Hello
I haven't been following the post, but may be one possibility of your
'noise' is related to time-frequency uncertainity of the Short Time
Fourier Transform. Please take into account, the shorter the window,
you get better time resolution but your frequency resolution will
suffer. The longer the window, you get more frequency resolution but
your time resolution will suffer. So may be give it a try with
different windows size, and you can get more nice results, depending on
your application.

Regards

Juan Pablo Narino


Rune Allnor ha escrito:

> Some nit-picking in your program:
>
> louis skrev:
>
> > 	//******* Create Hann Window ****************//
> > 	for (k = 0; k < fftsize; k++)
> > 	{
> > 		hanning[k] = 0.5 - 0.5 * cos(6.2832 * k / fftsize);
> > 	}
>
> Use a proper represntation of pi. There is a defined constant
> in math.h. I think it is called M_PI. Check with the docs.
>
> Also, check if the Hann window function is correct. I think it should
> be something like 
> 
> cos(M_PI*k/(fftsize -1));
> 
> Rune

Some nit-picking in your program:

louis skrev:

> 	//******* Create Hann Window ****************//
> 	for (k = 0; k < fftsize; k++)
> 	{
> 		hanning[k] = 0.5 - 0.5 * cos(6.2832 * k / fftsize);
> 	}

Use a proper represntation of pi. There is a defined constant
in math.h. I think it is called M_PI. Check with the docs.

Also, check if the Hann window function is correct. I think it should
be something like 

cos(M_PI*k/(fftsize -1));

Rune

louis skrev:
> Hi everyone,
>
> Okay so I just plotted the spectrogram of the superposition of two pure
> tones - a 1khz and a 3 khz, using my FFT/spectrogram code..
>
> I'm not evne sure if its right - it looks right I see two lines - one at
> the 3khz makr and the other at the 1khz mark which go across, however they
> seem to be fading in and out at the edges.. It doesnt look bad but i'm just
> wondering if this is accurate for the case of 2 pure tones..?? shoudlnt it
> be just a solid line , or is htis the effect of the Hann window??

It is not an effect of the Hann window, it might be an effect of
partially filled 
buffers at the ends. 

Rune

Thanks for your response, you're totally right. I changed that but I still
see the same spectrogram :s

btw I made a mistake when posting my code, below "wallbins" should be just
"bins" so:

"//-----------Remove frequencies less than 40 Hz -------//

frequencyResolution = (float)samplerate/(float)fftsize;
bins = ceil(filter/frequencyResolution);

for( j=0; j<=bins; j++ )         //<-----------
{
assorted_output[ fftsize/2 - j ] = 0;
assorted_output[ fftsize/2 + j ] = 0;
}"

not a huge deal but just in case there was confusion.

louis wrote:

[...]
> 		//------------------ Perform FFT -----------------------//
> 		kiss_fftr( kiss_fftr_state, rin, sout );
> 
> 		//------------------ Compute Magnitude -----------------//
> 		for( s=0; s<fftsize; s++ )
> 		{
> 			output_frame[s] = powf( sout[s].r , 2.0 ) + powf( sout[s].i , 2.0 );
> 		}

The output of the real forward fft is roughly half the FFT [0:pi].  The
transform is conjugate symmetric.  The second half can be easily
inferred from the first half.

So your frequency domain calculations should be using (fftsize/2+1)
complex pairs, not fftsize.

-- 
Mark Borgerding

btw.. "loop" is useless in the code i just posted, that was just for output
purposes.

So just to clarify, that code i posted basically creates a power spectra
at each "lp" value. Then I have other code which combines all the
powerspectra to make the spectrogram..

I can't figure out why its so noisy.. the audio doesn't sound noisy at
all.. but my spectrogram shows otherwise.. :(

>Hey Louis,
>
>You're welcome! However, I feel like I could help you more (and there's
>probably others here too) if you would include the entire listing
instead
>of a piece. For example, I don't know what data types you're using for 
>s, fftsize, etc. I also have no idea what the lp++ is for at the end
>of the loop. Also, what are you using input_window[] for? It looks 
>useless from the code you've provided.
>
>Your basic approach seems sound - there may be a problem in the coding,
>or in your understanding of what to expect. For the latter, I
>suggest you use something like octave (http://www.octave.org) to 
>take the fft of your input data and plot the resulting magnitude
>squared.


Hi Randy, Thanks so much for your help! (I can't believe I'm having so
much trouble getting a silly spectrogram!! argh!) 
Here is my full code programmed in VC++ using a current version of
kissfft:


#include <iostream>
#include <fstream>
#include <string>
using namespace std;

#include <math.h>
#include <stdlib.h>
#include "_kiss_fft_guts.h"
#include "kiss_fft.h"
#include "kiss_fftr.h"

const int fftsize = 1024;
const int hopsize = 512;
// buffsize hardcoded for now: 10 audio cycles x 44100 sampling rate x
0.92 seconds per audio cycle 
const int buffsize = 405720;	
const float filter=40;  // all values less than 40 Hz should be set to 0.
const int samplerate= 44100;

float hanning[fftsize];
float input_window[fftsize];
float output_frame[fftsize];
float assorted_output[fftsize];
float audiobuffer[buffsize];
kiss_fft_scalar rin[fftsize+2];
kiss_fft_cpx sout[fftsize];    

int main()
{
	kiss_fftr_cfg  kiss_fftr_state;
	int temp, loops, i, j, k, lp, s, index, bins; 
	char outfilename[128];
	float frequencyResolution;
	
        ifstream audiofile;
	ofstream powerspectfile;
	
	kiss_fftr_state = kiss_fftr_alloc(fftsize,0,0,0);
	kiss_fft_scalar zero;
        memset(&zero,0,sizeof(zero) );	
    
        filename = "audio.txt";
	audiofile.open( filename, ios::in );
	
	//******* Create Hann Window ****************//
	for (k = 0; k < fftsize; k++)
	{
		hanning[k] = 0.5 - 0.5 * cos(6.2832 * k / fftsize);
	}
	
 
	
	//****** Read-in samples from ascii audio file ******//
	
	for( i=0; i<buffsize; i++ )
	{
		audiofile>>audiobuffer[i];
	}
	audiofile.close();

	loops = buffsize/hopsize;
	lp = 0;

	//********Loop FFT **************************//
	
	for( index=0; index<buffsize-fftsize; index+=hopsize )
	{
		//-------------- Apply the Hann window -----------------//
		for( s=0; s<fftsize; s++ )
		{
			input_window[s] = audiobuffer[index+s]*hanning[s];
			rin[s] = input_window[s];				
		}

		//------------------ Perform FFT -----------------------//
		kiss_fftr( kiss_fftr_state, rin, sout );

		//------------------ Compute Magnitude -----------------//
		for( s=0; s<fftsize; s++ )
		{
			output_frame[s] = powf( sout[s].r , 2.0 ) + powf( sout[s].i , 2.0 );
		}
		
		//----------------- Assort power values ----------------//
		for( s=0; s<fftsize/2; s++ )
		{
			assorted_output[s+fftsize/2] = output_frame[s];
		}
		
		for( s=fftsize-1; s>=fftsize/2; s-- )
		{
			assorted_output[s-fftsize/2] = output_frame[s];
		}

		//-----------Remove frequencies less than 40 Hz -------//
		
		frequencyResolution = (float)samplerate/(float)fftsize;
		bins = ceil(filter/frequencyResolution);
		
		for( j=0; j<=wallbins; j++ )
		{
			assorted_output[ fftsize/2 - j ] = 0;
			assorted_output[ fftsize/2 + j ] = 0;
		}
		
		
		//------------------ Write out to file -----------------//
		sprintf( outfilename, "%s.powspec%5.5d", "cca", lp );
		powerspectfile.open(outfilename, ios::out);
		for ( j = 0; j < fftsize; j++ )
		{	
			powerspectfile<<assorted_output[j]<<endl;
		}
		powerspectfile.close();
		
		//------------------ Increment loop -------------------//
		lp++;
	}
	
	free(kiss_fft_state);
        free(kiss_fftr_state);

	return 0;
}

"louis" <lost_bits1110@hotmail.com> writes:

> Thanks so much, you're right! I added that '-1'.. however the output still
> seems wrong.. :S
> I guess I'll have to try fftw or something (though I'm sure there
> shouldn't be anything wrong with kissfft??) but i guess with programming
> you can never be 100% sure.. sigh.

Hey Louis,

You're welcome! However, I feel like I could help you more (and there's
probably others here too) if you would include the entire listing instead
of a piece. For example, I don't know what data types you're using for 
s, fftsize, etc. I also have no idea what the lp++ is for at the end
of the loop. Also, what are you using input_window[] for? It looks 
useless from the code you've provided.

Your basic approach seems sound - there may be a problem in the coding,
or in your understanding of what to expect. For the latter, I
suggest you use something like octave (http://www.octave.org) to 
take the fft of your input data and plot the resulting magnitude
squared.

--Randy

>
>>"louis" <lost_bits1110@hotmail.com> writes:
>>
>>> 	for( s=fftsize; s>=fftsize/2; s-- )
>>> 	{
>>> 		assorted_output[s-fftsize/2] = output_frame[s];
>>> 	}
>>
>>I haven't been following the thread, but in any case you
>>have an error here. Should be
>>
>> 	for( s=fftsize-1; s>=fftsize/2; s-- )
>> 	{
>> 		assorted_output[s-fftsize/2] = output_frame[s];
>>   }
>
>
>

-- 
%  Randy Yates                  % "Rollin' and riding and slippin' and
%% Fuquay-Varina, NC            %  sliding, it's magic."
%%% 919-577-9882                %  
%%%% <yates@ieee.org>           % 'Living' Thing', *A New World Record*, ELO
http://home.earthlink.net/~yatescr

Thanks so much, you're right! I added that '-1'.. however the output still
seems wrong.. :S
I guess I'll have to try fftw or something (though I'm sure there
shouldn't be anything wrong with kissfft??) but i guess with programming
you can never be 100% sure.. sigh.

>"louis" <lost_bits1110@hotmail.com> writes:
>
>> 	for( s=fftsize; s>=fftsize/2; s-- )
>> 	{
>> 		assorted_output[s-fftsize/2] = output_frame[s];
>> 	}
>
>I haven't been following the thread, but in any case you
>have an error here. Should be
>
> 	for( s=fftsize-1; s>=fftsize/2; s-- )
> 	{
> 		assorted_output[s-fftsize/2] = output_frame[s];
>   }

"louis" <lost_bits1110@hotmail.com> writes:

> 	for( s=fftsize; s>=fftsize/2; s-- )
> 	{
> 		assorted_output[s-fftsize/2] = output_frame[s];
> 	}

I haven't been following the thread, but in any case you
have an error here. Should be

 	for( s=fftsize-1; s>=fftsize/2; s-- )
 	{
 		assorted_output[s-fftsize/2] = output_frame[s];
   }
-- 
%  Randy Yates                  % "So now it's getting late,
%% Fuquay-Varina, NC            %    and those who hesitate
%%% 919-577-9882                %    got no one..."
%%%% <yates@ieee.org>           % 'Waterfall', *Face The Music*, ELO
http://home.earthlink.net/~yatescr