On Thursday, June 5, 2014 11:14:36 AM UTC-3, mako...@yahoo.com wrote:
> use a speaker and an oscillator and create a single frequency tone that you can measure directly with your SLM.
> 
> 
> 
> Run the data for the single tone through your analysis and see what you get.
> 
> 
> 
> Repeat with two tones
> 
> 
> 
> Repeat with N tones
> 
> 
> 
> Repeat with white noise
> 
> 
> 
> Soon you will see the pattern to point to whatever is the problem.
> 
> 
> 
> Mark

Thanks Mark for replying. See, I've tested something similar. The issue is that by doing what you're telling me I would test mainly the filters and I don't thinks the problem is in the filters, once I have got good results for white noise, pink noise and sine waves at different frequencies. The problem is probably in how I've been handling or acquiring the data.

On Thursday, June 5, 2014 9:14:01 AM UTC-3, Bob Masta wrote:
> On Wed, 4 Jun 2014 05:34:29 -0700 (PDT), Alisson Vieira
> 
> <alissonvieira01@gmail.com> wrote:
> 
> 
> 
> >On Wednesday, June 4, 2014 1:49:47 PM UTC+2, Bob Masta wrote:
> 
> >> On Wed, 4 Jun 2014 00:46:40 -0700 (PDT), Alisson Vieira
> 
> >>=20
> 
> >> <alissonvieira01@gmail.com> wrote:
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >> <snip>
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >> >Thanks Bob. Very nice from you. See. Sorry about this, but when I said t=
> 
> >hat I've used a microphone I have actually used a sound level meter 2238 Me=
> 
> >diator from B&K.In this case what would change? thank you
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >> Now I'm really confused.  From what I read about the 2238,
> 
> >>=20
> 
> >> it already comes with 1/3 octave analysis software.  And
> 
> >>=20
> 
> >> coming from B&K you should certainly be able to trust the
> 
> >>=20
> 
> >> calibration.
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >> So, if you don't use any of your own software, just the B&K
> 
> >>=20
> 
> >> analysis, do you get reasonable results?
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >> If so, then my first guess would be that when you are using
> 
> >>=20
> 
> >> raw data from the B&K to do your own analysis, that data
> 
> >>=20
> 
> >> might not include any calibration.  It might be raw volts or
> 
> >>=20
> 
> >> A/D counts or something.
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >> Best regards,
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >>=20
> 
> >> Bob Masta
> 
> >>=20
> 
> >> =20
> 
> >>=20
> 
> >>               DAQARTA  v7.50
> 
> >>=20
> 
> >>    Data AcQuisition And Real-Time Analysis
> 
> >>=20
> 
> >>               www.daqarta.com
> 
> >>=20
> 
> >> Scope, Spectrum, Spectrogram, Sound Level Meter
> 
> >>=20
> 
> >>  Frequency Counter, Pitch Track, Pitch-to-MIDI=20
> 
> >>=20
> 
> >>    FREE Signal Generator, DaqMusiq generator   =20
> 
> >>=20
> 
> >>           Science with your sound card!
> 
> >
> 
> >Let me try to be clearer. I am measuring the noise of an electronic steerin=
> 
> >g lock device. The noise only appears when I press a button from outside th=
> 
> >e anechoic chamber. When I do this the device will lock or unlock (it will =
> 
> >move a pin upwards or downwards).This process takes about 1 second. So, I h=
> 
> >ave this device 2238 mediator that uses serial filters. Probably, you know =
> 
> >everything about this but I will put here what the manual says about the me=
> 
> >ter just to try and explain my situation.
> 
> >Serial Filters - the sound level meter measures in one band at a time. The =
> 
> >meter will usually be fitted with a single filter circuit, which will be el=
> 
> >ectronically switched to measure the different bands. With most modern mete=
> 
> >rs you can set the filters to scan through from 31.5Hz to 16kHz automatical=
> 
> >ly. The advantage of serial octave band filters is that the meter has only =
> 
> >one filter circuit, keeping the cost and power consumption down.
> 
> >So, I can't use the device's frequency analysis software to get the 1/3 oct=
> 
> >ave bands because I would have to run the device at the precise moment that=
> 
> > it changes the frequency band. In order to solve this problem I've connect=
> 
> >ed the sound level meter electrical outputs to the data logger to get the S=
> 
> >PL variations in real time and then used the program in Matlab that uses pa=
> 
> >rallel filters to obtain the 1/3 Octave Bands. When I connect the sound lev=
> 
> >el meter to the data logger I can obtain two results the voltage variation =
> 
> >and the exactly SPL in dB, which the sound level meter is showing on the di=
> 
> >splay.  My question is: in order to filter this signal should I use the raw =
> 
> >voltage or should I transform these values in db to pressure as I did befor=
> 
> >e?
> 
> 
> 
> If I understand correctly, you are not getting SPL from the
> 
> data logger, but only from the sound level meter; the logger
> 
> is only giving voltage with an unknown calibration factor.
> 
> You can use the SPL reading to calibrate the logger voltage.
> 
> 
> 
> 
> 
> Take the FFT magnitude spectrum of the voltage (not 1/3
> 
> octaves).  Apply the same weighting curve that the SPL meter
> 
> uses.  (You mentioned previously that the SPL reading was
> 
> A-weighted.)  Now take the RMS sum of all the spectral
> 
> lines, or at least those under the curve.  That should be
> 
> equivalent to the metered SPL.  
> 
> 
> 
> You can now find a gain factor to apply to the spectrum data
> 
> to give this same SPL.  You can double-check this by using a
> 
> different weighting curve on the meter, which you likewise
> 
> apply to your FFT spectrum as above, and you should get
> 
> matching results.   You can set the meter to show a single
> 
> 1/3 octave SPL, then verify that your own 1/3 octave scheme
> 
> gives the same SPL when applied to this single octave.
> 
> 
> 
> Best regards,
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Bob Masta
> 
>  
> 
>               DAQARTA  v7.50
> 
>    Data AcQuisition And Real-Time Analysis
> 
>               www.daqarta.com
> 
> Scope, Spectrum, Spectrogram, Sound Level Meter
> 
>  Frequency Counter, Pitch Track, Pitch-to-MIDI 
> 
>    FREE Signal Generator, DaqMusiq generator    
> 
>           Science with your sound card!

Thanks again Bob. I will try this out and see what I get.

use a speaker and an oscillator and create a single frequency tone that you can measure directly with your SLM.

Run the data for the single tone through your analysis and see what you get.

Repeat with two tones

Repeat with N tones

Repeat with white noise

Soon you will see the pattern to point to whatever is the problem.

Mark

On Wed, 4 Jun 2014 05:34:29 -0700 (PDT), Alisson Vieira
<alissonvieira01@gmail.com> wrote:

>On Wednesday, June 4, 2014 1:49:47 PM UTC+2, Bob Masta wrote:
>> On Wed, 4 Jun 2014 00:46:40 -0700 (PDT), Alisson Vieira
>>=20
>> <alissonvieira01@gmail.com> wrote:
>>=20
>>=20
>>=20
>> <snip>
>>=20
>>=20
>>=20
>> >Thanks Bob. Very nice from you. See. Sorry about this, but when I said t=
>hat I've used a microphone I have actually used a sound level meter 2238 Me=
>diator from B&K.In this case what would change? thank you
>>=20
>>=20
>>=20
>> Now I'm really confused.  From what I read about the 2238,
>>=20
>> it already comes with 1/3 octave analysis software.  And
>>=20
>> coming from B&K you should certainly be able to trust the
>>=20
>> calibration.
>>=20
>>=20
>>=20
>> So, if you don't use any of your own software, just the B&K
>>=20
>> analysis, do you get reasonable results?
>>=20
>>=20
>>=20
>> If so, then my first guess would be that when you are using
>>=20
>> raw data from the B&K to do your own analysis, that data
>>=20
>> might not include any calibration.  It might be raw volts or
>>=20
>> A/D counts or something.
>>=20
>>=20
>>=20
>> Best regards,
>>=20
>>=20
>>=20
>>=20
>>=20
>> Bob Masta
>>=20
>> =20
>>=20
>>               DAQARTA  v7.50
>>=20
>>    Data AcQuisition And Real-Time Analysis
>>=20
>>               www.daqarta.com
>>=20
>> Scope, Spectrum, Spectrogram, Sound Level Meter
>>=20
>>  Frequency Counter, Pitch Track, Pitch-to-MIDI=20
>>=20
>>    FREE Signal Generator, DaqMusiq generator   =20
>>=20
>>           Science with your sound card!
>
>Let me try to be clearer. I am measuring the noise of an electronic steerin=
>g lock device. The noise only appears when I press a button from outside th=
>e anechoic chamber. When I do this the device will lock or unlock (it will =
>move a pin upwards or downwards).This process takes about 1 second. So, I h=
>ave this device 2238 mediator that uses serial filters. Probably, you know =
>everything about this but I will put here what the manual says about the me=
>ter just to try and explain my situation.
>Serial Filters - the sound level meter measures in one band at a time. The =
>meter will usually be fitted with a single filter circuit, which will be el=
>ectronically switched to measure the different bands. With most modern mete=
>rs you can set the filters to scan through from 31.5Hz to 16kHz automatical=
>ly. The advantage of serial octave band filters is that the meter has only =
>one filter circuit, keeping the cost and power consumption down.
>So, I can't use the device's frequency analysis software to get the 1/3 oct=
>ave bands because I would have to run the device at the precise moment that=
> it changes the frequency band. In order to solve this problem I've connect=
>ed the sound level meter electrical outputs to the data logger to get the S=
>PL variations in real time and then used the program in Matlab that uses pa=
>rallel filters to obtain the 1/3 Octave Bands. When I connect the sound lev=
>el meter to the data logger I can obtain two results the voltage variation =
>and the exactly SPL in dB, which the sound level meter is showing on the di=
>splay.  My question is: in order to filter this signal should I use the raw =
>voltage or should I transform these values in db to pressure as I did befor=
>e?

If I understand correctly, you are not getting SPL from the
data logger, but only from the sound level meter; the logger
is only giving voltage with an unknown calibration factor.
You can use the SPL reading to calibrate the logger voltage.


Take the FFT magnitude spectrum of the voltage (not 1/3
octaves).  Apply the same weighting curve that the SPL meter
uses.  (You mentioned previously that the SPL reading was
A-weighted.)  Now take the RMS sum of all the spectral
lines, or at least those under the curve.  That should be
equivalent to the metered SPL.  

You can now find a gain factor to apply to the spectrum data
to give this same SPL.  You can double-check this by using a
different weighting curve on the meter, which you likewise
apply to your FFT spectrum as above, and you should get
matching results.   You can set the meter to show a single
1/3 octave SPL, then verify that your own 1/3 octave scheme
gives the same SPL when applied to this single octave.

Best regards,





Bob Masta
 
              DAQARTA  v7.50
   Data AcQuisition And Real-Time Analysis
              www.daqarta.com
Scope, Spectrum, Spectrogram, Sound Level Meter
 Frequency Counter, Pitch Track, Pitch-to-MIDI 
   FREE Signal Generator, DaqMusiq generator    
          Science with your sound card!

On Wednesday, June 4, 2014 1:49:47 PM UTC+2, Bob Masta wrote:
> On Wed, 4 Jun 2014 00:46:40 -0700 (PDT), Alisson Vieira
> 
> <alissonvieira01@gmail.com> wrote:
> 
> 
> 
> <snip>
> 
> 
> 
> >Thanks Bob. Very nice from you. See. Sorry about this, but when I said that I've used a microphone I have actually used a sound level meter 2238 Mediator from B&K.In this case what would change? thank you
> 
> 
> 
> Now I'm really confused.  From what I read about the 2238,
> 
> it already comes with 1/3 octave analysis software.  And
> 
> coming from B&K you should certainly be able to trust the
> 
> calibration.
> 
> 
> 
> So, if you don't use any of your own software, just the B&K
> 
> analysis, do you get reasonable results?
> 
> 
> 
> If so, then my first guess would be that when you are using
> 
> raw data from the B&K to do your own analysis, that data
> 
> might not include any calibration.  It might be raw volts or
> 
> A/D counts or something.
> 
> 
> 
> Best regards,
> 
> 
> 
> 
> 
> Bob Masta
> 
>  
> 
>               DAQARTA  v7.50
> 
>    Data AcQuisition And Real-Time Analysis
> 
>               www.daqarta.com
> 
> Scope, Spectrum, Spectrogram, Sound Level Meter
> 
>  Frequency Counter, Pitch Track, Pitch-to-MIDI 
> 
>    FREE Signal Generator, DaqMusiq generator    
> 
>           Science with your sound card!

Let me try to be clearer. I am measuring the noise of an electronic steering lock device. The noise only appears when I press a button from outside the anechoic chamber. When I do this the device will lock or unlock (it will move a pin upwards or downwards).This process takes about 1 second. So, I have this device 2238 mediator that uses serial filters. Probably, you know everything about this but I will put here what the manual says about the meter just to try and explain my situation.
Serial Filters - the sound level meter measures in one band at a time. The meter will usually be fitted with a single filter circuit, which will be electronically switched to measure the different bands. With most modern meters you can set the filters to scan through from 31.5Hz to 16kHz automatically. The advantage of serial octave band filters is that the meter has only one filter circuit, keeping the cost and power consumption down.
So, I can't use the device's frequency analysis software to get the 1/3 octave bands because I would have to run the device at the precise moment that it changes the frequency band. In order to solve this problem I've connected the sound level meter electrical outputs to the data logger to get the SPL variations in real time and then used the program in Matlab that uses parallel filters to obtain the 1/3 Octave Bands. When I connect the sound level meter to the data logger I can obtain two results the voltage variation and the exactly SPL in dB, which the sound level meter is showing on the display. My question is: in order to filter this signal should I use the raw voltage or should I transform these values in db to pressure as I did before?

On Wednesday, June 4, 2014 1:49:47 PM UTC+2, Bob Masta wrote:
> On Wed, 4 Jun 2014 00:46:40 -0700 (PDT), Alisson Vieira
> 
> <alissonvieira01@gmail.com> wrote:
> 
> 
> 
> <snip>
> 
> 
> 
> >Thanks Bob. Very nice from you. See. Sorry about this, but when I said that I've used a microphone I have actually used a sound level meter 2238 Mediator from B&K.In this case what would change? thank you
> 
> 
> 
> Now I'm really confused.  From what I read about the 2238,
> 
> it already comes with 1/3 octave analysis software.  And
> 
> coming from B&K you should certainly be able to trust the
> 
> calibration.
> 
> 
> 
> So, if you don't use any of your own software, just the B&K
> 
> analysis, do you get reasonable results?
> 
> 
> 
> If so, then my first guess would be that when you are using
> 
> raw data from the B&K to do your own analysis, that data
> 
> might not include any calibration.  It might be raw volts or
> 
> A/D counts or something.
> 
> 
> 
> Best regards,
> 
> 
> 
> 
> 
> Bob Masta
> 
>  
> 
>               DAQARTA  v7.50
> 
>    Data AcQuisition And Real-Time Analysis
> 
>               www.daqarta.com
> 
> Scope, Spectrum, Spectrogram, Sound Level Meter
> 
>  Frequency Counter, Pitch Track, Pitch-to-MIDI 
> 
>    FREE Signal Generator, DaqMusiq generator    
> 
>           Science with your sound card!

Let me try to be clearer. I am measuring the noise of an electronic steering lock device. The noise appears when I press a button from outside the anechoic chamber. When I press this button the device will lock or unlock (it will move a pin upwards or downwards). So, I have this device 2238 mediator that uses serial filters. Probably, you know everything about this but I will put here what the manual says about the meter just to try and explain my situation.
Serial Filters - the sound level meter measures in one band at a time. The meter will usually be fitted with a single filter circuit, which will be electronically switched to measure the different bands. With most modern meters you can set the filters to scan through from 31.5Hz to 16kHz automatically. The advantage of serial octave band filters is that the meter has only one filter circuit, keeping the cost and power consumption down.
So, I can't use the device's frequency analysis software to get the 1/3 octave bands because I would have to run the device at the precise moment that it changes the frequency band. In order to solve this problem I've connected the sound level meter electrical outputs to the data logger to get the SPL variations in real time and then used the program in Matlab that uses parallel filters to obtain the 1/3 Octave Bands. When I connect the sound level meter to the data logger I can obtain two results the voltage variation and the exactly SPL in dB, which the sound level meter is showing on the display. My question is: in order to filter this signal should I use the raw voltage or should I transform these values in db to pressure as I did before?

On Wednesday, June 4, 2014 1:49:47 PM UTC+2, Bob Masta wrote:
> On Wed, 4 Jun 2014 00:46:40 -0700 (PDT), Alisson Vieira
> 
> <alissonvieira01@gmail.com> wrote:
> 
> 
> 
> <snip>
> 
> 
> 
> >Thanks Bob. Very nice from you. See. Sorry about this, but when I said that I've used a microphone I have actually used a sound level meter 2238 Mediator from B&K.In this case what would change? thank you
> 
> 
> 
> Now I'm really confused.  From what I read about the 2238,
> 
> it already comes with 1/3 octave analysis software.  And
> 
> coming from B&K you should certainly be able to trust the
> 
> calibration.
> 
> 
> 
> So, if you don't use any of your own software, just the B&K
> 
> analysis, do you get reasonable results?
> 
> 
> 
> If so, then my first guess would be that when you are using
> 
> raw data from the B&K to do your own analysis, that data
> 
> might not include any calibration.  It might be raw volts or
> 
> A/D counts or something.
> 
> 
> 
> Best regards,
> 
> 
> 
> 
> 
> Bob Masta
> 
>  
> 
>               DAQARTA  v7.50
> 
>    Data AcQuisition And Real-Time Analysis
> 
>               www.daqarta.com
> 
> Scope, Spectrum, Spectrogram, Sound Level Meter
> 
>  Frequency Counter, Pitch Track, Pitch-to-MIDI 
> 
>    FREE Signal Generator, DaqMusiq generator    
> 
>           Science with your sound card!

This is the problem. The 2238 mediator calculates the 1/3 Octave bands in a serial way (not in parallel). So, once i have only an impulsive noise it is almost impossible to measure the noise in each frequency band. That is why i used the data logger to measure the amplitude vs time and then tried to convert it in 1/3 Octave bands. Did you understand?

On Wed, 4 Jun 2014 00:46:40 -0700 (PDT), Alisson Vieira
<alissonvieira01@gmail.com> wrote:

<snip>

>Thanks Bob. Very nice from you. See. Sorry about this, but when I said that I've used a microphone I have actually used a sound level meter 2238 Mediator from B&K.In this case what would change? thank you

Now I'm really confused.  From what I read about the 2238,
it already comes with 1/3 octave analysis software.  And
coming from B&K you should certainly be able to trust the
calibration.

So, if you don't use any of your own software, just the B&K
analysis, do you get reasonable results?

If so, then my first guess would be that when you are using
raw data from the B&K to do your own analysis, that data
might not include any calibration.  It might be raw volts or
A/D counts or something.

Best regards,


Bob Masta
 
              DAQARTA  v7.50
   Data AcQuisition And Real-Time Analysis
              www.daqarta.com
Scope, Spectrum, Spectrogram, Sound Level Meter
 Frequency Counter, Pitch Track, Pitch-to-MIDI 
   FREE Signal Generator, DaqMusiq generator    
          Science with your sound card!

On Tuesday, June 3, 2014 3:35:01 PM UTC+2, Bob Masta wrote:
> On Mon, 2 Jun 2014 14:52:48 -0700 (PDT), Alisson Vieira
> 
> <alissonvieira01@gmail.com> wrote:
> 
> 
> 
> >On Saturday, May 31, 2014 2:29:51 PM UTC+2, Bob Masta wrote:
> 
> >> On Wed, 28 May 2014 08:58:41 -0700 (PDT), Alisson Vieira
> 
> >> 
> 
> >> <alissonvieira01@gmail.com> wrote:
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> >On Tuesday, May 27, 2014 11:33:45 AM UTC-3, Randy Yates wrote:
> 
> >> 
> 
> >> >> Alisson Vieira <alissonvieira01@gmail.com> writes:
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > Hi dears,
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > Good Morning!
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > I am new in signal processing and I am trying to do a work in noise
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > control of an electronic steering lock device (ESL). My aim is to
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > calculate the loudness (Zwicker Method- ISO 532 B) of this device. To
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > do so, first I need to obtain the 1/3 octave spectrum of a time signal
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > that I measure with a microphone. The problem is I keep getting
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > negative values in dB for the 1/3 Octave bands after filtering the
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > signal in the time domain to obtain the spectrum. I will explain here
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > the procedure I have used and hope that anyone sees what I am doing
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > wrong. Thanks in advance.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > I have done the following procedure by now:
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > 1- Sampled the noise signal (impulsive noise) by using a microphone
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > and a data logger (to record the data), which has a sample frequency
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > of 50K Hz. Then, after this step I have a Curve that it is Amplitude
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > (dBA) vs time (s), as shown below. Once the (dBA) value of a sound
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > level meter is calculated by 10*log10(p^2/p0^2), where p0 is 20e-6 Pa.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > I am able to evaluate the pressure variation (Pa) vs time and use it
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > as INPUT of the 1/3 Octave filters.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > 2- I get the vector INPUT (with 250000 points of pressure
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > (Pa)-measurements of 5s) and use a function in matlab, in order to
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > filter the signal in each each 1/3 octave band.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > 3- Then, the program calculates the rms value of the OUTPUT (after filt=
> 
> >> 
> 
> >> >ering). And this is the value that represents each frequency band.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > 4- Finally, I use the same expression used before to calculate the Magn=
> 
> >> 
> 
> >> >itude in dB for each 1/3 Octave band. 10*log10(p^2/p0^2), where p0 is 20e-6=
> 
> >> 
> 
> >> > Pa.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > The thing is the obtained 1/3 Octave is lower than 0 dB and this
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > doesn't make sense once I can hear the noise when I run the device,
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > moreover it doesn't make sense to calculate the loudness following the
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > ISO 532 B if we have negative third octave bands.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > It seems like the pressure that I have in time domain that is higher,
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > then the reference pressure somehow is attenuated and gets lower than
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > the reference pressure after filtering.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> >
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> > Does anybody know what i am doing wrong?
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> The only thing wrong is your interpretation of the results. While the
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> sound pressure of the entire signal may be well above 20e-6 Pa (RMS),=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> the sound pressure of a single band, especially in the lower frequency
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> area, may not be. At 300 Hz, 1 octave is 300 Hz and 1/3 octave is 100
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> Hz. If the noise was spread evenly in frequency, that would be=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>   20*log10(100/20000) =3D -46 dB
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> below the total power.=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> Not only this, but your noise spectrum is probably far from white
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> (flat). So most of the noise will be some frequency range, meaning other
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> ranges will be below 20e-6 Pa.
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> --=20
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> Randy Yates
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> Digital Signal Labs
> 
> >> 
> 
> >> >>=20
> 
> >> 
> 
> >> >> http://www.digitalsignallabs.com
> 
> >> 
> 
> >> >
> 
> >> 
> 
> >> > Thank you very much Randy Yates for replying.=20
> 
> >> 
> 
> >> >The problem is that I don't get positive values in any frequency band. The =
> 
> >> 
> 
> >> >whole spectrum is below 0 dB. Probably you are pretty busy but if you have =
> 
> >> 
> 
> >> >some time could you take a quick look in a word file which I've prepared wi=
> 
> >> 
> 
> >> >th pictures and results of the problem? could you give me your email or is =
> 
> >> 
> 
> >> >there any other way for me to show you this results? thanks in advance
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> What Randy is saying is that you shouldn't necessarily
> 
> >> 
> 
> >> expect positive SPL in any *single* band... you have to
> 
> >> 
> 
> >> combine all the bands (in RMS fashion) to get the overall
> 
> >> 
> 
> >> SPL.   
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> Even if you have a priori knowledge that a certain band is
> 
> >> 
> 
> >> well above 0 dB SPL, the individual spectral lines within
> 
> >> 
> 
> >> that band will be less than the band total.
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> As a simple example, consider the flat spectrum of a white
> 
> >> 
> 
> >> noise:  If you double the number of samples used to compute
> 
> >> 
> 
> >> the spectrum, the individual spectrum lines will be 3 dB
> 
> >> 
> 
> >> lower... but there will be twice as many of them, so you
> 
> >> 
> 
> >> still get the same RMS overall.
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> However, I must also ask if you are sure of your microphone
> 
> >> 
> 
> >> and system calibration (since you say you are new to DSP).
> 
> >> 
> 
> >> If a system doesn't have calibration data, it's common
> 
> >> 
> 
> >> practice to make full-scale the 0 dB reference, so all your
> 
> >> 
> 
> >> real-world data will of course be negative dB.  That's not
> 
> >> 
> 
> >> dB SPL, just relative dB.  
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> Best regards,
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> 
> 
> >> Bob Masta
> 
> >> 
> 
> >>  
> 
> >> 
> 
> >>               DAQARTA  v7.50
> 
> >> 
> 
> >>    Data AcQuisition And Real-Time Analysis
> 
> >> 
> 
> >>               www.daqarta.com
> 
> >> 
> 
> >> Scope, Spectrum, Spectrogram, Sound Level Meter
> 
> >> 
> 
> >>  Frequency Counter, Pitch Track, Pitch-to-MIDI 
> 
> >> 
> 
> >>    FREE Signal Generator, DaqMusiq generator    
> 
> >> 
> 
> >>           Science with your sound card!
> 
> >
> 
> >Thanks Bob for replying. Yep. I am not sure about the calibration. I will check this out tomorrow. What do you advise me to do? I have done some tests and it seems that the problem is either calibration or the way i handle the data before filtering.
> 
> 
> 
> You mentioned that you used a microphone and a data logger
> 
> to collect data.  You'll need calibration data for the mic,
> 
> either as a list of frequencies vs dB, or as a .FRD file
> 
> that the logger can read.  
> 
> 
> 
> I don't necessarily expect that the logger can incorporate
> 
> frequency response calibration data... you'll have to check
> 
> your logger docs.  Assuming it doesn't, you may be able to
> 
> do it yourself by subtracting the relevant dB value at each
> 
> frequency of the spectrum.  Or you may be able to skip all
> 
> that and just deal with the raw sensitivity, if the mic is
> 
> flat enough in the region of interest.
> 
> 
> 
> The calibration data is typically given as a base
> 
> sensitivity, such as SPL required to get 1 VRMS from the
> 
> mic, or dB relative to 1 V/Pa, etc.  Then there should be 
> 
> a list of dB deviations from that versus frequency.
> 
> 
> 
> If you have a lab-type condenser mic (such as those made by
> 
> B&K, ACO, etc) you may be able to get by without the
> 
> frequency response data because the response is very flat up
> 
> to some limit frequency.  (Smaller mics handle higher
> 
> frequencies, but are less sensitive.) You'll need a special
> 
> preamp/power supply to go with such a mic.  As long as the
> 
> frequency range of interest is well below the roll-off, you
> 
> can use the sensitivity spec alone.
> 
> 
> 
> Some suppliers offer calibrated electret mics which don't
> 
> need the fancy preamp.  They also supply the calibration
> 
> data, so you can check if the mic response is flat in the
> 
> region of interest.
> 
> 
> 
> I should offer a caveat here that calibration can be really
> 
> confusing, so it's a good idea to have some sort of reality
> 
> check.  Even a cheapie Radio Shack sound level meter can
> 
> help confirm that your overall calibration is correct for
> 
> some band that the meter can handle.
> 
> 
> 
> It sounds like you have some prior knowledge of how loud the
> 
> test unit should be (since you state that you are not seeing
> 
> that SPL).   What form is that data?  If it's a simple
> 
> A-weighted SPL at a specified distance, then of course the
> 
> original issue applies:  The *overall* SPL is the RMS sum of
> 
> all the individual spectral lines, so any individual line
> 
> will of course be less... maybe much less, depending on how
> 
> many lines are used.
> 
> 
> 
> Best regards,
> 
> 
> 
> 
> 
> Bob Masta
> 
>  
> 
>               DAQARTA  v7.50
> 
>    Data AcQuisition And Real-Time Analysis
> 
>               www.daqarta.com
> 
> Scope, Spectrum, Spectrogram, Sound Level Meter
> 
>  Frequency Counter, Pitch Track, Pitch-to-MIDI 
> 
>    FREE Signal Generator, DaqMusiq generator    
> 
>           Science with your sound card!


Thanks Bob. Very nice from you. See. Sorry about this, but when I said that I've used a microphone I have actually used a sound level meter 2238 Mediator from B&K.In this case what would change? thank you

On Mon, 2 Jun 2014 14:52:48 -0700 (PDT), Alisson Vieira
<alissonvieira01@gmail.com> wrote:

>On Saturday, May 31, 2014 2:29:51 PM UTC+2, Bob Masta wrote:
>> On Wed, 28 May 2014 08:58:41 -0700 (PDT), Alisson Vieira
>> 
>> <alissonvieira01@gmail.com> wrote:
>> 
>> 
>> 
>> >On Tuesday, May 27, 2014 11:33:45 AM UTC-3, Randy Yates wrote:
>> 
>> >> Alisson Vieira <alissonvieira01@gmail.com> writes:
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >> > Hi dears,
>> 
>> >>=20
>> 
>> >> > Good Morning!
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> > I am new in signal processing and I am trying to do a work in noise
>> 
>> >>=20
>> 
>> >> > control of an electronic steering lock device (ESL). My aim is to
>> 
>> >>=20
>> 
>> >> > calculate the loudness (Zwicker Method- ISO 532 B) of this device. To
>> 
>> >>=20
>> 
>> >> > do so, first I need to obtain the 1/3 octave spectrum of a time signal
>> 
>> >>=20
>> 
>> >> > that I measure with a microphone. The problem is I keep getting
>> 
>> >>=20
>> 
>> >> > negative values in dB for the 1/3 Octave bands after filtering the
>> 
>> >>=20
>> 
>> >> > signal in the time domain to obtain the spectrum. I will explain here
>> 
>> >>=20
>> 
>> >> > the procedure I have used and hope that anyone sees what I am doing
>> 
>> >>=20
>> 
>> >> > wrong. Thanks in advance.
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> > I have done the following procedure by now:
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> > 1- Sampled the noise signal (impulsive noise) by using a microphone
>> 
>> >>=20
>> 
>> >> > and a data logger (to record the data), which has a sample frequency
>> 
>> >>=20
>> 
>> >> > of 50K Hz. Then, after this step I have a Curve that it is Amplitude
>> 
>> >>=20
>> 
>> >> > (dBA) vs time (s), as shown below. Once the (dBA) value of a sound
>> 
>> >>=20
>> 
>> >> > level meter is calculated by 10*log10(p^2/p0^2), where p0 is 20e-6 Pa.
>> 
>> >>=20
>> 
>> >> > I am able to evaluate the pressure variation (Pa) vs time and use it
>> 
>> >>=20
>> 
>> >> > as INPUT of the 1/3 Octave filters.
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> > 2- I get the vector INPUT (with 250000 points of pressure
>> 
>> >>=20
>> 
>> >> > (Pa)-measurements of 5s) and use a function in matlab, in order to
>> 
>> >>=20
>> 
>> >> > filter the signal in each each 1/3 octave band.
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> > 3- Then, the program calculates the rms value of the OUTPUT (after filt=
>> 
>> >ering). And this is the value that represents each frequency band.
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> > 4- Finally, I use the same expression used before to calculate the Magn=
>> 
>> >itude in dB for each 1/3 Octave band. 10*log10(p^2/p0^2), where p0 is 20e-6=
>> 
>> > Pa.
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> > The thing is the obtained 1/3 Octave is lower than 0 dB and this
>> 
>> >>=20
>> 
>> >> > doesn't make sense once I can hear the noise when I run the device,
>> 
>> >>=20
>> 
>> >> > moreover it doesn't make sense to calculate the loudness following the
>> 
>> >>=20
>> 
>> >> > ISO 532 B if we have negative third octave bands.
>> 
>> >>=20
>> 
>> >> > It seems like the pressure that I have in time domain that is higher,
>> 
>> >>=20
>> 
>> >> > then the reference pressure somehow is attenuated and gets lower than
>> 
>> >>=20
>> 
>> >> > the reference pressure after filtering.
>> 
>> >>=20
>> 
>> >> >
>> 
>> >>=20
>> 
>> >> > Does anybody know what i am doing wrong?
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >> The only thing wrong is your interpretation of the results. While the
>> 
>> >>=20
>> 
>> >> sound pressure of the entire signal may be well above 20e-6 Pa (RMS),=20
>> 
>> >>=20
>> 
>> >> the sound pressure of a single band, especially in the lower frequency
>> 
>> >>=20
>> 
>> >> area, may not be. At 300 Hz, 1 octave is 300 Hz and 1/3 octave is 100
>> 
>> >>=20
>> 
>> >> Hz. If the noise was spread evenly in frequency, that would be=20
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >>   20*log10(100/20000) =3D -46 dB
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >> below the total power.=20
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >>=20
>> 
>> >> Not only this, but your noise spectrum is probably far from white
>> 
>> >>=20
>> 
>> >> (flat). So most of the noise will be some frequency range, meaning other
>> 
>> >>=20
>> 
>> >> ranges will be below 20e-6 Pa.
>> 
>> >>=20
>> 
>> >> --=20
>> 
>> >>=20
>> 
>> >> Randy Yates
>> 
>> >>=20
>> 
>> >> Digital Signal Labs
>> 
>> >>=20
>> 
>> >> http://www.digitalsignallabs.com
>> 
>> >
>> 
>> > Thank you very much Randy Yates for replying.=20
>> 
>> >The problem is that I don't get positive values in any frequency band. The =
>> 
>> >whole spectrum is below 0 dB. Probably you are pretty busy but if you have =
>> 
>> >some time could you take a quick look in a word file which I've prepared wi=
>> 
>> >th pictures and results of the problem? could you give me your email or is =
>> 
>> >there any other way for me to show you this results? thanks in advance
>> 
>> 
>> 
>> What Randy is saying is that you shouldn't necessarily
>> 
>> expect positive SPL in any *single* band... you have to
>> 
>> combine all the bands (in RMS fashion) to get the overall
>> 
>> SPL.   
>> 
>> 
>> 
>> Even if you have a priori knowledge that a certain band is
>> 
>> well above 0 dB SPL, the individual spectral lines within
>> 
>> that band will be less than the band total.
>> 
>> 
>> 
>> As a simple example, consider the flat spectrum of a white
>> 
>> noise:  If you double the number of samples used to compute
>> 
>> the spectrum, the individual spectrum lines will be 3 dB
>> 
>> lower... but there will be twice as many of them, so you
>> 
>> still get the same RMS overall.
>> 
>> 
>> 
>> However, I must also ask if you are sure of your microphone
>> 
>> and system calibration (since you say you are new to DSP).
>> 
>> If a system doesn't have calibration data, it's common
>> 
>> practice to make full-scale the 0 dB reference, so all your
>> 
>> real-world data will of course be negative dB.  That's not
>> 
>> dB SPL, just relative dB.  
>> 
>> 
>> 
>> Best regards,
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> Bob Masta
>> 
>>  
>> 
>>               DAQARTA  v7.50
>> 
>>    Data AcQuisition And Real-Time Analysis
>> 
>>               www.daqarta.com
>> 
>> Scope, Spectrum, Spectrogram, Sound Level Meter
>> 
>>  Frequency Counter, Pitch Track, Pitch-to-MIDI 
>> 
>>    FREE Signal Generator, DaqMusiq generator    
>> 
>>           Science with your sound card!
>
>Thanks Bob for replying. Yep. I am not sure about the calibration. I will check this out tomorrow. What do you advise me to do? I have done some tests and it seems that the problem is either calibration or the way i handle the data before filtering.

You mentioned that you used a microphone and a data logger
to collect data.  You'll need calibration data for the mic,
either as a list of frequencies vs dB, or as a .FRD file
that the logger can read.  

I don't necessarily expect that the logger can incorporate
frequency response calibration data... you'll have to check
your logger docs.  Assuming it doesn't, you may be able to
do it yourself by subtracting the relevant dB value at each
frequency of the spectrum.  Or you may be able to skip all
that and just deal with the raw sensitivity, if the mic is
flat enough in the region of interest.

The calibration data is typically given as a base
sensitivity, such as SPL required to get 1 VRMS from the
mic, or dB relative to 1 V/Pa, etc.  Then there should be 
a list of dB deviations from that versus frequency.

If you have a lab-type condenser mic (such as those made by
B&K, ACO, etc) you may be able to get by without the
frequency response data because the response is very flat up
to some limit frequency.  (Smaller mics handle higher
frequencies, but are less sensitive.) You'll need a special
preamp/power supply to go with such a mic.  As long as the
frequency range of interest is well below the roll-off, you
can use the sensitivity spec alone.

Some suppliers offer calibrated electret mics which don't
need the fancy preamp.  They also supply the calibration
data, so you can check if the mic response is flat in the
region of interest.

I should offer a caveat here that calibration can be really
confusing, so it's a good idea to have some sort of reality
check.  Even a cheapie Radio Shack sound level meter can
help confirm that your overall calibration is correct for
some band that the meter can handle.

It sounds like you have some prior knowledge of how loud the
test unit should be (since you state that you are not seeing
that SPL).   What form is that data?  If it's a simple
A-weighted SPL at a specified distance, then of course the
original issue applies:  The *overall* SPL is the RMS sum of
all the individual spectral lines, so any individual line
will of course be less... maybe much less, depending on how
many lines are used.

Best regards,


Bob Masta
 
              DAQARTA  v7.50
   Data AcQuisition And Real-Time Analysis
              www.daqarta.com
Scope, Spectrum, Spectrogram, Sound Level Meter
 Frequency Counter, Pitch Track, Pitch-to-MIDI 
   FREE Signal Generator, DaqMusiq generator    
          Science with your sound card!