high frequency components) may even go some way to explain why recorded music will never sound quite like live music. Of course notes (of complex sounds) also contain many harmonics, odd and even multiples of those notes. These are within our hearing range, but also beyond it. Just as the audible harmonics shape the character of the sounds we hear, it is thought that ultrasonic harmonics also have an effect. Since a sample rate of 44.1kHz means no note beyond 20 kHz is even possible you can see how this may curtail our response to 'real' sound. So it would follow that 96kHz is going to help a great deal in this respect.


    Of course this may all be complete bollocks

  • @ Ballantine: Bebo Moroni, from AudioReview. He should live in Rome.


    @ Garrincha: I'm not sure I made or not a translation mistake, my "power cable" was meant to translate "cavo di potenza", which is the Italian form for meaning just the cable feeding a passive cab :)


    As for those who state they have certain abilities, I'd not want us to make the mistake of believing that since some have revealed a fake, then there's no-one genuinely gifted. Of course such an ability is not for everyone.
    But, generally speaking, I mentioned such people just to mean that not everyone hear the same things. I'm sure that for at least 5.99 billion people on this planet, the sentence "hear? this combo breaks up like a typical class A EL34 amp" means almost nothing. And I include several musicians among them.


    There are many people able to do or feel things that for me are just science-fiction... I know there's more out there than just me :) Could have anyone, in March '09 believed someone would run 200 m in less than 19"20 within three months?


    A more technical post will follow if I have time to collect something interesting :P

  • Please forgive me for being so ruthless with these topics.
    This is another unscientific chain of reasoning with the potential for creating an urban myth.


    The natural speech contains loads of ultrasonic energy that will hit your ear and body when somebody talks or sings to you straight by air, not through a recording.
    Especially the noisy consonants can be considered at least having a pink noise spectrum. Let us assume that the consonants will be lowpass filtered by 6 dB per octave due to natural effects. Starting from 20 kHz the level of the noisy consonants would still sound only 36 dB attenuated at 1.28 MHz (!). Probably half of the acoustic energy lies in the ultrasonic spectrum.
    If deaf people could perceive ultrasonic frequencies at a regular level, they were not deaf, simply spoken.
    And the hearing humans should perceive it, too.


    If ultrasonic sound was so important, why would expanding our audio systems to 96 or 192 kHz sampling rate give a satisfying result? This is only one or two of so many octaves in the ultrasonic spectrum. This is it?


    My conclusion is: the trend for higher sampling rates is driven by urban myths and commercial interests.

  • Regardless of the science of what's "necessary" or not the practical reality is we don't do 41kHz or 42kHz, or any other, but 44.1, because it's a standard developed for digital video cassettes, not for hi-fi and general audio. 48kHz is also a standard which you must use if you want to work in film or modern television, so the fact that the KPA doesn't support the standard is kind of sad given that you must then use the analog outs only for that, which means no re-amping without an extra re-amping box and greater latency, it's just pride.


    48Khz gives an even bigger shelf for the lo-pass, which is why some may hear a difference with poorer quality convertors, working at 96kHz is overkill, but HD supports it and occasionally it is asked for, it's not one that I'd ever use but 48 I use all the time.


    It's up to the user and client to decide how they want to record, not the manufacturer of one box in a studio. 44.1 is just one standard out there, it's not the only one, it only exists because of PAL limiting the options. Standards aren't merely arbitrary numbers that people throw out there, they're based on what fits certain technologies and the power supply, and they become standards by being used. Please support 48kHz.

    Edited 2 times, last by Per ().


  • Every fully grown male individual with golden ears, that really could hear the shortcomings of the present audio and sampling science and technology would expose himself to blind A/B comparisons and become really famous by showing the world that the science has overlooked something. And there are thousands of scientists (incuding me) waiting for such a case to shake the world of audio science.

  • Regardless of the science of what's "necessary" or not the practical reality is we don't do 41kHz or 41kHz, or any other, but 44.1, because it's a standard developed for digital video cassettes, not for hi-fi and general audio. 48kHz is also a standard which you must use if you want to work in film or modern television, so the fact that the KPA doesn't support the standard is kind of sad given that you must then use the analog outs only for that, which means no re-amping without an extra re-amping box and greater latency, it's just pride.


    48Khz gives an even bigger shelf for the lo-pass, which is why some may hear a difference, working at 96kHz is overkill, but HD supports it and occasionally it is asked for, it's not one that I'd ever use but 48 I use all the time.


    It's up to the user and client to decide how they want to record, not the manufacturer of one box in a studio. 44.1 is just one standard out there, it's not the only one, it only exists because of PAL limiting the options. Standards aren't merely arbitrary numbers that people throw out there, they're based on what fits certain technologies and the power supply, and they become standards by being used. Please support 48kHz.


    Providing multiple sampling frequencies while preserving the unaltered sound is possible, but quite challenging for us.
    The Profiler is not the only hardware device providing only one sampling rate by good reasons.


    To clarify some more urban myths:
    A reamping box does not introduce latency.
    Oversampling DA-converters (since 15 years and more) do not use steep lowpass filters any more. This problem is history in our generation. Actual converters work in the Mhz region. The spectrum of a guitar amp and cab merely exceeds 12 kHz, as you know.

  • Every fully grown male individual with golden ears, that really could hear the shortcomings of the present audio and sampling science and technology would expose himself to blind A/B comparisons and become really famous by showing the world that the science has overlooked something. And there are thousands of scientists (incuding me) waiting for such a case to shake the world of audio science.

    CK, You're going break the stereotype of humorless Germans, if you don't try to restrain your fulllygrownmale self. :D


    I really enjoy your posts; and applaud you for your attention to the website....it is, in my experience, rather unusual - but most welcome.

  • I can't help feel a bit lame here, but whaddya gonna do??
    It turns out that my cables were the issue. I was using some TRS cables I had lying around here to connect the KPA direct outs to my RME interface.
    Well, since I thought I owed it to the KPA to have some good, designated cables, I ordered some Monster Studio Prolink XLR female -> TRS cables and they arrived today. When I replaced the old cables with these, the level into Logic just about tripled!
    Point is this: make sure your cables aren't the cause of any problems here! I'm loving this Kemper more and more each day.

  • That makes perfect sense and I'm a bit embarrased myself for not thinking about that and giving you a hint. The RME has a balanced input with a stereo jack and the TRS of the Kemper is unbalanced. I guess it's the configuration of the jacks that made you loose half of the signal. I had that once with a Creamware A16 and got away with plugging in the jacks only halfway in when connecting unbalanced synthesizer outputs to the balanced input of the A16.


    Again I'm embarrased a bit myself as I do know about it but didn't think of it when you described your problem.

  • Quote

    The
    RME has a balanced input with a stereo jack and the TRS of the Kemper
    is unbalanced. I guess it's the configuration of the jacks that made you
    loose half of the signal. I had that once with a Creamware A16 and got
    away with plugging in the jacks only halfway in when connecting
    unbalanced synthesizer outputs to the balanced input of the A16.

    This would be a perfect issue for the wikpa! Would you guys help me formalize it better?



    jtm wrote he bought an XLR/TRS cable, so – since the RME's input is TRS balanced – it seems he now uses the KPA's balanced output?