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References: <50010@apple.Apple.COM> <27517@netcom.COM>
            <50146@apple.Apple.COM>
In article <50146@apple.Apple.COM> malcolm@Apple.COM (Malcolm Slaney)
writes:

<In article <27517@netcom.COM> mcmahan@netcom.COM (Dave Mc Mahan) writes:
<>It all depends on the gain of your amplifiers and stuff.  The easiest way
     ...etc...
<
<What does 0VU really mean?  I assume it is power related but I'm not sure of
<all the nuances.....and I suspect that there is a temporal dimension to it.
<Otherwise why would Nackamichi make such a big deal about their peak VU
<meters?  Are VU's constant across frequencies?
<
<So, what is 0 VU really mean?  I want to know so that I can consistently 
<
   ...etc...

I think there are 3 interlocked issues here that have got in a tangle:

1. the nature of speech
2. what is VU
3. recording DIGITAL 8,16 bit speech on tape (tape recording parameters)

In my opinion, Dave McMahan's reply was a valid approach.  Here I'm
assuming you are going through a D/A to get DIGITAL speech (samples)
through an analog video recorder onto tape.

The binary (2's complement ?) code that represents your speech (be it
8 or 16 bit is referred to the maximum analog voltage swing your original
A/D was designed to handle - ie the peak-peak dynamic range. Lets use
5V as the example. Your D/A will also be referred to some voltage
reference (usually the supply voltage). Obviously, you can't apply a +,-5V
signal to the input of a video (audio) recorder without overloading the
circuits. So you have to scale the output from your D/A. 

Another approach  to Dave's might be to scale the maximum digital code
based on a 1KHz sinewave so that with your recorder gain control flat-out,
you insert sufficient external attenuation so that it gives you 0VU on
your meter. This will guarantee that you never saturate the tape, but
it *won't* necessarily give the best use of the available dynamic range
(of the tape).  Alternatively, you could scale to +3 or +6 VU. This
will take the signal to the upper limit of the tape flux-density and
give better use of the available dynamic range. What's important here
is not so much the absolute value/level chosen, but consistency. Do the
same thing every time.

The basis for what I've said above (and explain VU) follows.

For the present, sticking to just "analog" speech:

To answer 2, we have to explain a bit in 1. Fundamentally, the speech
signal can be described as a "quasi-stationary" signal. Over very short
time intervals (20 ms typical), a given "snapshot" of speech has a
well-defined amplitude and phase structure that you can shove through an
FFT and get an (approximation) of the spectral energy. However, every
other 20ms segment is observably different. If you "time average" the
spectral energies of many 20 ms segments you get the well known (long
term) formant structure of speech.

The non-stationary nature of speech thus causes problems when attempting
to define and measure signal power. rms and peak measures cannot be
applied in a meaningful way. To try and get around this problem the
"volume unit" (VU) was invented. This unit of measurement is related to the
LOUDNESS of the signal and relies on *both* an electrical reference level
relative to an rms value of a sinewave and the dynamic characteristics of
the (analog meter) instrument and the time constants (ballistics) of the
meter movement. Read:

CHINN HA, GANNETT DK, MORRIS RM : A New Standard Volume Indicator and
Reference Level, Bell System Tech. Journal, Vol 19 No 1, 1940,
pp 94-137 (January).

Life was fine until LED's and fast opamps came along which can provide
real-time peak responding meters.  Practical implementations and circuits
have time constants built into them  so that they represent the signal
power in some manner, but are capable of responding much faster than an
analog meter movement, so they can display "instantaneous peaks" more
accurately. However, for *speech*, instantaneous peaks are irrelevant
because they occur infrequently. This is related to the signal amplitude
probability distribution. In 1962 a researcher at British Telecom in
discussing a choice of PCM companding law indicated that a total dynamic
range of 62 dB was theoretically required (for telephone speech), but that
a dynamic range of 38 dB was ok for 99.95% of all conversations!

[ PURTON RF: A survey of Telephone Speech-Signal Statistics and their
  Significance in the choice of a PCM Companding Law., Proc. IEE, Vol
  109B 1962 pp 60-66 (the English IEE, *not* the IEEE)  ]

All of the above comments go out the window when we talk about (pure)
*music* signals because they are totaly different. However, audio and
video recorders find their main application in recording mixed-mode
signals.

It has been my experience (with analog audio recorders) that
the (VU) calibrations are based on an electrical rms sinewave reference
level that causes a recorded magnetic flux density of 250 nWb/m2
(nano-Webers per square-meter). In fact there are EIA (US) and DIN
(Europe) standards that define such things. In practice, I have found
for better quality tape decks, calibration methods specify 0dB sinewave
at 1KHz (or 800 Hz) = 0VU (and corresponds to the  250 nWb/m2) with the
front panel recording level control set flat out. The same sort of
approach must apply to your video recorder, otherwise as you observe, you
run straight into tape saturation problems.

The next trick is to define "0 dB". This I have found can be one of two
things: a) it represents an input rms sinewave (1KHz) analog level that
gives a prescribed test VOLTAGE level at some point in the recording/playback
chain, or b) it represents 0 dBm analog input. ( 0 dBm = 1 mw across 600
ohms).

============================ Richard Stephen ===============================
|   Technology Strategy             |      email: stephen@corp.telecom.co.nz
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