how do i filter a modulated signal using sinc

how do i filter a modulated signal using sinc function of 40k Hz frequency with sampling rate of 705.6 kHz. so i can eventually demodulate the signal and listen to the audio. Any help is appreciated. thanks.

%Set cutoff frequency and delay of the impulse response peak

Fc = 40e3;

tau = 142e-8;

%Two-sided bandwidth of the lowpass filter

B = 2*Fc;

%Set up time array and sampling variables for the "ideal" lowpass filter

%Impulse response length >> 1/B to approximate the ideal filter

time1 = linspace(-10/B,10/B,4000)';

dt1 = time1(2) - time1(1); %Sampling interval for ideal filter time array

Fs1 = 1/dt1; %Sampling frequency

dt2 = 1/(10*B); %Sampling interval for practical filter time array

time2 = (0:dt2:(2*tau))'; %Time array for the practical filter

Fs2 = 1/dt2; %Sampling frequency

N = length(time2); %Number of samples in practical filter

%Compute impulse responses for ideal and practical filters

ht_ideal = B*sinc(B*time1);

% ht_windowed = B*sinc(B*(time2 - tau)).*ones(N,1); %with rectangular window

ht_windowed = B*sinc(B*(time2 - tau)).*hanning(N); %with hanning window

figure subplot(2,2,1)

plot(time1,ht_ideal)

title('ideal low pass filter');

hold on subplot(2,2,2) plot(time2,ht_windowed)

title('practice low pass filter'); hold on

fid = fopen('C:\Users\birbahadur\Desktop\fall2016\signal and system\project1\dsbsc.bin','rb');

a = fread(fid,'float32');

fclose(fid);

iend=length(a);

subplot(2,2,3);

plot(a);

title('input signal');

hold on

allsongs=ifftshift(ifft(a));

tsong=allsongs(1:12:705600);

amsong1=conv(tsong,ht_ideal);

%soundsc(abs(song1)*100)

%z = amdemod(y,Fc,Fs)

song1 = amdemod(abs(amsong1),40000,705600);

subplot(2,2,4);

plot(abs(allsongs));

%soundsc(y,Fs,nBits)

soundsc(song1)