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Hello Everybody! It's my first post on this forum, can anyone help me? I'm newbie on DSP and I'm using TMS320C5402 DSK Board... I've used the codec.c digital loopback example to implement my own FIR filter as someone told already in a post from this forum. The problem it's that I think it's not working as it should... I'm going to attach the code for you guys to take a look... After the input sample is stored at "data" variable, I loop over the coefficients from matlab (multiplied by 32768) and convolved with the input samples. Afterwards the convolution is divided by 32768 to give me the real result (tried to <<32769 but didn't seem to work) and output that value to the codec... APARENTLY it filtered the sound... can't really because I don't have an oscilloscope or a function generator nearby to observe the shape of the waves. I say that because if I change the code and throw the input sample directly to the output, but leaving the filtering process (loops for convolution) unaltered, It should output the sound with no changes... but that doesn't happen, the sound gets a little distorced. (Can test by commenting the convolution loops) I believe that there's a timing problem on the code, but I don't know for sure, cause I've already read something about TMS320C5402 being used on video, and some heavy jobs... can't it support a simple FIR filter??? I've got also DSPLib and read SPRU518D on how to use it... but honestly... did anyone understant it? Example on Fir directory didn't help too much... actually didn't even work!? I'm studying Assembly in an attempt to optimize code, but it's very confuse... it's my first time to deal with Assembly... so imagine... If someone knows what's happening, could please help me? Alse if anyone has a WORKING Fir code for the TMS320C5402 could send a copy to me, please? Needed this working as soon as possible. Thanks a LOT for the attention... Good Day to all of you! |
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/*****************************************************************************/ /* Codec.c */ /* */ /* Digital Loopback example */ /* */ /*****************************************************************************/ #include <type.h> #include <board.h> #include <codec.h> #include <mcbsp54.h> /*****************************************************************************/ /* Function Prototypes */ /*****************************************************************************/ /* This delay routine does not conflict with DSP/BIOS. It is used in this */ /* example rather than brd_delay_msec which causes DSP/BIOS conflicts just */ /* because of this. If you are not using DSP/BIOS, you can change the code */ /* to use brd_delay_msec. */ void delay(s16 period); void move(int); /*****************************************************************************/ /* Global Variables */ /*****************************************************************************/ HANDLE hHandset; s16 data, out, sinal[100]; long conv; /* High-Pass Filter - Kaiser FIR fs = 16000 fc = [1000 1500] mags = [0 1] dev = [0.1 0.1] [n,wn,beta,ftype]=kaiserord(fc, mags, dev, fs) [b]=fir1(n,wn,ftype,kaiser(n+1,beta)) */ s16 coef[29] = { -421, -80, 338, 746, 1041, 1128, 938, 440, -345, -1346, -2450, -3519, -4411, -5001, 28120, -5001, -4411, -3519, -2450, -1346, -345, 440, 938, 1128, 1041, 746, 338, -80, -421}; /* Low-Pass Filter - Kaiser FIR fs = 16000 fc = [1000 1500] mags = [1 0] dev = [0.1 0.1] [n,wn,beta,ftype]=kaiserord(fc, mags, dev, fs) [b]=fir1(n,wn,ftype,kaiser(n+1,beta)) s16 coef[28] = { 284, -133, -588, -978, -1194, -1146, -779, -86, 883, 2028, 3210, 4278, 5086, 5520, 5520, 5086, 4278, 3210, 2028, 883, -86, -779, -1146, -1194, -978, -588, -133, 284}; */ /* Test-Mode for Loopback s16 coef[28] = { 32767, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}; */ int i, nsamples = 29; /* nsamples must be equal to number os coefficients of filter */ /*****************************************************************************/ /* MAIN */ /*****************************************************************************/ void main() { s16 cnt=2; if (brd_init(100)) return; /* blink the leds a couple times */ while ( cnt-- ) { brd_led_toggle(BRD_LED0); /* brd_delay_msec(1000); */ delay(1000); brd_led_toggle(BRD_LED1); /* brd_delay_msec(1000); */ delay(1000); brd_led_toggle(BRD_LED2); /* brd_delay_msec(1000); */ delay(1000); } /* Open Handset Codec */ hHandset = codec_open(HANDSET_CODEC); /* Acquire handle to codec */ /* Set codec parameters */ codec_dac_mode(hHandset, CODEC_DAC_15BIT); /* DAC in 15-bit mode */ codec_adc_mode(hHandset, CODEC_ADC_15BIT); /* ADC in 15-bit mode */ codec_ain_gain(hHandset, CODEC_AIN_6dB); /* 6dB gain on analog input to ADC */ codec_aout_gain(hHandset, CODEC_AOUT_MINUS_6dB); /* -6dB gain on analog output from DAC */ codec_sample_rate(hHandset,SR_16000); /* 16KHz sampling rate */ /* Polling and digital loopback */ while (1) { /* Wait for sample from handset */ while (!MCBSP_RRDY(HANDSET_CODEC)) {}; /* Read sample from and write back to handset codec */ data = *(volatile u16*)DRR1_ADDR(HANDSET_CODEC); /* ********** INIO DO FILTRO *********** */ sinal[nsamples-1] = data; conv = 0; for (i = 0; i < nsamples; i++) conv += sinal[nsamples-1-i] * coef[i]; move(nsamples); out = conv/32768; /* *********** FIM DO FILTRO *********** */ *(volatile u16*)DXR1_ADDR(HANDSET_CODEC) = out; } } void delay(s16 period) { int i, j; for(i = 0; i < period; i++) { for(j = 0; j < period >> 1; j++); } } void move (int n) { int cont; for (cont = 1; cont < n; cont ++) sinal[cont - 1] = sinal[cont]; } |
FIR timing problem on 5402???
Started by ●February 23, 2005
