%% Introduction - Project, Stage 2, Option E

% Same as option D except the input .wav file can have a sampling rate of either 8kHz, 16kHz,
% 22.05kHz or 44.1kHz. Your system should convert the input (with known sampling rate) to a
% signal sampled at 16kHz and then pass it through the equaliser and output via
% speakers/headphones. You should ensure that any aliasing due to resampling is inaudible (to
% the lab demonstrators’ ears) while also ensuring that you do not audibly distort the signal of
% interest with your anti-aliasing filter.

% using filterDesigner


%% System I/O setup

deviceReader = dsp.AudioFileReader('Filename','PF_16000.wav','PlayCount',1,'SamplesPerFrame', 50);
deviceWriter = audioDeviceWriter('SampleRate',deviceReader.SampleRate);

% Butterworth filter paramaters
n = [10 10 10 10 10] % order of filters
ch_div = [0 500 1000 2000 4000 8000]; % freq divisions for equaliser
ch_div = ch_div * 2 / deviceReader.SampleRate;
ch_gain = [1 1 1 1 1]; % freq

lp1 = designfilt( 'lowpassfir', 'FilterOrder', n(1), 'CutoffFrequency', ch_div(2) , 'SampleRate', deviceReader.SampleRate);
bp1 = designfilt('bandpassfir', 'FilterOrder', n(2), 'CutoffFrequency1', ch_div(2), 'CutoffFrequency2', ch_div(3), 'SampleRate', deviceReader.SampleRate);
bp2 = designfilt('bandpassfir', 'FilterOrder', n(3), 'CutoffFrequency1', ch_div(3), 'CutoffFrequency2', ch_div(4), 'SampleRate', deviceReader.SampleRate);
bp3 = designfilt('bandpassfir', 'FilterOrder', n(4), 'CutoffFrequency1', ch_div(4), 'CutoffFrequency2', ch_div(5), 'SampleRate', deviceReader.SampleRate);
hp1 = designfilt('highpassfir', 'FilterOrder', n(5), 'CutoffFrequency', ch_div(6) , 'SampleRate', deviceReader.SampleRate);

%Fs = Fs_options(2);  % sampling rate in Hz (8Khz, 16kHz, 22.05kHz, 44.1Khz)

fileName = 'PF_16000.wav'; % name of file

% resample to 16Khz
switch deviceReader.SampleRate
    case 8000
        % Low pass filter
        % Interpolation
    case 22050
        % High pass filter
        % Decimation
    case 44100
        % High pass filter
        % Decimation
end

% anti-aliasing filter for down sampled signals (spectral components)

%% Main Loop

% Create slider
%sld = uicontrol('Style', 'slider', 'Min',1,'Max',50,'Value',41, 'Position', [400 20 120 20], 'Callback', ch_gain );

while ~isDone(deviceReader)
    mySignal = deviceReader();
    % resample
    % pass throgh Frequency scaling filter
    myProcessedSignal = filter(lp1,mySignal) + filter(bp1,mySignal) + filter(bp2,mySignal) + filter(bp3,mySignal) + filter(hp1,mySignal);
    % dtft on sample and display on equaliser
    %disp('Begin Signal Input...');
    deviceWriter(myProcessedSignal);
    %deviceWriter(mySignal);
end

release(deviceReader)
release(deviceWriter)