olubern
Guest
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 Posted:
Sun Dec 04, 2005 5:15 pm Post subject:
real time FFT on c6713 DSK |
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I'm trying to do a 2048 pt real FFT using fft6713.c as my reference code.
When I tested this code with a CD music, (fs changed to 44.1KHz) I realiz
some of the samples from the ADC were overwritten while the FFT is bein
done (Obviously I can't do the FFT process in one sample time)
Is there a way to store samples from the ADC and do the FFT prcoessin
simultaneouly on the C6713 DSK.
The main file is posted below. I'll appreciate all the help I can get o
this.
Thanks in advance,
Bernard.
void main()
{
// Codec data handle structure
DSK6713_AIC23_CodecHandle hCodec;
for (i = 0 ; i<PTS ; i++) // set up twiddle constants in w
{
w[i].real = cos(2*PI*i/2048.0); //Re component of twiddle constants
w[i].imag =-sin(2*PI*i/2048.0); //Im component of twiddle constants
}
/* Initialize the board support library, must be called first */
DSK6713_init();
/* Start the codec */
hCodec = DSK6713_AIC23_openCodec(0, &config);
// Set sampling frequency via the number before KHZ in the define.
// Choose from 8, 16, 24, 32, 44.1, 48, or 96 Khz.
DSK6713_AIC23_setFreq(hCodec, DSK6713_AIC23_FREQ_8KHZ);
// Infinite loop so that we can continue to process samples forever
while(1)
{
while (!DSK6713_AIC23_read(hCodec, &xL));
while (!DSK6713_AIC23_read(hCodec, &xR));
//Input left channel sample from ADC
iobuffer[buffercount++] = (float)((short) xL);
//Use the next line to noise test the filter
//input = 0.5*((short) rand_int()); //scale by 1/2
if (buffercount >= PTS) //if iobuffer full
{
buffercount = 0; //reinit buffercount
flag = 1; //set flag
}
// When iobuffer is full fft the data after copying into a
// the complex samples[].real. Note we set the imaginary part
// equal 0, i.e., samples[i].imag = 0.0. You can view the FFT
// result by opening a CCS graph window on the array x or length
// PTS. AT this time PTS is set to 2048, but can be changed.
if (flag == 1) //wait until iobuffer is full
{
flag = 0; //reset flag
for (i = 0 ; i < PTS ; i++) //swap buffers
{
samples[i].real=iobuffer[i]; //buffer with new data
iobuffer[i] = x[i]; //processed frame to iobuffer
}
for (i = 0 ; i < PTS ; i++)
samples[i].imag = 0.0; //imag components = 0
FFT(samples,PTS); //call function FFT.c
for (i = 0 ; i < PTS ; i++) //compute magnitude
{
x[i] = sqrt(samples[i].real*samples[i].real
+ samples[i].imag*samples[i].imag)/32;
}
x[0] = 32000.0; //negative spike
} //end of FFT frame process
// The input samples are written to the output as a loop through,
// but when the FFT is performed (once per frame) there is a
// large delay in writing samples to the output, and fact
// consecutive input samples are dropped, thus making the IO
// non real-time. With interrupt processing, instead of this
// polling scheme, this could be avoided.
// Write a 32 bit signal sample to the serial port.
// The writing ping-pongs between left and right channels.
while (!DSK6713_AIC23_write(hCodec, xL));
while (!DSK6713_AIC23_write(hCodec, xR));
}
/* Close the codec in theory, but unreachable with the above whil
loop*/
//DSK6713_AIC23_closeCodec(hCodec);
} |
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Howard Long
Guest
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| Posted:
Mon Dec 05, 2005 1:35 pm Post subject:
Re: real time FFT on c6713 DSK |
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"olubern" <w101boa@wright.edu> wrote in message
news:442dnfqvgYImaw_enZ2dnUVZ_sqdnZ2d@giganews.com...
| Quote: | I'm trying to do a 2048 pt real FFT using fft6713.c as my reference code.
When I tested this code with a CD music, (fs changed to 44.1KHz) I realize
some of the samples from the ADC were overwritten while the FFT is being
done (Obviously I can't do the FFT process in one sample time)
Is there a way to store samples from the ADC and do the FFT prcoessing
simultaneouly on the C6713 DSK.
The main file is posted below. I'll appreciate all the help I can get on
this.
|
You can use an interrupt service routine on the MCBSP on Int11 to service
each ADC sample, storing them in a buffer. Once the buffer is full, the ISR
signals the main() process to say it's ready. Main() then processes the FFT.
One caveat: make sure that the buffer that's being processed in main() isn't
overwritten by the ISR. Either swap between two sets of buffers, or
immediately take a copy of the data in main() when it's been signalled.
Tachnically, double buffering would be the safest solution.
The problem with using an ISR, especially at higher sampling rates is that
the rate of interrupts is rather high.
As an alternative to a MCBSP ISR, the use of the EDMA module is preferred,
although it's a lot more complicated to achieve.
Rulph Chassaing's book "Digital Signal Processing and Applications with the
C6713 and C6416 DSK" carries the FastConvo example covering the interrupt
method.
You can modify the dsk_app DSK example to provide a similar result using the
EDMA module.
Regards, Howard |
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