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comment
Save and load WAV sound files. Analyze frequencies.
{w,rate}=loadwave(filename); loads a WAV file.
w=loadwave(filename); loads a WAV file. r is saved to defaultrate.
savewave(filename,v[,rate,bits]); saves a WAV file.
analyze(v[,fmin,fmax,rate,points]); plots the frequencies of v
mapsound(w,[,dt,fmin,fmax,simpl,rate]); show the sound history
soundsec(n[,rate]); generates t values for n seconds.
Default sample rate is 22050 or the sample rate of the last read WAV file.
endcomment
defaultrate=22050;
function soundsec (n,rate=0)
## Return n seconds of t parameter.
global defaultrate;
if rate==0; rate=defaultrate; endif;
return 0:2*pi/rate:2*n*pi;
endfunction
function savewave (filename,v,rate=0,bits=16)
## Save a vector of sound data in WAV format.
## Return the length of the sample in seconds.
## If rate=0, the default sampling rate is taken.
## bits may be 8 or 16.
global defaultrate;
if rate==0; rate=defaultrate; endif;
open(filename,"wb");
write("RIFF");
if bits==8;
n=cols(v);
putlongword(n+37);
write("WAVEfmt ");
putlongword(16); putword(1); putword(1);
putlongword(rate); putlongword(rate);
putword(1); putword(8);
write("data");
putlongword(n);
m=max(abs(v)); vv=v/m*64+128;
putuchar(vv);
putchar(0);
elseif bits==16;
n=cols(v);
putlongword(2*n+36);
write("WAVEfmt ");
putlongword(16); putword(1); putword(1);
putlongword(rate); putlongword(2*rate);
putword(2); putword(16);
write("data");
putlongword(2*n);
m=max(abs(v)); vv=v/m*64*256;
putword(vv);
else error("Bits must be 8 or 16");
endif
close();
return cols(v)/rate;
endfunction
function loadwave (filename)
## Read a WAV file.
## The sample rate is stored to defaultrate.
global defaultrate;
open(filename,"rb");
if getstring(4)<>"RIFF"; error("No Wave file!"); endif;
getlongword(); .. This is the total file length minus 8
if getstring(8)<>"WAVEfmt "; error("No Wave file!"); endif;
offset=getlongword(); .. ??? Is always 16, sometimes 18
getword(); .. ??? Is always 1.
if (getword()<>1); error("Stereo sample!"); endif;
rate=getlongword(); getlongword();
byte=getword();
bits=getword();
if offset>16; getuchar(offset-16); endif;
if getstring(4)<>"data"; error("No Wave file!"); endif;
if byte==1;
w=getuchar(getlongword());
elseif byte==2;
w=getword(getlongword()/2);
else error("Not 8 or 16 bit!");
endif;
if cols(w)>0; w=w-sum(w)/cols(w); w=w/max(abs(w)); endif;
close();
defaultrate=rate;
return w;
endfunction
function analyze (w,fmin=10,fmax=1500,rate=0,points=8192)
## Make a frequency plot of the signal w with sampling rate.
## The data must be at least points long.
## The maximal frequency plotted will be fmax, the minimal fmin.
global defaultrate;
if rate==0; rate=defaultrate; endif;
v=w[1:points];
f=abs(fft(v));
i=fmin/rate*points:fmax/rate*points;
setplot(fmin,fmax,0,max(f[i]));
fr=i/points*rate;
xplot(fr,f[i]);
return "";
endfunction
function mapsound (w,dt=0.1,fmin=100,fmax=1500,simpl=1,rate=0)
## Plots a sound map for a sound.
## It does FFT at time increments dt.
## rate is the sampling rate.
## simpl points are collected for speed reasons.
n=cols(w);
global defaultrate;
if rate==0; rate=defaultrate; endif;
dp=dt*rate;
points=2^floor(log(dp)/log(2));
ind=fmin/rate*points/simpl:fmax/rate*points/simpl;
f=abs(fft(w[1:points]));
f=sum(redim(f,points/simpl,simpl))';
M=f[ind];
i=1;
repeat
i=i+dp;
if i+points>n; break; endif;
f=abs(fft(w[i:i+points-1]));
f=sum(redim(f,points/simpl,simpl))';
M=M_f[ind];
end;
setplot(fmin,fmax,0,n/rate);
density(-M,0.99);
xplot();
return ""
endfunction
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