matthiasm@0: function [f_audio_out,timepositions_afterDegr] = degradationUnit_applyAliasing(f_audio, samplingFreq, timepositions_beforeDegr, parameter)
matthiasm@0: %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
matthiasm@0: % Name: degradation_applyAliasing
matthiasm@0: % Version: 1
matthiasm@0: % Date: 2013-01-23
matthiasm@0: % Programmer: Matthias Mauch
matthiasm@0: %
matthiasm@0: % Description:
matthiasm@0: % - downsamples without filtering, then upsamples again using S&H
matthiasm@0: %
matthiasm@0: % Input:
matthiasm@0: %   f_audio      - audio signal \in [-1,1]^{NxC} with C being the number of
matthiasm@0: %                  channels
matthiasm@0: %   samplingFreq - sampling frequency of f_audio
matthiasm@0: %   timepositions_beforeDegr - some degradations delay the input signal. If
matthiasm@0: %                             some points in time are given via this
matthiasm@0: %                             parameter, timepositions_afterDegr will
matthiasm@0: %                             return the corresponding positions in the
matthiasm@0: %                             output. Set to [] if unavailable. Set f_audio
matthiasm@0: %                             and samplingFreq to [] to compute only
matthiasm@0: %                             timepositions_afterDegr.
matthiasm@0: %
matthiasm@0: % Input (optional): parameter
matthiasm@0: %   .dsFrequency = 8000 - destination sampling frequency
matthiasm@0: %   .normalizeOutputAudio = 1 - peak normalize output audio
matthiasm@0: %
matthiasm@0: % Output:
matthiasm@0: %   f_audio      - audio output signal
matthiasm@0: %
matthiasm@0: %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
matthiasm@0: 
matthiasm@0: %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
matthiasm@0: % Audio Degradation Toolbox
matthiasm@0: %
matthiasm@0: % Centre for Digital Music, Queen Mary University of London.
matthiasm@0: % This file copyright 2013 Sebastian Ewert, Matthias Mauch and QMUL.
matthiasm@0: %    
matthiasm@0: % This program is free software; you can redistribute it and/or
matthiasm@0: % modify it under the terms of the GNU General Public License as
matthiasm@0: % published by the Free Software Foundation; either version 2 of the
matthiasm@0: % License, or (at your option) any later version.  See the file
matthiasm@0: % COPYING included with this distribution for more information.
matthiasm@0: %
matthiasm@0: %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
matthiasm@0: 
matthiasm@0: %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
matthiasm@0: % Check parameters
matthiasm@0: %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
matthiasm@0: if nargin<4
matthiasm@0:     parameter=[];
matthiasm@0: end
matthiasm@0: if nargin<3
matthiasm@0:     timepositions_beforeDegr=[];
matthiasm@0: end
matthiasm@0: if nargin<2
matthiasm@0:     error('Please specify input data');
matthiasm@0: end
matthiasm@0: 
matthiasm@0: if isfield(parameter,'dsFrequency')==0
matthiasm@0:     parameter.dsFrequency = 8000;
matthiasm@0: end
matthiasm@0: if isfield(parameter,'normalizeOutputAudio')==0
matthiasm@0:     parameter.normalizeOutputAudio = 1;
matthiasm@0: end
matthiasm@0: 
matthiasm@0: %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
matthiasm@0: % Main program
matthiasm@0: %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
matthiasm@0: 
matthiasm@0: f_audio_out = [];
matthiasm@0: if ~isempty(f_audio)
matthiasm@0:     
matthiasm@0:     % We cannot employ resample as it always imposes a lowpass pass filter to
matthiasm@0:     % counter aliasing
matthiasm@0:     
matthiasm@0:     nSample = size(f_audio, 1);
matthiasm@0:     nSampleNew = round(nSample / samplingFreq * parameter.dsFrequency);
matthiasm@0:     
matthiasm@0:     tOld = (0:(nSample-1))/samplingFreq;
matthiasm@0:     tNew = (0:(nSampleNew-1))/parameter.dsFrequency;
matthiasm@0:     
matthiasm@0:     temp = interp1(tOld, f_audio, tNew, 'nearest');
matthiasm@0:     f_audio_out = resample(temp, samplingFreq, parameter.dsFrequency);
matthiasm@0:     
matthiasm@0:     if parameter.normalizeOutputAudio
matthiasm@0:         f_audio_out = adthelper_normalizeAudio(f_audio_out, samplingFreq);
matthiasm@0:     end
matthiasm@0: end
matthiasm@0: 
matthiasm@0: % This degradation does not impose a delay
matthiasm@0: timepositions_afterDegr = timepositions_beforeDegr;
matthiasm@0: 
matthiasm@0: end