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Thread: dBVU vs. dBFS

  1. #21
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    Have you read anything on this thread...?

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    stupid stuff

    ok so first off i dont know what the heck im doing!! i haev a line 6 toneport ux2 or soemthing like that and i've been trying to lay down some tracks with cool edit and when i select my driver it dont work ...i know its powered through the usb but my cousin had it working just on that with sonar ..but i dont understand why it wont work like that on cool edit ..any suggestions?

    wes
    PS ....im VERY new to this and am horrable with computers to begin with

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    You might be better asking this as a new topic in the Cool Edit forum.

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    There is still the potential for ambiguity with RMS amplitude measurements, since some use the RMS value of a full-scale square wave for 0 dBFS (which corresponds with a 0 dBFS peak amplitude measurement), and some use a full-scale sine wave (which corresponds with typical analog RMS measurements).

    * In the case of a FS square wave = 0 dBFS, all possible dBFS measurements are negative numbers. A sine wave of larger amplitude than −3 dBFS would be clipping by this convention.
    * In the case of a FS sine wave = 0 dBFS, a FS square wave would be at +3 dBFS.

    The measured dynamic range of a digital system is the ratio of the full scale signal level to the RMS noise floor. The theoretical dynamic range of a digital system is often derived by the equation

    \mathrm{DR} = \mathrm{SNR} = 20 \log_{10}(2^n) \approx 6.02 \cdot n

    This comes from a model of quantization noise equivalent to a uniform random fluctuation between two neighboring quantization levels. For instance, 16-bit audio has a quoted dynamic range of 96.33 dB.

    To make an equivalent measurement of a system's noise floor, the full-scale square wave convention is used. A signal which fluctuates randomly between two neighboring quantization levels will measure at −96.33 dBFS with this convention.
    I give espskully the splats

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    There is still the potential for ambiguity with RMS amplitude measurements, since some use the RMS value of a full-scale square wave for 0 dBFS (which corresponds with a 0 dBFS peak amplitude measurement), and some use a full-scale sine wave (which corresponds with typical analog RMS measurements).

    * In the case of a FS square wave = 0 dBFS, all possible dBFS measurements are negative numbers. A sine wave of larger amplitude than −3 dBFS would be clipping by this convention.
    * In the case of a FS sine wave = 0 dBFS, a FS square wave would be at +3 dBFS.

    The measured dynamic range of a digital system is the ratio of the full scale signal level to the RMS noise floor. The theoretical dynamic range of a digital system is often derived by the equation

    DR = SNR = 20 \log_{10}(2^n) \approx 6.02 . n
    This comes from a model of quantization noise equivalent to a uniform random fluctuation between two neighboring quantization levels. For instance, 16-bit audio has a quoted dynamic range of 96.33 dB.

    To make an equivalent measurement of a system's noise floor, the full-scale square wave convention is used. A signal which fluctuates randomly between two neighboring quantization levels will measure at −96.33 dBFS with this convention.
    I give espskully the splats

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    I think the point is that you aren't supposed to be, and don't have to be, anywhere near clipping when recording in 24 bits.
    Jay Walsh
    Farview Recording. I am also the forum spokesmodel for Terasyne Amplification

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    Hi guys, I always have trouble configuring my mic volume....

    I'm using Adobe audition 1.5 and when I record, the bar is moving..is that the VU meter? Somebody told me that the recording should not go above -18 dB...is that the same -18 in the meter in Adobe audition 1.5??

    I don't know why, but when I record my stuff, it seem the meter ALWAYS go above -18 and it goes red 0dB in louder parts... If I lower the volume, it seem my recording is not loud enough...

    Now, I just download a podcast from a web site and some mp3 files, it seem their recording peak is around -2, -1dB in the meter of my Adobe Audition. So I hope they are not talking about limiting our vocal to -18dB...

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    The 'bar going up and down is a peak meter, not a VU meter. -18 on that meter is line level. Your signal should average -18, not peak at -18. If you are clipping, you are recording too loud. If the mix is too quiet, then you need to compress or limit it to get the volume up. The recording level and the mix volume are two separate things that are not necessarily related.

    Those podcasts were probably compressed then normalized to get the peak level at -2.
    Jay Walsh
    Farview Recording. I am also the forum spokesmodel for Terasyne Amplification

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    Quote Originally Posted by Farview
    The 'bar going up and down is a peak meter, not a VU meter. -18 on that meter is line level. Your signal should average -18, not peak at -18. If you are clipping, you are recording too loud. If the mix is too quiet, then you need to compress or limit it to get the volume up. The recording level and the mix volume are two separate things that are not necessarily related.

    Those podcasts were probably compressed then normalized to get the peak level at -2.

    Okay, if we average at -18dB at that adobe audition bar, then we can peak at around -3dB right? So it never go too red red....Oh that means adobe audition's bar is showing dBFS and not dBVU??

    But sometime when we use compression(with output gain more than 0) then yeah, our track's volume will be higher, but I always feel that the compression process may distort my sound track more or less....

  10. #30
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    Quote Originally Posted by VictorGalaxy
    Okay, if we average at -18dB at that adobe audition bar, then we can peak at around -3dB right? So it never go too red red....Oh that means adobe audition's bar is showing dBFS and not dBVU??
    dbVU is only in the analog world, dbfs is only in the digital world. You should never go into the red. Your peaks can be at -3, as long as it is only occasionally (or you are recording percussion)

    Quote Originally Posted by VictorGalaxy
    But sometime when we use compression(with output gain more than 0) then yeah, our track's volume will be higher, but I always feel that the compression process may distort my sound track more or less....
    You have your choice of either pristene dynamics and a quiet recording, or a compressed loud one. You should not be getting distortion from your compressor. You may have the attack and release times set wrong.
    Jay Walsh
    Farview Recording. I am also the forum spokesmodel for Terasyne Amplification

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