function [outsig, fc] = dau1996_preproc(insig, fs, varargin);
%dau1996_preproc   Auditory model from Dau et. al. 1996
%   Usage: [outsig, fc] = dau1996_preproc(insig,fs);
%          [outsig, fc] = dau1996_preproc(insig,fs,...);
%
%   Input parameters:
%     insig  : input acoustic signal.
%     fs     : sampling rate.
%
%   *Warning:* This code is incorrect, the Dau 1996 models uses a
%   transmission-line model from Strube 1985, while this code erroneously
%   uses the gammatone filters. If/when the Strube model is included in
%   AMToolbox, this function will be fixed.
%
%   DAU1996_PREPROC(insig,fs) computes the internal representation of the
%   signal insig sampled with a frequency of fs Hz as described in Dau,
%   Puschel and Kohlrausch (1996a).
%  
%   [outsig,fc]=DAU1996_PREPROC(...) additionally returns the center frequencies of
%   the filter bank.
%
%   The Dau 1996 model consists of the following stages:
%   
%     1) a gammatone filter bank with 1-erb spaced filtes.
%
%     2) an envelope extraction stage done by half-wave rectification
%        followed by low-pass filtering to 1000 Hz.
%
%     3) an adaptation stage modelling nerve adaptation by a cascade of 5
%        loops.
%
%     4) a modulation low pass filter liming modulations to below 50 Hz.
%
%   Any of the optinal parameters for AUDITORYFILTERBANK, IHCENVELOPE
%   and ADAPTLOOP may be specified for this function. They will be passed
%   to the corresponding functions.
%
%   The model implemented in this file is not identical to the model
%   published in Dau et. al. (1996a). An overshoot limit has been added to
%   the adaptation stage to fix a problem where abrupt changes in the
%   input signal would cause unnaturally big responses. This is described
%   in Dau et. al. (1997a).
%
%   See also: auditoryfilterbank, ihcenvelope, adaptloop, dau1997_preproc
%
%   References:
%     T. Dau, D. Pueschel, and A. Kohlrausch. A quantitative model of the
%     effective signal processing in the auditory system. I. Model structure.
%     J. Acoust. Soc. Am., 99(6):3615-3622, 1996a.
%     
%     T. Dau, D. Pueschel, and A. Kohlrausch. A quantitative model of the
%     "effective" signal processing in the auditory system. II. Simulations
%     and measurements. J. Acoust. Soc. Am., 99:3623-3631, 1996b.
%     
%     T. Dau, B. Kollmeier, and A. Kohlrausch. Modeling auditory processing
%     of amplitude modulation. I. Detection and masking with narrow-band
%     carriers. J. Acoust. Soc. Am., 102:2892-2905, 1997a.
%     
%
%   Url: http://amtoolbox.sourceforge.net/amt-0.9.8/doc/modelstages/dau1996_preproc.php

% Copyright (C) 2009-2015 Piotr Majdak and Peter L. Søndergaard.
% This file is part of AMToolbox version 0.9.8
%
% 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/>.

%   AUTHOR : Torsten Dau, Morten Løve Jepsen, Peter L. Søndergaard

error(['This code of this function is incorrect. Please see the description ' ...
       'in the help text.']

% ------ Checking of input parameters ------------

if nargin<2
  error('%s: Too few input arguments.',upper(mfilename));
end;

if ~isnumeric(insig) 
  error('%s: insig must be numeric.',upper(mfilename));
end;

if ~isnumeric(fs) || ~isscalar(fs) || fs<=0
  error('%s: fs must be a positive scalar.',upper(mfilename));
end;

definput.import={'auditoryfilterbank','ihcenvelope','adaptloop'};
definput.importdefaults={'ihc_dau','adt_dau'};
definput.keyvals.subfs=[];

[flags,keyvals]  = ltfatarghelper({'flow','fhigh'},definput,varargin);

% ------ do the computation -------------------------

% Apply the auditory filterbank
[outsig, fc] = auditoryfilterbank(insig,fs,'argimport',flags,keyvals);

% 'haircell' envelope extraction
outsig = ihcenvelope(outsig,fs,'argimport',flags,keyvals);

% non-linear adaptation loops
outsig = adaptloop(outsig,fs,'argimport',flags,keyvals);

% Calculate filter coefficients for the 20 ms (approx.eq to 8 Hz) modulation
% lowpass filter.
% This filter places a pole /very/ close to the unit circle.
mlp_a = exp(-(1/0.02)/fs);
mlp_b = 1 - mlp_a;
mlp_a = [1, -mlp_a];

% Apply the low-pass modulation filter.
outsig = filter(mlp_b,mlp_a,outsig);

% Apply final resampling to avoid excessive data
if ~isempty(keyvals.subfs)
  outsig = fftresample(outsig,round(length(outsig)/fs*subfs));
end;