Forums

1/3 Octave Analysis

Started by Alisson Vieira May 27, 2014
Hi dears,
Good Morning!

I am new in signal processing and I am trying to do a work in noise control of an electronic steering lock device (ESL). My aim is to calculate the loudness (Zwicker Method- ISO 532 B) of this device. To do so, first I need to obtain the 1/3 octave spectrum of a time signal that I measure with a microphone. The problem is I keep getting negative values in dB for the 1/3 Octave bands after filtering the signal in the time domain to obtain the spectrum. I will explain here the procedure I have used and hope that anyone sees what I am doing wrong. Thanks in advance. 

I have done the following procedure by now:

1- Sampled the noise signal (impulsive noise) by using a microphone and a data logger (to record the data), which has a sample frequency of 50K Hz. Then, after this step I have a Curve that it is Amplitude (dBA) vs time (s), as shown below. Once the (dBA) value of a sound level meter is calculated by 10*log10(p^2/p0^2), where p0 is 20e-6 Pa. I am able to evaluate the pressure variation (Pa) vs time and use it as INPUT of the 1/3 Octave filters.



2- I get the vector INPUT (with 250000 points of pressure (Pa)-measurements of 5s) and use a function in matlab, in order to filter the signal in each each 1/3 octave band.

3- Then, the program calculates the rms value of the OUTPUT (after filtering). And this is the value that represents each frequency band.

4- Finally, I use the same expression used before to calculate the Magnitude in dB for each 1/3 Octave band. 10*log10(p^2/p0^2), where p0 is 20e-6 Pa.

The thing is the obtained 1/3 Octave is lower than 0 dB and this doesn't make sense once I can hear the noise when I run the device, moreover it doesn't make sense to calculate the loudness following the ISO 532 B if we have negative third octave bands. 
It seems like the pressure that I have in time domain that is higher, then the reference pressure somehow is attenuated and gets lower than the reference pressure after filtering.

Does anybody know what i am doing wrong?
Alisson Vieira <alissonvieira01@gmail.com> writes:

> Hi dears, > Good Morning! > > I am new in signal processing and I am trying to do a work in noise > control of an electronic steering lock device (ESL). My aim is to > calculate the loudness (Zwicker Method- ISO 532 B) of this device. To > do so, first I need to obtain the 1/3 octave spectrum of a time signal > that I measure with a microphone. The problem is I keep getting > negative values in dB for the 1/3 Octave bands after filtering the > signal in the time domain to obtain the spectrum. I will explain here > the procedure I have used and hope that anyone sees what I am doing > wrong. Thanks in advance. > > I have done the following procedure by now: > > 1- Sampled the noise signal (impulsive noise) by using a microphone > and a data logger (to record the data), which has a sample frequency > of 50K Hz. Then, after this step I have a Curve that it is Amplitude > (dBA) vs time (s), as shown below. Once the (dBA) value of a sound > level meter is calculated by 10*log10(p^2/p0^2), where p0 is 20e-6 Pa. > I am able to evaluate the pressure variation (Pa) vs time and use it > as INPUT of the 1/3 Octave filters. > > > > 2- I get the vector INPUT (with 250000 points of pressure > (Pa)-measurements of 5s) and use a function in matlab, in order to > filter the signal in each each 1/3 octave band. > > 3- Then, the program calculates the rms value of the OUTPUT (after filtering). And this is the value that represents each frequency band. > > 4- Finally, I use the same expression used before to calculate the Magnitude in dB for each 1/3 Octave band. 10*log10(p^2/p0^2), where p0 is 20e-6 Pa. > > The thing is the obtained 1/3 Octave is lower than 0 dB and this > doesn't make sense once I can hear the noise when I run the device, > moreover it doesn't make sense to calculate the loudness following the > ISO 532 B if we have negative third octave bands. > It seems like the pressure that I have in time domain that is higher, > then the reference pressure somehow is attenuated and gets lower than > the reference pressure after filtering. > > Does anybody know what i am doing wrong?
The only thing wrong is your interpretation of the results. While the sound pressure of the entire signal may be well above 20e-6 Pa (RMS), the sound pressure of a single band, especially in the lower frequency area, may not be. At 300 Hz, 1 octave is 300 Hz and 1/3 octave is 100 Hz. If the noise was spread evenly in frequency, that would be 20*log10(100/20000) = -46 dB below the total power. Not only this, but your noise spectrum is probably far from white (flat). So most of the noise will be some frequency range, meaning other ranges will be below 20e-6 Pa. -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com
On Tuesday, May 27, 2014 11:33:45 AM UTC-3, Randy Yates wrote:
> Alisson Vieira <alissonvieira01@gmail.com> writes: > > > > > Hi dears, > > > Good Morning! > > > > > > I am new in signal processing and I am trying to do a work in noise > > > control of an electronic steering lock device (ESL). My aim is to > > > calculate the loudness (Zwicker Method- ISO 532 B) of this device. To > > > do so, first I need to obtain the 1/3 octave spectrum of a time signal > > > that I measure with a microphone. The problem is I keep getting > > > negative values in dB for the 1/3 Octave bands after filtering the > > > signal in the time domain to obtain the spectrum. I will explain here > > > the procedure I have used and hope that anyone sees what I am doing > > > wrong. Thanks in advance. > > > > > > I have done the following procedure by now: > > > > > > 1- Sampled the noise signal (impulsive noise) by using a microphone > > > and a data logger (to record the data), which has a sample frequency > > > of 50K Hz. Then, after this step I have a Curve that it is Amplitude > > > (dBA) vs time (s), as shown below. Once the (dBA) value of a sound > > > level meter is calculated by 10*log10(p^2/p0^2), where p0 is 20e-6 Pa. > > > I am able to evaluate the pressure variation (Pa) vs time and use it > > > as INPUT of the 1/3 Octave filters. > > > > > > > > > > > > 2- I get the vector INPUT (with 250000 points of pressure > > > (Pa)-measurements of 5s) and use a function in matlab, in order to > > > filter the signal in each each 1/3 octave band. > > > > > > 3- Then, the program calculates the rms value of the OUTPUT (after filtering). And this is the value that represents each frequency band. > > > > > > 4- Finally, I use the same expression used before to calculate the Magnitude in dB for each 1/3 Octave band. 10*log10(p^2/p0^2), where p0 is 20e-6 Pa. > > > > > > The thing is the obtained 1/3 Octave is lower than 0 dB and this > > > doesn't make sense once I can hear the noise when I run the device, > > > moreover it doesn't make sense to calculate the loudness following the > > > ISO 532 B if we have negative third octave bands. > > > It seems like the pressure that I have in time domain that is higher, > > > then the reference pressure somehow is attenuated and gets lower than > > > the reference pressure after filtering. > > > > > > Does anybody know what i am doing wrong? > > > > The only thing wrong is your interpretation of the results. While the > > sound pressure of the entire signal may be well above 20e-6 Pa (RMS), > > the sound pressure of a single band, especially in the lower frequency > > area, may not be. At 300 Hz, 1 octave is 300 Hz and 1/3 octave is 100 > > Hz. If the noise was spread evenly in frequency, that would be > > > > 20*log10(100/20000) = -46 dB > > > > below the total power. > > > > Not only this, but your noise spectrum is probably far from white > > (flat). So most of the noise will be some frequency range, meaning other > > ranges will be below 20e-6 Pa. > > -- > > Randy Yates > > Digital Signal Labs > > http://www.digitalsignallabs.com
Thank you very much Randy Yates for replying. 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 with 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
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!
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.
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!
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 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 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?