Audio Measurement and Analysis Software

 Frequently asked questions

Q: I have a calibration file for the microphone FR. Is there possibility to include such a calibration file in ARTA or do you plan such a feature?

A: Starting from version 1.0.1, ARTA has capability to apply frequency response compensation to spectrum and frequency response curves. It is expected that frequency response, i.e. response of microphone  that has to be compensated, is  in ASCII formatted textual file with name extension .MIC. Format of that file is as follows:

a) .MIC file contains lines of text

b) Lines that start with digits or dot character are expected to contains two numeric value separated with spaces or tab characters; first value is frequency in Hz and second value is magnitude of  frequency response in dB. After these two values the line can contain any text up to the end of line. Usually, a point in the middle of the frequency range is chosen as reference 0dB value.

c) All other lines are treated as comment

d) You can edit .MIC file with any text editor (i.e. Notepad), but keep in mind that frequencies must be entered in sorted order.

Here you can download a sample of .MIC file for the microphone MB550.

Good source of cheap calibrated microphones is at http://lasip.hifi-selbstbau.de/.

You can define .MIC compensation files for other purposes. I.e. Peter de Jong has contibuted  .MIC files (RIAA_MIC.ZIP) to compensate for phone RIAA response.


Q: What are differences between MLS and Fourier Analyzer?

A: Both types of analyzers use crosscorelation method and signal/spectrum averaging to estimate the impulse/frequency response. The fundamental difference is how crosscorrelation estimation is done; MLS analyzers uses Hadamard transform, while Fourier analyzer uses Fourier transform. The second difference is that MLS analyzers exclusively use the MLS  excitation signal, while Fourier analyzers use various excitation signals: noise, periodic noise, swept-sine or wideband music signals. The ARTA has implemented both measurement methods. It is left to users to decide which type of analyzers will be used for response estimation, but ARTA User Manual suggests the use of Fourier analyzer with swept-sine or periodic pink-noise excitation.

Following papers discuss the problem more deeply:

I. Mateljan, K. Ugrinovic: The Comparison of Room Impulse Response Measuring Systems, Proceedings of the First Congress of Alps Adria Acoustics Association,  Portoroz, Slovenia, 2003, ISBN 961-6238-73-6

I. Mateljan: Audio Quality Measurements in Communication Systems, Proceedings of the Second Congress of Alps Adria Acoustics Association, Opatija, Croatia, 2005, ISBN 953-95097-0-X.


Q: Why ARTA uses "dual channel Fourier analyzer" mode and "single channel Fourier analyzer" mode?

A: A dual channel mode is proper mode to estimate frequency/impulse response. Single channel mode always gives biased IR estimation, but if you have a low quality soundcard "single channel mode" can give better S/N at low frequencies. Single channel mode is obligate if you use the soundcard microphone input channel, as it is, on most soundcards, implemented as mono channel

Q: When measuring loudspeaker impedance with LIMP I get a noisy measurement results. What is the problem?

A: LIMP uses fast method to estimate the loudspeaker impedance. The method is susceptible to noise (remember that loudspeaker acts as microphone for environmental noise). To eliminate the influence of noise:

a) Use the power amplifier to drive the loudspeaker through a serial reference impedance in the range 10-100 ohms,
b) Adjust levels on soundcard input channels to be in the range -10 to -1 dBFS.
c) Monitor the possible soundcard input channel overflow (input level 0dBFS)

If you use the soundcard line out channel for impedance measurements, then reference resistor must be at least 600 ohms. In that case it is normal to have "noisy" measurement results. You can improve measurement if you have quiet measurement environment and if you apply a large number of averaging.


Q: I have a soundcard that works in 16-bit and 24-bit modes, but in both cases I measure the same noise floor in spectrum analyzer mode. What I made wrong?

A: Nothing is wrong with your measurement. Many souncards that are declared as 24-bit soundcards have effective noise floor higher than some high quality 16-bit soundcards. In that case use soundcard in 16-bit mode to get more processing power.


Q: I have a soundcard that works in Extensible 24-bit mode but it not function properly in Extensible 32-bit mode.  Is my soundcard driver defective?

A: Probably you soundcard driver is OK. The 32-bit mode is alternative way of transferring 24-bit recorded data to computer, and not all soundcard drivers support this mode.


Q: My soundcard works properly in 24-bit and 32-bit mode. Shall I use 24-bit or 32-bit mode?

A: In this case it is recommended to use 32-bit mode as it uses less processing power.


Q: I've measured loudspeaker FR with ARTA in free space (outdoors).  I've got high noise at low frequencies. Help me solve this problem.

A: Measurements outdoors, in free field,  are usually affected with wind and possible traffic noise. In both cases you have large level of noise and distortion in measurement results.  For measurement of frequency response outdoors use STEPS, but if you must use ARTA, to get impulse response, use microphone wind protector and dual channel measurement with averaging. 


Q: Why ARTA is showing impulse response with "zero time" shifted to sample position 300?

A: Modern digital systems often use some kind of finite impulse response (FIR) filtering or equalization. Such kind of filtering introduce predelay (or pre-ringing) in impulse response. The predelay usually uses from 64 to 256 samples. To see effects of digital filter predelay in single channnel mode, ARTA shifts "zero time" to sample position 300.


Q: I have tried to measure the frequency response with ARTA and impedance with LIMP with sampling rate 96kHz. I was hoping that I shall  get response up to 46kHz, instead I got noise like response at frequencies above 22 kHz. What is wrong with my measurement setup?

A: This problem usually arises on some soundcards when cut-off frequency of the aliasing filter is fixed to 22 kHz. How is that possible? Some soundcards can do sample rate conversion by software and by hardware  (i.e. Soundblasater Audigy ZS and almost all soundcards with Firewire interface). That soundcards have their own control panel (mixer) for adjustment of the hardware sampling rate. In cases when soundcard setup in ARTA and soundcard hardware setup have different sampling rates, the common sampling rate it is being adjusted in software. The problem is that this conversion does not change the cut-off frequency of the antialiasing filter. It is always close to the half the sampling rate that is fixed by hardware control-panel setup.