Additional functions

Scaling of the input signal

Most of the nonlinear elements of an auditory model, e.g., at the level of mechanical-to-neural transduction and neural adaptation are level-dependent. Therefore, it is necessary to fix the amplitude range and its relation to the sound pressure level of a real acoustic signal. Here, we adopt use a convention that the root-mean-square, RMS, value 1 in the digital signal corresponds to the sound pressure level of 0 dB. The function Pascalize can be used to scale a steady-state signal such that it corresponds to a sound pressure level of A dB. For example,

>> ssig=Pascalize(sig,60);

scales a signal sig such that its sound pressure level is 60 dB.

Design of gammatone filters

If the gammatone filterbank is defined in the field model.cochlea.gt, see above, the function Gammatone designs a set of filter and use them to do the cochlear filtering. It is usually more efficient to design the filterbank in advance, store the filter coefficient in a file and use model.cochlea.fb.file field to specify the filterbank.

Make_GTbank is an interactive function which can be used to design various types of gammatone filterbanks. This is called by

>> [f,b]=Make_GTbank;

The function asks first user to specify the type of the filterbank. There are three alternatives: (1) conventional gammatone implementation is the one introduced by Malcolm Slaney [8]. The second and third alternatives are a (2) fourth-order and a (3) sixth-order complex-valued gammatone filterbank for which HUTear has an efficient implementation which uses cgtbank.mex** function. Derivation of these filters can be found Here.

Next, the user is asked to give sampling rate in Hertz. After that, the function gives three alternatives: (1) specify all center frequencies in Hertz, (2) specify all center frequencies in ERB (1-42), and (3) give the number of channels and the lowest center frequency.

After that, the user is asked if he/she wants to store the coefficients directly to a file or to just take the coefficient matrices to Matlab's workspace. AudMod assumes that the coefficients are row vectors in matrices which named f and b as above.

Design of an outer and middle ear filter

As shown above, the model for outer and middle ear is usually a linear filter. The coefficients of the filter are in a mat-file which is set by the model.outmid.file field. This filter may be designed using the Make_OEM function.

For example,

>> [a,b]=Make_OEM(44100,255,'ELC','FIR');
>> save filename.mat a b

designs a 255th order FIR approximation at 44100 Hz sampling rate which is matched with the Equal Loudness Curves at the loudness level of 0 phon. The coefficients are stored here in a mat-file (filename.mat). It is important that the coefficient vectors are named a and b. Otherwise, AudMod is not able to identify the filter coefficients.

It is possible to design also IIR filters using the same function, see help Make_OEM. In addition, if WarpTB is available, it is also possible to design and use frequency-warped FIR and IIR filters.

There is an interactive graphical user interface for Make_OEM in the toolbox. It is called by

>> OEMtool

It can be used to compare different implementation technique as a function of model order and finally to store the coefficients to a named file.