THE AUDITORY MODELING TOOLBOX
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DIETZ2011 - Sound lateral direction
Program code:
function [fine,fc,ild,env] = dietz2011(insig,fs,varargin)
%DIETZ2011 Sound lateral direction
% Usage: [fine,fc,ild,env] = dietz2011(insig,fs);
%
% Input parameters:
% insig : binaural signal for which values should be calculated
% fs : sampling rate (Hz)
%
% Output parameters:
% fine : Information about the fine structure (see below)
% fc : center frequencies of gammatone filterbank
% ild : interaural level difference in dB
% env : Information about the envelope (see below)
%
% DIETZ2011(insig,fs) calculates interaural phase, time and level
% differences of fine- structure and envelope of the signal, as well as
% the interaural coherence, which can be used as a weighting function.
%
% The output structures fine and env have the following fields:
%
% itf transfer function
%
% ipd phase difference in rad
%
% itd interaural time difference based on instantaneous frequency
%
% itd_C interaural time difference based on center frequency
%
% f_inst_1 instantaneous frequencies of left ear signal
%
% f_inst_2 instantaneous frequencies of right ear canal signal
%
% f_inst instantaneous frequencies (average of f_inst1 and 2)
%
% ic interaural coherence
%
% rms rms value of frequency channels for weighting
%
% ild_lp based on low passed-filtered insig, level difference in dB
%
% ipd_lp based on lowpass-filtered itf, phase difference in rad
%
% itd_lp based on lowpass-filtered itf, interaural time difference
%
% itd_C_lp based on lowpass-filtered itf, interaural time difference
%
% f_inst_lp lowpass instantaneous frequencies
%
% The _lp values are not returned if the 'nolowpass' flag is set.
%
% The steps of the binaural model to calculate the result are the
% following (see also Dietz et al., 2011):
%
% 1) Middle ear filtering (500-2000 Hz 1st order bandpass)
%
% 2) Auditory bandpass filtering on the basilar membrane using a
% 4th-order all-pole gammatone filterbank, employing 23 filter
% bands between 200 and 5000 Hz, with a 1 ERB spacing. The filter
% width was set to correspond to 1 ERB.
%
% 3) Cochlear compression was simulated by power-law compression with
% an exponent of 0.4.
%
% 4) The transduction process in the inner hair cells was modelled
% using half-wave rectification followed by filtering with a 770-Hz
% 5th order lowpass.
%
% 5) Modulationfilterbank with three different filters applied to every
% frequency channel. One 2nd order gammatone filter for the fine structure
% centered at the center frequency of the frequency channel. One 2nd order
% gammatone filter for the envelope of the signal centered at 135 Hz.
% And a 2nd order lowpass filter with a cutoff frequency of 30 Hz to
% extract the ILD of the signal.
%
% 6) Calculation of binaural parameters such as IPD, ITD, IC for fine
% structure and envelope filter signals and ILD for the ILD filter.
%
%
% DIETZ2011 accepts the following optional parameters:
%
% 'flow',flow Set the lowest frequency in the filterbank to
% flow. Default value is 200 Hz.
%
% 'fhigh',fhigh Set the highest frequency in the filterbank to
% fhigh. Default value is 5000 Hz.
%
% 'basef',basef Ensure that the frequency basef is a center frequency
% in the filterbank. The default value is 1000.
%
% 'filters_per_ERB',filters_per_erb
% Filters per erb. The default value is 1.
%
% 'middle_ear_thr',r
% Bandpass frequencies for middle ear transfer. The
% default value is [500 2000].
%
% 'middle_ear_order',n
% Order of middle ear filter. Only even numbers are
% possible. The default value is 2.
%
% 'compression_power',cpwr
% Applied compression of the signal on the cochlea with
% ^compression_power. The default value is 0.4.
%
% 'alpha',alpha Internal noise strength. Convention 65dB = 0.0354.
% The default value is 0.
%
% 'int_randn' Internal noise by adding random noise with rms = alpha.
% This is the default.
%
% 'int_mini' Internal noise by setting all values < alpha to alpha.
%
% 'filter_order',fo
% Filter order for the two gammatone filter used for the fine
% structure and envelope of the modulation filter bank. The
% default value is 2.
%
% 'filter_attenuation_db',fadb
% Filter attenuation for the two gammatone filter used for the fine
% structure and envelope of the modulation filter bank. The
% default value is 10.
%
% 'fine_filter_finesse',fff
% Filter finesse (determines the bandwidth with fc/finesse)
% for the fine structure gammatone filter. The default value
% is 3.
%
% 'mod_center_frequency_hz',mcf_hz
% Center frequency of the gammatone envelope filter. The
% default value is 135.
%
% 'mod_filter_finesse',mff
% Filter finesse (determines the bandwidth with fc/finesse)
% for the envelope gammatone filter. The default value is 8.
%
% 'level_filter_cutoff_hz',lfc_hz
% Cutoff frequency off the low pass filter used for ILD
% calculation. The default value is 30.
%
% 'level_filter_order',lforder
% Order of low pass filter for the ILD calculation. The
% default value is 2.
%
% 'tau_cycles',tau_cycles
% Temporal resolution of binaural processor in terms of
% cycles per frequency channel. The default value is 5.
%
% 'signal_level_dB_SPL',signal_level
% Sound pressure level of left channel. Used for data
% display and analysis. Default value is 70.
%
% 'lowpass' Calculate the interaural parameters of the lowpassed
% signal/ITF (_lp return values). This is the default.
%
% 'nolowpass' Don't calculate the lowpass based interaural parameters.
% The _lp values are not returned.
%
% 'debug' Display what is happening.
%
%
% See also: dietz2011_interauralfunctions, dietz2011_filterbank,
% ihcenvelope, auditoryfilterbank itd2angle itd2angle_lookuptable
% sig_competingtalkers dietz2011_filterbank dietz2011_interauralfunctions
% wierstorf2013_estimateazimuth dietz2011_unwrapitd
% exp_steidle2019 exp_dietz2011 breebaart2001 wierstorf2013 hohmann2002
% baumgartner2017
%
% References:
% M. Dietz, S. D. Ewert, and V. Hohmann. Auditory model based direction
% estimation of concurrent speakers from binaural signals. Speech
% Communication, 53(5):592--605, 2011. [1]http ]
%
% References
%
% 1. http://www.sciencedirect.com/science/article/pii/S016763931000097X
%
%
% Url: http://amtoolbox.org/amt-1.1.0/doc/models/dietz2011.php
% Copyright (C) 2009-2021 Piotr Majdak, Clara Hollomey, and the AMT team.
% This file is part of Auditory Modeling Toolbox (AMT) version 1.1.0
%
% This program is free software: you can redistribute it and/or modify
% it under the terms of the GNU General Public License as published by
% the Free Software Foundation, either version 3 of the License, or
% (at your option) any later version.
%
% This program is distributed in the hope that it will be useful,
% but WITHOUT ANY WARRANTY; without even the implied warranty of
% MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
% GNU General Public License for more details.
%
% You should have received a copy of the GNU General Public License
% along with this program. If not, see <http://www.gnu.org/licenses/>.
% This model got a major update, and the figures of exp_dietz2011() look now
% slightly different than before. If you want to restore the original version of
% the model please checkout version 29a048b with git or install AMToolbox 0.9.5
% Copyright (C) 2002-2012 AG Medizinische Physik,
% Universitaet Oldenburg, Germany
% http://www.physik.uni-oldenburg.de/docs/medi
% #StatusDoc: Perfect
% #StatusCode: Perfect
% #Verification: Verified
% #Requirements: M-Signal
% #Author: Tobias Peters (tobias@medi.physik.uni-oldenburg.de) 2002
% #Author: Mathias Dietz (mathias.dietz@uni-oldenburg.de) 2006-2009
% #Author: Martin Klein-Hennig (martin.klein.hennig@uni-oldenburg.de) 2011
% #Author: Martin Klein-Hennig (for AMT)
% #Author: Hagen Wierstorf (for AMT)
if nargin<2
error('%s: Too few input parameters.',upper(mfilename));
end;
if ~isnumeric(insig) || min(size(insig))~=2
error('%s: insig has to be a numeric two channel signal!',upper(mfilename));
end
if ~isnumeric(fs) || ~isscalar(fs) || fs<=0
error('%s: fs has to be a positive scalar!',upper(mfilename));
end
% import default arguments from other functions
definput.import={'auditoryfilterbank','ihcenvelope','dietz2011_filterbank','dietz2011_interauralfunctions'};
% changing default parameters
definput.importdefaults = { ...
'flow',200, ... % gammatone lowest frequency / Hz
'fhigh',5000, ... % gammatone highest frequency / Hz
'basef',1000, ... % auditory filter should be centered at basef / Hz
'ihc_breebaart2001' ... % use haircell parameters as in Breebarts model
};
% Preprocessing parameters
definput.keyvals.middle_ear_thr = [500 2000]; % Bandpass freqencies for middle ear transfer
definput.keyvals.middle_ear_order = 2; % Only even numbers possible
definput.keyvals.alpha = 0; % Internal noise strength
% 65dB = 0.0354
% randn: add random noise with rms = alpha
% mini: set all values < alpha to alpha
definput.flags.int_noise_case = {'int_randn','int_mini'};
% debugging messages
definput.flags.disp = {'no_debug','debug'};
[flags,kv] = ltfatarghelper({},definput,varargin);
%% Model processing starts here %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
amt_disp('DIETZ2011:',flags.disp);
%% ---- middle ear band pass filtering ------
% Compare Puria et al. (1997)
amt_disp(' band pass filtering of input according to middle ear transfer charact.',flags.disp);
[b,a] = butter(kv.middle_ear_order,kv.middle_ear_thr(2)/(fs/2),'low');
inoutsig = filter(b,a,insig);
[b,a] = butter(kv.middle_ear_order,kv.middle_ear_thr(1)/(fs/2),'high');
inoutsig = filter(b,a,inoutsig);
%% ---- inner ear ------
% gammatone filterbank
amt_disp(' splitting signal into frequency channels',flags.disp);
[inoutsig,fc] = auditoryfilterbank(inoutsig,fs,'argimport',flags,kv);
% cochlea compression
inoutsig = sign(inoutsig).*abs(inoutsig).^kv.compression_power;
% rectification and lowpass filtering of filtered signals
amt_disp(' haircell processing of frequency bands',flags.disp);
inoutsig = ihcenvelope(inoutsig,fs,'argimport',flags,kv);
%% ---- internal noise ------
% additive white noise
if flags.do_int_randn
amt_disp(' adding internal random noise',flags.disp);
inoutsig = inoutsig + kv.alpha*randn(size(inoutsig));
end
% replace values<kv.alpha with kv.alpha
if flags.do_int_mini
amt_disp(' adding internal noise via minimum',flags.disp);
inoutsig = max(inoutsig,kv.alpha);
end
%% ---- modulation filterbank ------
% filter signals with three different filters to get fine structure, envelope
% and ILD low pass
amt_disp(' apply second filterbank',flags.disp)
[inoutsig_fine,fc_fine,inoutsig_env,fc_env,inoutsig_ild] = ...
dietz2011_filterbank(inoutsig,fs,fc,'argimport',flags,kv);
%% ---- binaural processor ------
% calculate interaural parameters for fine structure and envelope and calculate
% ILD
% -- fine structure
amt_disp(' calculating interaural functions from haircell fine structure',flags.disp);
fine = dietz2011_interauralfunctions(inoutsig_fine,fs,fc_fine,'argimport',flags,kv);
% --envelope
amt_disp(' calculating interaural functions from haircell modulation',flags.disp);
env = dietz2011_interauralfunctions(inoutsig_env,fs, ...
kv.mod_center_frequency_hz+0*fc_env,'argimport',flags,kv);
% -- ILD
% interaural level difference, eq. 5 in Dietz (2011)
% max(sig,1e-4) avoids division by zero
amt_disp(' determining ILD',flags.disp);
ild = 20/kv.compression_power*log10(max(inoutsig_ild(:,:,2),1e-4)./max(inoutsig_ild(:,:,1),1e-4));
end
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