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.

Equalisation for PC mic input/line input
 

"Mike Walsh" wrote in message
...


"David Peters (UK)" wrote:

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

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.

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


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

Correct.

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.

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.

Equalisation for PC mic input/line input
 

In message , Serge Auckland
writes

(snip)

The EBU (European Broadcasting Union) have defined 0dBFS digital to mean
+18dBu analogue after conversion.


Interesting.
Can you let me have a reference to the technical paper for that.
Cheers,
--
Kevin Seal (at home)
FZS600 in Banana
{kevin at the hyphen seal hyphen house dot freeserve dot co dot uk}

Equalisation for PC mic input/line input
 

"Kevin Seal" wrote in message
...
In message , Serge Auckland
writes

(snip)

The EBU (European Broadcasting Union) have defined 0dBFS digital to mean
+18dBu analogue after conversion.


Interesting.
Can you let me have a reference to the technical paper for that.
Cheers,
--
Kevin Seal (at home)
FZS600 in Banana
{kevin at the hyphen seal hyphen house dot freeserve dot co dot uk}


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

S.

Equalisation for PC mic input/line input
 

In message , Serge Auckland
writes

"Kevin Seal" wrote in message
...
In message , Serge Auckland
writes

(snip)

The EBU (European Broadcasting Union) have defined 0dBFS digital to mean
+18dBu analogue after conversion.


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.

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!
--
Kevin Seal (at home)
FZS600 in Banana
{kevin at the hyphen seal hyphen house dot freeserve dot co dot uk}

Equalisation for PC mic input/line input
 

"Kevin Seal" wrote in message
...
In message , Serge Auckland
writes

"Kevin Seal" wrote in message
...
In message , Serge Auckland
writes

(snip)

The EBU (European Broadcasting Union) have defined 0dBFS digital to mean
+18dBu analogue after conversion.


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.

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!
--
Kevin Seal (at home)
FZS600 in Banana
{kevin at the hyphen seal hyphen house dot freeserve dot co dot uk}

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

S.

Equalisation for PC mic input/line input
 

On Sun, 12 Mar 2006 13:41:15 GMT, "David Peters (UK)"
wrote:

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.


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.

Equalisation for PC mic input/line input
 

On 12 Mar 2006, Serge Auckland wrote:


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.

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.


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.

Equalisation for PC mic input/line input
 

"David Peters (UK)" wrote in message
...
Lots snipped

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.

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/Scot...o/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.

Equalisation for PC mic input/line input
 

In message , Rich Wilson
writes

"Serge Auckland" wrote in message
...

"Kevin Seal" wrote in message
...
In message , Serge Auckland
writes

"Kevin Seal" wrote in message
...
In message , Serge Auckland
writes

(snip)

The EBU (European Broadcasting Union) have defined 0dBFS digital to
mean
+18dBu analogue after conversion.


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.

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!

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...)


Are you trying to put us out of a job? :)
--
Kevin Seal (at home)
FZS600 in Banana
{kevin at the hyphen seal hyphen house dot freeserve dot co dot uk}

Equalisation for PC mic input/line input
 

On Mon, 13 Mar 2006 18:37:33 +0000, Kevin Seal wrote:

In message , Rich Wilson
writes

"Serge Auckland" wrote in message
...

"Kevin Seal" wrote in message
...
In message , Serge Auckland
writes

"Kevin Seal" wrote in message
.. .
In message , Serge Auckland
writes

(snip)

The EBU (European Broadcasting Union) have defined 0dBFS digital to
mean
+18dBu analogue after conversion.


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.

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!

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...)

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.

--

Stewart Pinkerton | Music is Art - Audio is Engineering

Equalisation for PC mic input/line input
 

In article ,
Serge Auckland wrote:
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...........

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.

--
*Re-elect nobody

Dave Plowman London SW
To e-mail, change noise into sound.

Equalisation for PC mic input/line input
 

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.

"David Peters (UK)" wrote in message
...
On 12 Mar 2006, Serge Auckland wrote:


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.

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.


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.

Equalisation for PC mic input/line input
 

Crossposting removed.

Hi,

"don" wrote in message
...
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

No, in this context it's dBu (note the capital B, for Bell). It's equally
valid to use a reference of 1 microWatt, but that's not what is used
commonly in professional audio. It might appear, for example, as
dB(?V/m) for electric field strength, relative to 1 microvolt per
metre. Not sure if the mu will come through in ASCII - apologies
if it doesn't.

If the load is 600 ohms, then dBu=dBm.

Regards,

Glenn.

Equalisation for PC mic input/line input
 

"Glenn Booth" wrote in message
...
Crossposting removed.

Hi,

"don" wrote in message
...
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

No, in this context it's dBu (note the capital B, for Bell). It's equally
valid to use a reference of 1 microWatt, but that's not what is used
commonly in professional audio. It might appear, for example, as
dB(?V/m) for electric field strength, relative to 1 microvolt per
metre. Not sure if the mu will come through in ASCII - apologies
if it doesn't.

If the load is 600 ohms, then dBu=dBm.

Regards,

Glenn.

Glenn is correct. dBu(micro)V is generally used for field strength
measurements and is referred to 1 microvolt/m.

dBu and dBm refer to the same *voltage* level, but different power levels.
dBm refers for 1mW into 600 ohms, dBu is the same voltage level (0.775v rms)
but without reference to a load. I have never heard of dBm being referred
to anything other than 1mW into 600 ohms, nor dBu being referred to 1 micro
watt. The point about dBu is that it is a *voltage* level reference, not a
power reference.

S.

Equalisation for PC mic input/line input
 

On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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


No, the impedance does not need to be the same, and there are not two
power values, but one - specified as dB with respect to one milliwatt.
Impedance does not appear anywhere in this figure.

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


dBu is indeed dB (unloaded). It is a relic of 600 ohm line audio
systems and is the voltage that would have produced 0dBm in 600 ohms,
but since we now run into high impedances instead, must be specified
otherwise - hence dBu.

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

dBV, actually, not dBv.

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.

No, no ambiguity, dB below full scale does not depend on wave shape,
merely how many digital levels remain unused.

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

It has nothing to do with signal to noise ratio - it is all happening
at the other end of the scale.

d

Pearce Consulting
http://www.pearce.uk.com

Equalisation for PC mic input/line input
 

"Don Pearce" wrote in message
...
On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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


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.



dBu is indeed dB (unloaded). It is a relic of 600 ohm line audio
systems and is the voltage that would have produced 0dBm in 600 ohms,
but since we now run into high impedances instead, must be specified
otherwise - hence dBu.

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

dBV, actually, not dBv.


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

It has nothing to do with signal to noise ratio - it is all happening
at the other end of the scale.

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.

Equalisation for PC mic input/line input
 

On Sat, 18 Mar 2006 19:56:17 -0000, "Serge Auckland"
wrote:


"Don Pearce" wrote in message
...
On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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


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.


No, it is just one milliwatt - no ohms needed. You have only come
across it in relation to 600 ohms because you have been worked in
audio, and that is all you have been exposed to. If you ever worked in
RF, you would have found exactly the same power, referred to in
exactly the same way in 50, 62.5 and 74 ohms. The power is the same in
all of these - and 600 ohms too.


dBu is indeed dB (unloaded). It is a relic of 600 ohm line audio
systems and is the voltage that would have produced 0dBm in 600 ohms,
but since we now run into high impedances instead, must be specified
otherwise - hence dBu.

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

dBV, actually, not dBv.


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

It has nothing to do with signal to noise ratio - it is all happening
at the other end of the scale.

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.


Exactly.

d

Pearce Consulting
http://www.pearce.uk.com

Equalisation for PC mic input/line input
 

Hi,

"Don Pearce" wrote in message
...
On Sat, 18 Mar 2006 19:56:17 -0000, "Serge Auckland"
wrote:


"Don Pearce" wrote in message
.. .
On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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


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.


No, it is just one milliwatt - no ohms needed. You have only come
across it in relation to 600 ohms because you have been worked in
audio, and that is all you have been exposed to. If you ever worked in
RF, you would have found exactly the same power, referred to in
exactly the same way in 50, 62.5 and 74 ohms. The power is the same in
all of these - and 600 ohms too.

Agreed. We had the same conversation on uk.r.a back in 2003. I've
quoted a bit of it here as Serge might be interested:

I said:
I don't disagree that the reference must always be given, but for
measurements of power, such as those that reference dBm (dB referenced
to 1mW) surely the impedance is totally redundant? (Unless what one is
really trying to describe is voltage, but using a power ratio to do
so). The 50R says nothing that I can see about the power, it only
allows one to relate the voltage that will be dropped across that
particular impedance/resistance with that dB worth of power being
dissipated.

To which a certain Mr. Lesurf said this:

However, bear in mind two points:

1) That in most cases (in RF at least) the quoted systems will be based
upon assuming the system is impedance matched and then give the power that
will be delivered to the source. Hence quoting the impedance tells the user
that this is the required matched impedance for optimum power transfer.


2) That in most cases the receiver will tend to be designed to work over a
given voltage range due to finite voltage rails, etc. Hence the impedance
is useful for establishing the voltage levels that must be expected.


It is therefore useful to confirm the assumed impedances. In RF/microwave
we have the annoyance that 50 Ohm is common for system and lab work, but
other impedances like 75 Ohm, etc, crop up for specific purposes/areas.


In principle, though: yes, once you've quoted the signal power in dBm
you've established the power available. You could then use a transformer to
alter the impedance (and hence signal voltage) if so desired.


At the time, I hadn't considered case (1) that Jim pointed out, as I know

less than nothing about RF.

Who says Usenet doesn't go around in circles?

Regards,

Glenn.

Equalisation for PC mic input/line input
 

"Glenn Booth" wrote in message
...
Hi,

"Don Pearce" wrote in message
...
On Sat, 18 Mar 2006 19:56:17 -0000, "Serge Auckland"
wrote:


"Don Pearce" wrote in message
. ..
On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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


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.


No, it is just one milliwatt - no ohms needed. You have only come
across it in relation to 600 ohms because you have been worked in
audio, and that is all you have been exposed to. If you ever worked in
RF, you would have found exactly the same power, referred to in
exactly the same way in 50, 62.5 and 74 ohms. The power is the same in
all of these - and 600 ohms too.

Agreed. We had the same conversation on uk.r.a back in 2003. I've
quoted a bit of it here as Serge might be interested:

I said:
I don't disagree that the reference must always be given, but for
measurements of power, such as those that reference dBm (dB referenced
to 1mW) surely the impedance is totally redundant? (Unless what one is
really trying to describe is voltage, but using a power ratio to do
so). The 50R says nothing that I can see about the power, it only
allows one to relate the voltage that will be dropped across that
particular impedance/resistance with that dB worth of power being
dissipated.

To which a certain Mr. Lesurf said this:

However, bear in mind two points:

1) That in most cases (in RF at least) the quoted systems will be based
upon assuming the system is impedance matched and then give the power that
will be delivered to the source. Hence quoting the impedance tells the
user
that this is the required matched impedance for optimum power transfer.


2) That in most cases the receiver will tend to be designed to work over a
given voltage range due to finite voltage rails, etc. Hence the impedance
is useful for establishing the voltage levels that must be expected.


It is therefore useful to confirm the assumed impedances. In RF/microwave
we have the annoyance that 50 Ohm is common for system and lab work, but
other impedances like 75 Ohm, etc, crop up for specific purposes/areas.


In principle, though: yes, once you've quoted the signal power in dBm
you've established the power available. You could then use a transformer
to
alter the impedance (and hence signal voltage) if so desired.


At the time, I hadn't considered case (1) that Jim pointed out, as I know

less than nothing about RF.

Who says Usenet doesn't go around in circles?

Regards,

Glenn.

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.

Equalisation for PC mic input/line input
 

On 2006-03-18, Don Pearce wrote:
On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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.

No, no ambiguity, dB below full scale does not depend on wave shape,
merely how many digital levels remain unused.

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.

--
John Phillips

Equalisation for PC mic input/line input
 

On 19 Mar 2006 09:37:00 GMT, John Phillips
wrote:

On 2006-03-18, Don Pearce wrote:
On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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.

No, no ambiguity, dB below full scale does not depend on wave shape,
merely how many digital levels remain unused.

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.

Think of it this way:

By how many dB would you need to increase the signal level to hit the
limit of the ADC?

That is how many dB below full scale you are, and it ties in perfectly
with my definition. You don't concern yourself with what shape the
wave is - merely how tall it is. So yes, it is the peak-to-peak
amplitude that determines this, not the RMS. The former can be derived
from the latter for known wave shapes, but not for music.

d

Pearce Consulting
http://www.pearce.uk.com

Equalisation for PC mic input/line input
 

"John Phillips" wrote in message
...
On 2006-03-18, Don Pearce wrote:
On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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.

No, no ambiguity, dB below full scale does not depend on wave shape,
merely how many digital levels remain unused.

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.

--
John Phillips

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.

Equalisation for PC mic input/line input
 

In article ,
Serge Auckland wrote:
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.

The great beauty of the analogue PPM is that it gives a good indication of
perceived loudness as well as the electrical value. It's the Holy Grail to
find something which does this better - but it hasn't happened yet.

--


Dave Plowman London SW
To e-mail, change noise into sound.

Equalisation for PC mic input/line input
 

"Dave Plowman (News)" wrote in message
...
In article ,
Serge Auckland wrote:
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.

The great beauty of the analogue PPM is that it gives a good indication of
perceived loudness as well as the electrical value. It's the Holy Grail to
find something which does this better - but it hasn't happened yet.

--


Dave Plowman London SW
To e-mail, change noise into sound.

It's relatively trivial to make a PPM with an LED analogue scale, arranged
in an arc if that's what's more familiar. The PPM's software can be set for
BBC dynamics, both rise and fall, or true-peak rise and conventional fall,
(or any other dynamics that you may care to think of). When we supplied
digital desks to various radio stations, we started with the PPMs indicating
true-peak rise, but within a week or two, the user always reset them to
mimic conventional mechanical pointer rise and fall. It seems that nobody's
actually interested in what the real levels are, just what it looks like -
as you say, they have a mental map of perceived loudness, and that's more
important than the actual level - after all, isn't 10dB headroom enough to
catch any nasties?

S.

Equalisation for PC mic input/line input
 

In article , John Phillips
wrote:
On 2006-03-18, Don Pearce wrote:

No, no ambiguity, dB below full scale does not depend on wave shape,
merely how many digital levels remain unused.

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.

Alas, this is another one of the areas where it is easy for statements to
be ambiguous. Partly due to the confusions between instantaneous peak
levels versus rms, partly due to unspoken assumptions at times that you are
dealing with a sinewave.

To make things even more confusing wrt terminology I am currently doing
measurements and statistics of how the 'short term' peak level varies with
time with some audio waveforms. Thus I'm using peak levels, and then having
to say what the 'peak' peak level is, and how often a given 'peak' level
occurs... There are times when normal English can become hard to use to
deal with such things. :-)

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html
Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html
Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html

Equalisation for PC mic input/line input
 

In article , Don Pearce
wrote:


Think of it this way:

By how many dB would you need to increase the signal level to hit the
limit of the ADC?

That is how many dB below full scale you are, and it ties in perfectly
with my definition. You don't concern yourself with what shape the wave
is - merely how tall it is. So yes, it is the peak-to-peak amplitude
that determines this, not the RMS. The former can be derived from the
latter for known wave shapes, but not for music.

Also for 'random noise' ... Although all being well, this isn't a
worry in terms of FS clipping. If it is, statisics may be the least
of your concerns. :-)

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html
Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html
Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html

Equalisation for PC mic input/line input
 

In article , Serge Auckland
wrote:


When we supplied digital desks to various radio stations, we started
with the PPMs indicating true-peak rise, but within a week or two, the
user always reset them to mimic conventional mechanical pointer rise and
fall. It seems that nobody's actually interested in what the real levels
are, just what it looks like - as you say, they have a mental map of
perceived loudness, and that's more important than the actual level -
after all, isn't 10dB headroom enough to catch any nasties?

FWIW My impression is that R3 at least are generally well clear of
clipping. For example, from DAB I've not yet seen a single sample that got
to the clipping level, or even within a dB or two of it! However unless
they are clipping earlier in the chain, I guess it must happen
occasionally, simply due to the statistics of the real world, and the Laws
of Murphy. ;-

So I guess the answer to your question is similar to that for, "Will I
survive one pull of the trigger when playing Russian Roulette?"... i.e.
"Probably!" Alas, there is a distinction between trying this once, and
repeating it on a regular basis... 8-]

Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Audio Misc http://www.st-and.demon.co.uk/AudioMisc/index.html
Armstrong Audio http://www.st-and.demon.co.uk/Audio/armstrong.html
Barbirolli Soc. http://www.st-and.demon.co.uk/JBSoc/JBSoc.html

Equalisation for PC mic input/line input
 

On 2006-03-19, Don Pearce wrote:
On 19 Mar 2006 09:37:00 GMT, John Phillips
wrote:

On 2006-03-18, Don Pearce wrote:
On Sat, 18 Mar 2006 17:54:15 GMT, "don" wrote:

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.

No, no ambiguity, dB below full scale does not depend on wave shape,
merely how many digital levels remain unused.

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.

Think of it this way:

By how many dB would you need to increase the signal level to hit the
limit of the ADC?

That is how many dB below full scale you are, and it ties in perfectly
with my definition. You don't concern yourself with what shape the
wave is - merely how tall it is. So yes, it is the peak-to-peak
amplitude that determines this, not the RMS. The former can be derived
from the latter for known wave shapes, but not for music.

Exactly. I think it's the Wikipedia definition of dBFS that's puzzling.
I was wondering about re-writing the first bit to something like:


'''dBFS''' is short for "[[decibel]]s [[full scale]]". It is an
abbreviation for decibel amplitude levels in digital systems which have
a maximum available level (for example [[PCM]] encoding). By convention
0 dBFS is assigned to the maximum available level.

There is a potential for ambiguity when assigning a level on the dBFS
scale to a waveform rather than to a specific amplitude, since some
derive the characteristic level of the waveform from its peak value
while others use its [[RMS]] value. Consider a sine wave and a square
wave both of whose peak amplitudes reach the maximum avaiable level.

* Both have a peak amplitude of 0 dBFS.

* The RMS amplitude of the sine wave is approximately -3 dBFS while
the RMS amplitude of the square wave is 0 dBFS.

It is conventional to use a waveform's peak value when assigning it a
level on the dBFS scale. This is probably the more useful because -x
dBFS then means that only x dB increase can be applied to the waveform's
amplitude before [[clipping]] takes place. This is independent of the
waveform in question.

Note that there is no direct connection between a level on the dBFS
scale and any analogue signal level. If a connection is required then
a calibration level must be specified and the equipment must be set up
to achieve this. For example +18 dBu RMS sine wave = 0 dBFS peak is a
common European broadcasting calibration for analogue/digital signal
interchange. The calibration may be different in Japan and the USA.


--
John Phillips