Equalisation for PC mic input/line input

QUESTION:
Do the PC mic and line inputs use the same equalisation (on the
frequency spectrum)?


BACKGROUND:
I have some noisy voice tapes from an old analogue dictation machine.
I'm reading the recordings from the "ear" socket of my dictation
machine into my PC and then coverting the recordings to MP3.

Later I will get some software to clean up the noise on the MP3s.

I didn't expect it but my PC allows me to set a decent recording
level whether I record through the mic input or the line input.

Until I clean up the sound, the noise from the original recording
makes it hard to tell if I'm getting a better result from the mic
input or the line input.

I was wondering if there was a different equalisation used by the PC
for the mic and line inputs. If so then I would make sure I used


ANOTHER QUESTION:
What is the input level at which the mic and line inputs are rated?
I had thought mic inputs were about 3 or 3 mV and line inputs were
200 mV.

-------

NOTE:
My PC motherboard chipset is VIA KT266A + VT8235.
A PC reporting utility says it detects a VT8233/A AC97 Enhanced Audio
Controller.

 

There is not equalization with line input, i.e. it uses flat frequency response. I am not sure about microphone input; the biggest difference seems to be they operate at a lower signal level.

You should use the line input with the headphone output, as both operate at relatively high signal level with flat frequency response.

Back in the days of analog recording 0 db was 1 volt. Since this was analog the 0 db level could be and was often exceed. With the advent of digital CDs the 0 db level became the maximum level, which can not be exceeded because of the digital format, and is supposed to be 2 volts. Since these are maximum levels the average will much lower.
Microphone levels are lower and vary widely.

 

Microphone inputs are also flat, i.e. with no equalisation. As you say, they
operate at a much lower level, typically a few millivolts.

Not quite. Firstly, a dB is a relative level, not an absolute, so without
stating the reference, a figure of "xdB" is meaningless. Originally, 0dB was
referenced to a power of 1mW into a load of 600 ohms, and was referred to as
0dBm. Later, the same voltage level, but unloaded, that is, without
reference to a 600 ohm load became 0dBu (that is, unloaded) Note that the
voltage level is the same in both cases (0.775v, or 1mW into 600 ohm) There
was a strange semi-standard evolved of referring to 1V rather than 0.775v
and that was 0dBv.

Digital outputs are referred to maximum digital output (when all the bits
are 1) and that is called 0dBFS (0dB Full Scale). It has NO analogue
equivalent, as analogue can keep getting bigger without limit, digital can't
get any bigger than when all the bits are 1. In Digital-Analogue conversion,
a number of different conversion levels have become more-or-less standard.
The EBU (European Broadcasting Union) have defined 0dBFS digital to mean
+18dBu analogue after conversion. The USA prefers that 0dBFS = +24dBu
because that provides 20dB headroom above 0VU. A few dissidents prefer
+25dBu as that's 1dB better than +24...........

CD players have evolved a standard output of 2v analogue for 0dBFS, but as
far as I'm aware, there is no official standard for this.

S.

 

Interesting.
Can you let me have a reference to the technical paper for that.
Cheers,

 

EBU R68-2000. I'm emailing you a copy directly.

S.

 

Received, thanks.

With 0dBFS as =18dBU, that would mean OVU (+4dBU) would be -14dBFS. Most
people I know line-up their Pro Tools rigs for -18dBFS for 0VU hence
OdBFS is going to be +22dBU.
Isn't it a lovely world!

 

Standard are great, that's why we have so many of them!

S.

 

The short answer is that for best results you should:

- use a quality tape deck with line-out, not the earphone
jack of a dictation system

- use the line-in on a fair quality sound card, not
integrated motherboard audio with a really cheap codec.

 

Decibels, to me, seem to be overused, particularly with digital audio. And
particularly because silence is negative infinity decibels, which isn't a
lot of good if you're writing a computer program that can only cope with
real numbers. What's wrong with plain old 0% to 100%?!
(Rhetorical question, don't feel obliged to answer...)

 

I can't say I understand all of what you write but the parts I do
understand are very useful to me. Thank you for posting.

Are there any web sites or documents which explain this sort of thing
for a beginner: rigorously but not going too fast.

 

Lots snipped>

I don't know of any specific websites that explain all the ins and outs of
analogue and digital audio. I've learned all this during my professional
life in audio. There used to be a great magazine called Studio Sound, which
had technical articles explaining the basics in rigorous but understandable
form. Sadly SS has been extinct for several years, but you may find copies
in larger public libraries.

You may also want to look at Jim Lesurf's web sites -
Electronics http://www.st-and.ac.uk/~www_pa/Scots_Guide/intro/electron.htm
Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html

He's got a lot of useful information, and what's more, it's correct!

Canford Audio have useful information in their catalogue, you may want to
contact them and see if they'll put you on the mailing list for the
catalogue. www.canford.co.uk

S.

 

Are you trying to put us out of a job?

 

Not much danger of that, if he's such a bad programmer!

BTW, what's wrong with 0-100% is that our hearing is logarithmic, so
deciBels give a much better idea of how things sound. A 10dB increase
in SPL sounds twice as loud, but takes ten times the power.

 

It's quite interesting to look at levels off Freeview. I lined up the
workshop receiver to read PPM 4 on a rare occasion when there was a test
card and line up tone available. And as expected TV progs peak to no more
than PPM 6. But some of the radio ones wrap the PPMs round the end stops.
;-)

My best Freeview receiver out of several is a Sony VTX-D800U and when I
changed it from the previous freebie Sagem which kept crashing I
immediately noticed the audio level was low. Switch the set from the same
channel on analogue to Freeview via a SCART and the difference was too
much. Correspondence with Sony showed that they thought the TV
broadcasters would peak to 0dBFS on FreeView instead of using the normal
EBU line up of peak being -10 dBFS.

 

dbm is as stated a reference of two like power values to a 1mW reference
however the impedence does not need to be 600 ohms, it can be any value of
ohms as long as both power values are based on the same impedence

dbu is not unloaded but db(micro) it is as above but with a reference of
1microWatt it is not actually a u but the greek character mu

dbv would be a ratio based on two voltage levels and a reference of 1 volt

power db caclulations are 10 log Pout/Pin
Voltage db calculations are 20 log Vout/Vin

dbFS is "decibels full scale". It is an abbreviation for decibel amplitude
levels in digital systems which have a maximum available level (like PCM
encoding). 0 dBFS is assigned to the maximum possible level. There is still
the potential for ambiguity, since some use the RMS value of a full-scale
square wave for 0 dBFS, and some use a sine wave.

this is treated the same as voltag calculations because it is based on the
signal to noise ratio.

 

Don,

Do you have a reference for this statement? In 34 years in Pro-Audio I have
never heard it expressed in this way, always referred to 1mW into 600 ohms.

I have no diea what the above is referring to: dBFS has nothing to do with
signal-to-noise, it is just how many dB below full-scale.

S.

 

Since there is no simple way to measure power independently of
knowledge of the circuit impedance (nor any real need to do so in most
cases), the measuring device is simply an ac voltmeter capable of
accurately measing ac volts (or milli or micro volts) at the frequencies
concerned and such (relative) 'power' calibration in dbs on a voltage
scale need to at least mention a reference impedance if absolute power
levels are of any importance or at the very least what the actual
voltage is for the chosen 0db level so that a correction factor can be
applied when used with impedances (notional or actual) other than what
the meter was origionally calibrated for.

The lower the impedance, the lower the voltage reference for 1mW 0db
level. The correction factor for a meter calibrated in db for a 600 ohm
impedance would be plus 9db when used on a 75 ohm circuit if absolute
power levels need to be known.

If nothing else, knowing the reference impedance to which a level
measuring meter is calibrated will inform you of the 1mW voltage
calibration point. Of course, such meters can simply be referenced to a
voltage, usually V or mV or Micro V, without the word 'impedance' being
invoked. Either way, it _is_ important that you can identify the actual
voltage reference point, either directly or indirectly implied by the
intended circuit impedance it's calibrated for.

 

Thanks to Don and Glenn. I've learnt something. Looking through my old
college texts books, I can see you're right. What comes of a narrow
upbringing.

S.

 

This puzzled me.

The first quote (from don, not Don) is the opening part of the DBFS
entry in Wikipedia - see http://en.wikipedia.org/wiki/DBFS. I think it
is correct at least up to the final sentence about ambiguity. Then it
becomes at least ambiguous itself.

The actual ambiguity seem to be whether, when a waveform is said to
have amplitude x dBFS, you mean the peak amplitude of the waveform or
its RMS amplitude. Thus I think the fundamental ambiguity is not as
stated in the Wikipedia article about whether you use a sine or square
wave as reference.

Like Don (not don) I always assumed with dBFS you implicitly meant the
peak value of the waveform because of the nature of its representation
in a system having a waveform-independent overload level of 0 dBFS.

I had to think about this a bit when doing some FFTs (which usually work
in power/energy terms) on quantized signals. Maybe some people are more
comfortable to think of waveforms in power or energy terms however they
are represented, even when power or energy is probably no longer relevant.

 

The wave-shape doesn't matter when talking about digital signals. 0dBFS is
reached when any part of the waveform sets "all the bits to 1"
This can be the crest of a sine-wave, the tip of a sawtooth or the flat top
of a square-wave. If you have a meter that indicates dBFS, with a true-peak
characteristic, you will get the same indication whatever the waveform.
However, if you have a conventional rms reading analogue meter, driven from
a D-A converter, then the waveform will affect the indication, just as it
will for analogue waveforms that *all have the same peak value* The
commonly-used EBU standard of +18dBu=0dBFS is only valid for sine waves.

As an aside, in radio, digital metering is still done on conventional BBC
style PPMs, which under-read by anything between 1-4dB depending on the
programme content.(some will say even up to 7dB) I and others have tried
persuading radio stations to use a true-peak meter, even if it is calibrated
with the familiar BBC 1-7 scale. The universal reaction was that the signal
was too quiet, and everyone prefered to go back to a meter they were
familiar with, even if it didn't tell the truth, and rely on the 10dB
headroom between the +8dBu UK peak operating level and the +18dBu maximum to
accomodate any unseen peaks. US practice is even less precise as they still
use VU meters and rely on the 20dB headroom between 0VU (+4dBu) and their
+24dBu=0dBFS.

S.