
From:  Rick T 
Subject:  Re: How can I increase/decrease (frequency/pitch) and phase using fft/ifft tia sal22 
Date:  Thu, 7 Apr 2011 04:43:55 1000 
%resample_repmat signal
clear all, clf
Fs = 1000; % Sampling rate
Ts = 1/Fs; %sampling interval
t=0:Ts:1Ts; %sampling period
freq_orig=1;
y=sin(2*pi*t*freq_orig)'; %gives a short wave
freq_new=9;
y2=resample(y,1,freq_new); %resample matrix
y3=repmat (y2,freq_new,1); %replicate matrix
[r_orig,c_orig] = size(y) %get orig number of rows and cols
[r_new,c_new] = size(y3) %get new number of rows and cols
subplot(2,1,1),plot(y),title('Orginal signal')
title(['rows=',num2str(r_orig),' cols=',num2str(c_orig)])
subplot(2,1,2),plot(y3),title('New signal')
title(['rows=',num2str(r_new),' cols=',num2str(c_new)])
Subject: Re: How can I increase/decrease (frequency/pitch) and phase using fft/ifft tia sal22
To: "Sergei Steshenko" <address@hidden>
Cc: address@hidden
Date: Monday, April 4, 2011, 3:37 PM
Sorry, but this is mostly nonsense. FFT can't be faster than just moving
I need fft/ifft due to the fact that I have to alter various cells in the array the signal is stored in (in the frequency domain. The script is very long. In my experience if you post hundreds of lines of code people will not look at it. That's why I kept it simple and basic. And asked How can I increase/decrease (frequency/pitch) and phase using fft/ifft.
PS: Unfortunately Sergei the nice solution you sent won't work, I'm dealing with large arrays that are exported back out as audio files and fft/ifft seems to be the fastest.
On Mon, Apr 4, 2011 at 11:33 AM, Sergei Steshenko <address@hidden> wrote:
 On Mon, 4/4/11, Rick T <address@hidden> wrote:
From: Rick T <address@hidden>
Subject: How can I increase/decrease (frequency/pitch) and phase using fft/ifft tia sal22
To: address@hidden
Date: Monday, April 4, 2011, 2:11 PM
How can I increase/decrease (frequency/pitch) and phase using fft/ifft
I think I have the basic code but I’m not sure what to do next
PS: Thanks for the help on the last question everyone I decided not to use the FOR loop and sin/cos values and just use fft/ifft
to see if this will work.
Example I have a signal that repeats 1 time every second and I want to
have it repeat 3 times a second instead.
%Voiceprint raise lower freq phase conjugate signal
tic
clear all, clc,clf,tic
%% Sound /beep calculation complete
filerawbeepStr='calculations_complete.wav';
filerawbeeppathStr='/home/rat/Documents/octave/raw/';
filevoiceprepathStr='/home/rat/Documents/octave/eq_research/main/
transform/voice/';
filewavpathStr='/home/rat/Documents/octave/eq_research/main/transform/
wav/';
[ybeep, Fsbeep, nbitsbeep] =
wavread(strcat(filerawbeeppathStr,filerawbeepStr));
%addpath(”/home/rat/Documents/octave/eq_research/main/transform/”);
%add path to location of functions
%1a voice print import
[vp_sig_orig, fs_rate, nbitsraw] =
wavread(strcat(filevoiceprepathStr,'voice8000fs.wav'));
%vp_sig_orig=vp_sig_orig’;
vp_sig_len=length(vp_sig_orig);
%2a create frequency domain
ya_fft = fft(vp_sig_orig);
vp_sig_phase_orig = unwrap(angle(ya_fft));
%get Magnitude
ya_fft_mag = abs(ya_fft);
%3a frequency back to time domain
ya_ifft=real(ifft(ya_fft));
%adjust frequency/phase here? How?
vp_sig_new=real(ifft(ya_fft_mag.*exp(i*vp_sig_phase_orig)));
subplot(3,1,1), plot(vp_sig_orig),title('1 original time domain')
subplot(3,1,2), plot(ya_ifft),title('2 rebuild time domain')
subplot(3,1,3), plot(vp_sig_new),title('3 adjusted time')
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So, regarding your
"
Example I have a signal that repeats 1 time every second and I want to
have it repeat 3 times a second instead.
"
 why do you need FFT in the first place ?
I.e. if 'x' is your signal, why not simply write
x = linspace(0, 1, 10); % or whatever other way to generate your signal
y = [x x x]; % repeat x 3 times
plot(y);
?
Regards,
Sergei.

data in memory  the latter is my solution.
If your audio comes in frequency domain, then by just _one_ 'ifft' you
convert it into time domain, and then my solution works.
You have already been given a link to 'resample' function.
You appear not to understand fundamental things regarding pitch shift. If
your signal gets repeated a number of times, it is not pitch shift.
Pitch shift does not imply signal repetitions and does not imply change
of number of output samples.
AFAIK pitch shift is implemented through overlapping relatively (compared
to the length of the whole musical piece) short FFTs, and the spectrum is
shifted (rather, scaled  you typically need all spectral componenets to be multiplied by the same factor) in order to achieve pitch shift  number of samples, as I said, does _not_ change.
Start from http://en.wikipedia.org/wiki/Audio_timescalepitch_modification > http://en.wikipedia.org/wiki/Audio_timescalepitch_modification#Pitch_scaling .
Regards,
Sergei.
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