15 Aug
2006
15 Aug
'06
11:32 a.m.
Hello all,
Does anybody know, how i can scale the incoming jack signals to dbSPL,
which is in the range of 0 to 120. An is it possible to calculate from dbFS
(which is used in normal soundapp in range -inf to 12db) into dbSPL.
thanks c~
15 Aug
15 Aug
11:59 a.m.
Le Tue, 15 Aug 2006 15:30:47 +0200,
conrad berhörster <conrad.berhoerster(a)gmx.de> a écrit :
Hello all,
Does anybody know, how i can scale the incoming jack signals to dbSPL,
which is in the range of 0 to 120. An is it possible to calculate from dbFS
(which is used in normal soundapp in range -inf to 12db) into dbSPL.
thanks c~
Look here:
http://en.wikipedia.org/wiki/Sound_pressure
http://en.wikipedia.org/wiki/DBFS
http://en.wikipedia.org/wiki/Decibel
The dbFS 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.
The dbSPL correspond to the physical sound pressure level you have in a
given area. It can be in a range of whatever you want or can archive.
It will depend on many factors and you have to measure it. And more, it
will be true only at the location you have done the measurement.
It you want to archive a constant sound level in a great area, you have
to use multiple speakers and amplifiers, and setup the volume level in
each amplifier with measurement of the sound level in different points
of the area.
Cheers,
Dominique
12:09 p.m.
On Tue, Aug 15, 2006 at 03:30:47 +0200, conrad berhörster wrote:
Hello all,
Does anybody know, how i can scale the incoming jack signals to dbSPL,
which is in the range of 0 to 120. An is it possible to calculate from dbFS
(which is used in normal soundapp in range -inf to 12db) into dbSPL.
dB SPL is a measurement of physical sound energy, so in general software
was no control over it. It depends on the efficientcy of your monitors,
gain of your amps, desk etc.
The only way you can relate dB SPL to dB FS is by calibrating your
particular setup using a sound level meter. There's a SMPTE standard for
the relationship between dB SPL and dBu or dBv (I forget which), which
gets you half way there, but noone uses it anyway. I had my home system
calibrated to SMPTE for a while, and it was just anoying.
Bob Katz advocates the "K-system" calibration, which is different again,
and also not very common.
http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_pag…
- Steve
1:29 p.m.
Whoow , that was fast....
maybe i was a little bit misunderstandable.
when i wrote about jack, i meant the jackserver. i write an app, which has an
microphon input. now i will take this microphon signal and show it as a gain
meter. this is no magic and everything works.
if i understand all of you right, the pressure on the microphon is something
greater 0. and normally it is lower 2, right?
first question: is this the same pressure which is measured in the literature
in pascal (pa)
if so, its easy to calculate
out_value = 20.00*(Math.log(in_value/0.00002)/Math.log(10.00));
I know, i need to calibrate the room at the end. But in the moment, my
question is simply to get a signal and show the loundness in dbSPL.
Can i use the formula above or not?
thanks c~
Am Dienstag 15 August 2006 16:09 schrieb Steve Harris:
On Tue, Aug 15, 2006 at 03:30:47 +0200, conrad
berhörster wrote:
Hello all,
Does anybody know, how i can scale the incoming jack signals to dbSPL,
which is in the range of 0 to 120. An is it possible to calculate from
dbFS (which is used in normal soundapp in range -inf to 12db) into
dbSPL.
dB SPL is a measurement of physical sound energy, so in general software
was no control over it. It depends on the efficientcy of your monitors,
gain of your amps, desk etc.
The only way you can relate dB SPL to dB FS is by calibrating your
particular setup using a sound level meter. There's a SMPTE standard for
the relationship between dB SPL and dBu or dBv (I forget which), which
gets you half way there, but noone uses it anyway. I had my home system
calibrated to SMPTE for a while, and it was just anoying.
1:51 p.m.
--- conrad berhörster <conrad.berhoerster(a)gmx.de>
wrote:
Whoow , that was fast....
maybe i was a little bit misunderstandable.
when i wrote about jack, i meant the jackserver. i
write an app, which has an
microphon input. now i will take this microphon
signal and show it as a gain
meter. this is no magic and everything works.
if i understand all of you right, the pressure on
the microphon is something
greater 0. and normally it is lower 2, right?
first question: is this the same pressure which is
measured in the literature
in pascal (pa)
if so, its easy to calculate
out_value =
20.00*(Math.log(in_value/0.00002)/Math.log(10.00));
Its not a case of calibrating the room, its basically
a case of needing to know what sample value 20uPa
equates to.
This is dependent on the microphone, preamp and sound
card in use, or for the output case, the soundcard,
power amp gain and speaker sensitivity (plus room
effects).
You need to ensure that 120dbSPL at the transducer
does not clip either hhe mic, preamp or soundcard and
then find out what sample value this gives with your
hardware. Calibrate that point as 120dbSPL, then as
long as all your hardware is linear the rest of the
thing will just work.
Regards, Dan.
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2:12 p.m.
On Tue, 2006-08-15 at 16:51 +0100, Dan Mills wrote:
This is dependent on the microphone, preamp and sound
card in use, or for the output case, the soundcard,
power amp gain and speaker sensitivity (plus room
effects).
You need to ensure that 120dbSPL at the transducer
does not clip either hhe mic, preamp or soundcard and
then find out what sample value this gives with your
hardware. Calibrate that point as 120dbSPL, then as
long as all your hardware is linear the rest of the
thing will just work.
FWIW, there is some work going on in ALSA to make the drivers use dB
units for the mixer, rather than arbitrary scales. But of course this
will only work for devices for which the developers have the full
hardware specs.
Lee
2:22 p.m.
Le Tue, 15 Aug 2006 12:01:11 -0400,
Lee Revell <rlrevell(a)joe-job.com> a écrit :
On Tue, 2006-08-15 at 16:51 +0100, Dan Mills wrote:
It is a good news.
With jack, mixers as jackeq or jackmix are already using db scales.
Dominique
This is dependent on the microphone, preamp and
sound
card in use, or for the output case, the soundcard,
power amp gain and speaker sensitivity (plus room
effects).
You need to ensure that 120dbSPL at the transducer
does not clip either hhe mic, preamp or soundcard and
then find out what sample value this gives with your
hardware. Calibrate that point as 120dbSPL, then as
long as all your hardware is linear the rest of the
thing will just work.
FWIW, there is some work going on in ALSA to make the drivers use dB
units for the mixer, rather than arbitrary scales. But of course this
will only work for devices for which the developers have the full
hardware specs.
Lee
3:03 p.m.
--- Dominique Michel <dominique.michel(a)citycable.ch>
wrote:
With jack, mixers as jackeq or jackmix are already
using db scales.
Yeeees, but they deal with gain which is unitless in
the DB system.
What is being discussed is how to reference things to
external signal levels (measured in volts, SPL or
whatever), this requires a mapping from dbFS to the
external signal level and that is heavily hardware
dependent.
Regards, Dan.
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3:37 p.m.
Le Tue, 15 Aug 2006 18:03:17 +0100 (BST),
Dan Mills <dmills_00(a)yahoo.co.uk> a écrit :
--- Dominique Michel <dominique.michel(a)citycable.ch>
wrote:
The db system always use a reference and a db of any kind have no unit.
dbFS use as reference the maximum possible digital signal and the other
term will be too a possible digital signal. dbSPL use as reference a
given sound pressure, but the other term will be too a sound pressure,
so the result in both cases is unitless.
sound_pressure/sound_pressure give the rate between the 2
sound_pressure, and a rate have no unit, and that even after the
multiplication by 10 or 20 log base 10. Only the knowledge of the
reference give you the knowledge of the unit (as well as the math).
The db inside jack are dbFS with a maximum possible signal of 0 db.
Now, both jacqeq and jackmix give you a maximum conrol level at +6dB.
It mean at +6dB in those EQ is equal to 0dbFS in jack.
With jack, mixers as jackeq or jackmix are
already
using db scales.
Yeeees, but they deal with gain which is unitless in
the DB system.
What is being discussed is how to reference things to
external signal levels (measured in volts, SPL or
whatever), this requires a mapping from dbFS to the
external signal level and that is heavily hardware
dependent.
Regards, Dan.
I know that. I said in a precedent message at he must measure the sound
level in a given point of the local to know the relation between the
dbSPL at that point and the dbFS in the box.
Best,
Dominique
4:03 p.m.
The db inside jack are dbFS with a maximum possible
signal of 0 db.
Now, both jacqeq and jackmix give you a maximum conrol level at +6dB.
It mean at +6dB in those EQ is equal to 0dbFS in jack.
there are no dB units inside of JACK. some JACK applications use dBFS,
that much is true. however, it is not true that because the allow a gain
level of +6dB (or any value for that matter) that +6dB in that app
equals 0dBFS. this is because JACK's floating point sample format allows
values above 0dBFS, even though when such a value is passed to DAC, it
will be clipped and caused distortion or worse.
within JACK, 0dBFS corresponds to a sample value of +/1.0, and will
emerge from the other side of your DAC with the maximum voltage level
that the DAC can emit. values above 0 dBFS are legal within JACK, but
should probably never be routed to a physical output port.
--p
4:43 p.m.
Le Tue, 15 Aug 2006 14:03:34 -0400,
Paul Davis <paul(a)linuxaudiosystems.com> a écrit :
Thank you for this explanation.
Dominique
The db inside
jack are dbFS with a maximum possible signal of 0 db.
Now, both jacqeq and jackmix give you a maximum conrol level at +6dB.
It mean at +6dB in those EQ is equal to 0dbFS in jack.
there are no dB units inside of JACK. some JACK applications use dBFS,
that much is true. however, it is not true that because the allow a gain
level of +6dB (or any value for that matter) that +6dB in that app
equals 0dBFS. this is because JACK's floating point sample format allows
values above 0dBFS, even though when such a value is passed to DAC, it
will be clipped and caused distortion or worse.
within JACK, 0dBFS corresponds to a sample value of +/1.0, and will
emerge from the other side of your DAC with the maximum voltage level
that the DAC can emit. values above 0 dBFS are legal within JACK, but
should probably never be routed to a physical output port.
--p
9:36 p.m.
Dominique Michel wrote:
Le Tue, 15 Aug 2006 12:01:11 -0400,
Lee Revell <rlrevell(a)joe-job.com> a écrit :
ALSA is starting to support "dB Gain" indications for mixer controls. It
currently supports Creative Audigy cards and USB sound cards. Support
for other sound cards will follow in time.
So, at least people will know where the 0dB gain point is for the mic
and line in.
James
On Tue, 2006-08-15 at 16:51 +0100, Dan Mills
wrote:
It is a good news.
With jack, mixers as jackeq or jackmix are already using db scales.
Dominique This is dependent on the microphone, preamp and
sound
card in use, or for the output case, the soundcard,
power amp gain and speaker sensitivity (plus room
effects).
You need to ensure that 120dbSPL at the transducer
does not clip either hhe mic, preamp or soundcard and
then find out what sample value this gives with your
hardware. Calibrate that point as 120dbSPL, then as
long as all your hardware is linear the rest of the
thing will just work.
FWIW, there is some work going on in ALSA to make the
drivers use dB
units for the mixer, rather than arbitrary scales. But of course this
will only work for devices for which the developers have the full
hardware specs.
Lee
12:25 p.m.
--- conrad berhörster <conrad.berhoerster(a)gmx.de>
wrote:
Hello all,
Does anybody know, how i can scale the incoming jack
signals to dbSPL,
which is in the range of 0 to 120. An is it possible
to calculate from dbFS
(which is used in normal soundapp in range -inf to
12db) into dbSPL.
thanks c~
Look up what voltage corresponds to 0dbFS, then given
a known sensitivity figure for the microphone and a
known gain for the preamp do the math.
For example if 0dbFS is specified as corresponding to
+22dbu (actually a voltage reference BTW), and you
establish that 100dbSPL gives an output of say -50dbu
from the mic, then if your preamp has a gain of say
50db, the math looks like this:
100dbSPL = -50dbu at mic, so 0dbSPL = -150dbu at mic
(burried in the nose floor in all probability), the
preamp has 50db gain so the output will be -100dbu at
0dbSPL.
At 120dbSPL (assuming a well behaved mic and preamp),
you will have +20dbu, now the Full sacle value is
specified as +22dbu, so a 20dbu input will read -2dbFS
, therefore in this example SPL = 122 + DBFS (always
negative).
This calibration obviously needs to be made for each
microphone/preamp/card combination if you want decent
accuracy.
In practice, I would try to set up for the top of my
measurement range to be about 10db or so below 0dbFS
as you typically want RMS measurements and 0dbFS is a
peak value and a very unforgiving place.
Some of the better card manufacturers specify things
like 0dbu output for -20dbFS which is obviously saying
that 0DBFS = +20dbu
Regards, Dan.
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12 comments
7 participants
participants (7)
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conrad berhörster -
Dan Mills -
Dominique Michel -
James Courtier-Dutton -
Lee Revell -
Paul Davis -
Steve Harris