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    #11
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    Quote Originally Posted by jcs View Post
    lol, if you disagree why not post a rebuttal with facts/evidence/examples/math so other readers can see your point? Years of work and what one does for a living doesn't trump facts/math/evidence, does it? There's an echo chamber joke here
    And maybe you can take your own advice. Perception is not math and you hear with your brain, so practice and experience do come into play. It's a bad rabbit hole many inexperience people coming into sound spend some time running down. What works on paper is great when you are dealing with machines and algorithms but it often falls short when dealing with humans. There are a ton of subjects where raw science is pretty much it. I'm not talking about that. Your facts may be valid and still not work well for perception. Also the biggest mistake in science is asking the wrong question. The right answer to the wrong question is often a much bigger waste of effort than the wrong answer to the right question.

    As a couple of examples. A collection of mics that have the same noise level on paper will have different perceived noise levels in practice. The reason is in the specific "shape" of the noise. Some mics have done a much better job of masking the perceived noise even though they are no less noisy.

    In your other post you say you will normalize to 0dBfs for the web, but in fact science will tell you that if you peak at 0dBfs you will be clipping part of the spectrum because of D/A converter overshoot. You don't perceive the issue so you ignore the science. Your experience has told you it doesn't matter for what you do.

    Which is kind of my point. Science is important and those spec sheets and sound theory are important to know. But that is just the base. You won't interpret that science and theory effectively till you have had experience seeing how the hard math translates to perception. I'm only talking about the arts here.

    On the general topic I don't generally use normalization partly out of habit and remembering how badly early normalizers worked. I do realize that the math is so good on todays computers that there is not likely to be a perceived problem. And for some SFX I have at times intentionally normalized over zero just to clip the peaks (adds some impact to things like punches).
    The "science" is that on very loud sounds your ears cant handle it and they transmit that distortion to the brain. A loud sound with slightly clipped peaks "sounds" louder than it is because it sounds closer to what a human experiences with very loud sounds. I learned that from experience...

    Also I spent a lot of time at one point in sound theory and I never heard a decent explanation for a very common perception. And that is one sounds like one and two sounds like two but three sounds like a bunch. The closest I came was in discussions with Walter Murch and his theory that humans can only perceive three distinct things at anyone time. Which actually fits with the next level that is about 90% true that four sounds like a mess.

    And finally if you work in the arts you are studying the science of perception pretty much constantly. So in a pretty real sense experience IS science.
    Cheers
    SK


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    #12
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    Quote Originally Posted by Noiz2 View Post
    And maybe you can take your own advice. Perception is not math and you hear with your brain, so practice and experience do come into play. It's a bad rabbit hole many inexperience people coming into sound spend some time running down. What works on paper is great when you are dealing with machines and algorithms but it often falls short when dealing with humans. There are a ton of subjects where raw science is pretty much it. I'm not talking about that. Your facts may be valid and still not work well for perception. Also the biggest mistake in science is asking the wrong question. The right answer to the wrong question is often a much bigger waste of effort than the wrong answer to the right question.

    As a couple of examples. A collection of mics that have the same noise level on paper will have different perceived noise levels in practice. The reason is in the specific "shape" of the noise. Some mics have done a much better job of masking the perceived noise even though they are no less noisy.

    In your other post you say you will normalize to 0dBfs for the web, but in fact science will tell you that if you peak at 0dBfs you will be clipping part of the spectrum because of D/A converter overshoot. You don't perceive the issue so you ignore the science. Your experience has told you it doesn't matter for what you do.

    Which is kind of my point. Science is important and those spec sheets and sound theory are important to know. But that is just the base. You won't interpret that science and theory effectively till you have had experience seeing how the hard math translates to perception. I'm only talking about the arts here.

    On the general topic I don't generally use normalization partly out of habit and remembering how badly early normalizers worked. I do realize that the math is so good on todays computers that there is not likely to be a perceived problem. And for some SFX I have at times intentionally normalized over zero just to clip the peaks (adds some impact to things like punches).
    The "science" is that on very loud sounds your ears cant handle it and they transmit that distortion to the brain. A loud sound with slightly clipped peaks "sounds" louder than it is because it sounds closer to what a human experiences with very loud sounds. I learned that from experience...

    Also I spent a lot of time at one point in sound theory and I never heard a decent explanation for a very common perception. And that is one sounds like one and two sounds like two but three sounds like a bunch. The closest I came was in discussions with Walter Murch and his theory that humans can only perceive three distinct things at anyone time. Which actually fits with the next level that is about 90% true that four sounds like a mess.

    And finally if you work in the arts you are studying the science of perception pretty much constantly. So in a pretty real sense experience IS science.
    OK cool- can you show evidence to support your position, specifically: "In your other post you say you will normalize to 0dBfs for the web, but in fact science will tell you that if you peak at 0dBfs you will be clipping part of the spectrum because of D/A converter overshoot.". An example of one way to prove your point would be to record a high quality speaker with a high quality mic on a D/A converter you suspect will clip and cause audible artifacts. Another way would be to use an oscilloscope and show the clipping graphically. And finally, show example audio with one normalized to 0dB, and another carefully compressed to sound as loud but not normalized to 0dB and see if people can tell the difference (without cheating by loading into an audio editor).

    I admit that using a Sound Devices DAC + Stax electrostatic headphones, I can't hear any undesirable artifacts. Nor can I hear any issues on an iPhone speaker or headphones, nor on Tannoy monitors, Sony 7506, Focal, etc. No complaints from others re: sound artifacts. Everything I produce ends up on the web as MP3/AAC, thus I suggest to anyone delivering to these formats to maximize the dynamic range of their audio, including normalizing for those targets. This is final product delivery vs. recording sound elements and delivering to others for additional processing and mixing. Are you guys creating final deliverables to the general public or to others for additional processing?

    Debating perception is impossible, as everyone perceives reality differently, right? Isn't that the point of science, to determine elements of reality that we can agree on are 'real', and the point of math and physics?

    For noise perception and psychoacoustics, we can perform double-blind tests on human subjects, as was done when developing MP3/AAC to determine how the human ear+brain act as filters of sound and perceptions of frequencies.

    To your question about one sounds like one, etc., this is something that can be analyzed using machine learning / AI. In previous discussions I linked to papers/videos where machine learning/AI were parsing voices from crowds, known as the 'cocktail party problem" in psychoacoustics, and folks here didn't seem to be interested (down voting the post etc.).

    EDIT: while some DACs that clipped a 0dB signal were an issue years ago, the much bigger issue was zero crossing distortion: https://www.stereophile.com/content/...gma-1-bit-dacs . While a 0dB signal is perfectly valid and any hardware producing erroneous output (clipping/distortion) is technically buggy, it makes sense to work around known bugs until they are fixed (the same is true for software). The work-around for the zero crossing bug would be to have no zero crossings, resulting in a massive loss in dynamic range (halve the signal and add 1/2MaxSample DC offset)! Sounds like these issues were solved around 20 years ago. 0dB and zero crossings won't be a problem today, especially for content consumed online (not likely going through vintage audio hardware).
    Last edited by jcs; 12-06-2018 at 06:10 PM.


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    #13
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    OK cool- can you show evidence to support your position, specifically: "In your other post you say you will normalize to 0dBfs for the web, but in fact science will tell you that if you peak at 0dBfs you will be clipping part of the spectrum because of D/A converter overshoot.". An example of one way to prove your point would be to record a high quality speaker with a high quality mic on a D/A converter you suspect will clip and cause audible artifacts. Another way would be to use an oscilloscope and show the clipping graphically. And finally, show example audio with one normalized to 0dB, and another carefully compressed to sound as loud but not normalized to 0dB and see if people can tell the difference (without cheating by loading into an audio editor).
    No need, since as a sound professional I know that work has already been done, see science!
    What you want to search for is "inter-sample peaks" and "true peak metering"
    But a quickie link you may like is this one from the iZotope Tech Blog

    There is also this AES paper
    The paper I was searching for, and I think was one of the first to bring the issue up to the sound community was by Bob Katz but I couldn't find it.

    The basics are actually pretty "duh" in that if you have two samples at 0dBfs and your signal is a sine wave then to complete the curve the converter would need to go past 0dBfs. 3dB is about the max overshoot so that is where the -3dBfs comes from and what Bob Katz was recommending in the paper I can't find.

    The problem is actually more of an issue with cheap gear, ie internet playback etc.
    Cheers
    SK


    Scott Koue
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    ďIt ainít ignorance that causes all the troubles in this world, itís the things that people know that ainít soĒ

    Edwin Howard Armstrong
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    #14
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    Here's an example with normalized audio (Schoeps CMC641 into the C300 II), and no other audio edits (gap between speaking was removal of a 1 year old making noise):



    The same video using normalized audio with the addition of an FFT EQ to reduce sibilance and an expander (Dynamics Processing in Premiere Pro CC) to remove background noise:



    The second video is an example of normalizing audio and removing noise using an expander, which when it work, tends to sound much better than FFT/spectral-based noise reduction processes. The challenge with FFT/spectral noise reduction methods is artifacts being generated (not present in original sound), whereas the challenge with expanders is tuning the transition between speaking and silence to make it seamless.

    I experimented in the editor between -3dB and 0dB and could not hear any issues at 0dB compared to -3dB using Focal or Stax headphones using a Sound Devices DAC.


     

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    Quote Originally Posted by Noiz2 View Post
    No need, since as a sound professional I know that work has already been done, see science!
    What you want to search for is "inter-sample peaks" and "true peak metering"
    But a quickie link you may like is this one from the iZotope Tech Blog

    There is also this AES paper
    The paper I was searching for, and I think was one of the first to bring the issue up to the sound community was by Bob Katz but I couldn't find it.

    The basics are actually pretty "duh" in that if you have two samples at 0dBfs and your signal is a sine wave then to complete the curve the converter would need to go past 0dBfs. 3dB is about the max overshoot so that is where the -3dBfs comes from and what Bob Katz was recommending in the paper I can't find.

    The problem is actually more of an issue with cheap gear, ie internet playback etc.
    Thanks for the links, I'm familiar with sampling and reconstruction (FFTs and Fourier synthesis)- I've written software to do this (in real-time). While it's true older hardware had issues with clipping, the point is that 0dB is a valid input signal. It's up to the playback hardware to play it correctly. If it needs to attenuate due to analog filter or sampling limitations, that's the responsibility of the hardware manufacturer. The good news is that modern hardware is spec compliant- i.e. can handle 0dB (and zero crossings!) with no issues. The difference between 0 and -3dB is minor, so if one is concerned about ancient audio hardware playback issues, normalizing to -3dB isn't a big deal.

    Two samples at 0dB sounds like clipping and/or a square wave, which I'd expect to sound distorted/square-wave like.

    I didn't hear any issues at 0dB in the above real-world examples provided.


     

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    Sound Ninja Noiz2's Avatar
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    Now I remember why I blocked your posts...

    Do what you want. Though I will point out that the first of these last two points you are ignoring the science for your personal perceptions. And in the second you are ignoring the engineering and science. Two samples at zero won't even show as a clip on most meters, the most conservative software requires three in a row and most mastering engineers would put the perceivable point somewhere between four and six samples. Even that is not going to sound like a "square wave" it's going to sound like a click. A two sample square wave would be at 24 kHz and nobody but dogs would hear it.

    Anyway I won't see your next post so go for it.
    Cheers
    SK


    Scott Koue
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    ďIt ainít ignorance that causes all the troubles in this world, itís the things that people know that ainít soĒ

    Edwin Howard Armstrong
    creator of modern radio


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    Quote Originally Posted by Noiz2 View Post
    Now I remember why I blocked your posts...

    Do what you want. Though I will point out that the first of these last two points you are ignoring the science for your personal perceptions. And in the second you are ignoring the engineering and science. Two samples at zero won't even show as a clip on most meters, the most conservative software requires three in a row and most mastering engineers would put the perceivable point somewhere between four and six samples. Even that is not going to sound like a "square wave" it's going to sound like a click. A two sample square wave would be at 24 kHz and nobody but dogs would hear it.

    Anyway I won't see your next post so go for it.
    In psychology this is called projection: "And in the second you are ignoring the engineering and science."

    Repeating samples at max scale will create a square wave (provided a cycle is happening between between +/- max scale): https://en.wikipedia.org/wiki/Square_wave


    Another reason a (max sample) square wave was discussed is the fact that it will overshoot when played back on analog hardware (I was agreeing with your point of two repeated samples causing overshoot), as seen with Fourier synthesis we cannot exactly represent a square wave with sums of sine waves (see link above and example below):


    Sample rate doesn't matter and a dog's hearing is irrelevant: it's math and physics of Fourier synthesis and related to digital to analog converters in practice.

    The examples posted in this thread illustrate the concept that normalizing before NR or expanding (Dynamics Processing) work fine even at 0dB: modern hardware won't overshoot/clip/distort- it's now handled correctly (same as the zero crossing bug).

    Anyone can try these methods and see the truth for themselves.


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