calibrate sound card algorithm for obtain good frequency response

[peter.go]

Hi community:

I need calibrate my sound card of the PC, to measurements audio signal, because the frequency response is not good at all. I need some algorithm or method to do this. Please help me

[GerdW]

Hi Peter,

you know Wikipedia?

There are a lot of entries on (basic) technical topics - like yours…

[James.M]

What have you done, what have you tried? What you've stated here sounds like the type of thing you would send to a contractor.

[peter.go]

I now the concept of frequency response this is not the problem, the questions is how compense the nonlinear characteristics of my sound card in the 315-6300 Hz range to make a precision measurement Vrms of audio signal in the line input in this bandwith

[James.M]

So what is your question?

Do you need help acquiring an audio input signal?

Do you need help altering the input data?

Do you need help displaying the signal?

Do you need help developing the mathematics?

Have you tried any of this?

Do you have code to show us what you've tried?

Do you know how to use LabVIEW?

[peter.go]

I need equalizer or calibrate my sound card for obtain a linear freq response

I generate a sweep frequency audio signal with constant amplitude by the headphone output of my PC and I read this signal by the line input of my PC, if my sound card have a linear frequency response i obtain the same amplitude in each one of frequency. But because the sound card have a nonlinear frequency response I need of some way compensate this value to obtain a linear response

All programs i see in internet that use a sound card for measurements like REW say that the first step is calibrate the sound card, How i do this in Labview'??????

[James.M]

Why don't you use the REW to do the calibration?

[peter.go]

because REW´s calibrations values onnly work when you use REW for measurements not other programs like LabVIEW.

I need make some similar in LabVIEW because the generation and measurements of the audio signal I make in LabVIEW enviroments

[johnsold]

To do a calibration requires a known reference source or signal. If you are generating and measuring the sound with LabVIEW and the sound card in the computer, you have no known reference.

- LabVIEW generates numbers. They may be quite precise but do not relate to sound levels in any absolute manner.

- The sound card D/A converter changes the numbers generated by LabVIEW to a voltage or current. All such devices have both random and systematic errors in the conversion process. Further you may not have access to the manufacturer's nominal conversion factor (number to voltage).

- The sound card amplifier has an unknown frequency response, distortion, and noise.

- The speaker has an unknown frequency response and distortion and is likely one of the largest contributors to the non-ideal overall response.

- The micorphone, like the speaker, has an unknown frequency response and distortion. It may also be a major contributor to the non-ideal overall response.

- The preamplifer has an unknown frequency response, distortion, and noise.

- The A/D converter has the same kinds of errors as the D/A converter.

Lynn

[Logan_K]

To summarize... if you want to calibrate the audio input, you need an accurate audio output signal. "Accurate" is a relative term that depends on the needs of  your application. If your sound card has a reasonably accurate Audio output (specifically more accurate than the input), then using a loopback method might work. If not, then you would need a different audio output device that is more accurate. In order answer this question you'll need to know your target flatness spec, and the flatness of the output device you plan to use.