FIR on ADSP-21160 | Analog Devices DSPs

Hi, guys. Hope you can help ne. I have a program to implement acouple of FIR filters on ADDS-21160 EzKit. When doing just the low pass filtering, everything is fine. But when I implement both (Low pass for a channel, Highpass for the other), nothing comes put from the codec (no sound at all). Could you help me to fix it, please? Here is the code: /* ADSP-21160 System Register bit definitions */ #include "def21160.h" #define TAPS 949 /* filter taps */ /*-------------------------------------------------------------------- -------*/ .segment /pm pm_data; .var LP_Coeff[TAPS] = ".\Source\LPCoeff.dat"; /* low pass filter coeffs */ .var HP_Coeff[TAPS] = ".\Source\HPCoeff.dat"; /* high pass filter coeffs */ .endseg; .segment /dm dm_data; .var LP_Delay[TAPS]; /* low pass filter delay line */ .var HP_Delay[TAPS]; /* high pass filter delay line */ .endseg; .segment /pm pm_code; .EXTERN RX_left_flag; .EXTERN Left_Channel_In; .EXTERN Right_Channel_In; .EXTERN Left_Channel_Out; .EXTERN Right_Channel_Out; .GLOBAL process_audio; .GLOBAL Filters_Init; /* insert DAG initialization code for delay or filter buffers here */ Filters_Init: // Initialize LP Filter b0 = LP_Delay; l0 = @LP_Delay; // Pointer to delay line m1 = 1; b8 = LP_Coeff; l8 = @LP_Coeff; // Pointer to coeffs m9 = 1; r12 = 0; // Clear delay line i3 = LP_Delay; l3 = 0; lcntr = TAPS, do clear_LP until lce; clear_LP: dm(i3, 1) = r12; // Initialize HP Filter b2 = HP_Delay; l2 = @HP_Delay; // Pointer to delay line m3 = 1; b10 = HP_Coeff; l10 = @HP_Coeff; // Pointer to coeffs m11 = 1; r14 = 0; // Clear delay line i3 = HP_Delay; l3 = 0; lcntr = TAPS, do clear_HP until lce; clear_HP: dm(i3, 1) = r12; rts; /* /////////////////////////////////////////////////////////////////// ///////////// * * audio processing routine * * //////////////////////////////////////////////////////////////////// //////////// */ process_audio: fir: /* process Left Channel Input */ r1 = -31; /* scale the sample to the range of +/-1.0 */ r0 = DM(Left_Channel_In); /* get left input sample */ f0 = float r0 by r1; /* convert to floating point */ r2 = DM(Right_Channel_In); /* get left input sample */ f2 = float r2 by r1; /* convert to floating point */ f8 = f0; if FLAG2_IN jump no_filter; /* in case we are in bypass mode, copy to output */ // Set to LP Filter b8 = LP_Coeff; l8 = @LP_Coeff; m3 = 1; b0 = LP_Delay; l0 = @LP_Delay; m11 = 1; do_LP_filtering: r12=r12 xor r12, dm(i0,m1)=f0; /* clear f12, put sample in delay line */ f8=pass f12, f0=dm(i0,m1), f4=pm(i8,m9); /* clear f8, get data & coef */ lcntr=TAPS-1, do LP_macloop until lce; LP_macloop: f12=f0*f4, f8=f8+f12, f0=dm(i0,m1), f4=pm(i8,m9); f12=f0*f4, f8=f8+f12; /* perform the last multiply */ f8=f8+f12; /* perform the last accumulate */ f9 = f8; // Switch to HP filter b8 = HP_Coeff; l8 = @HP_Coeff; m3 = 1; b0 = HP_Delay; l0 = @HP_Delay; m11 = 1; f8 = f2; f10 = f2; do_HP_filtering: r12=r12 xor r12, dm(i0,m1)=f0; /* clear f12, put sample in delay line */ f8=pass f12, f0=dm(i0,m1), f4=pm(i8,m9); /* clear f8, get data & coef */ lcntr=TAPS-1, do HP_macloop until lce; HP_macloop: f12=f0*f4, f8=f8+f12, f0=dm(i0,m1), f4=pm(i8,m9); f12=f0*f4, f8=f8+f12; /* perform the last multiply */ f8=f8+f12; /* perform the last accumulate */ exit_filter: r1 = 31; /* scale the result back up to MSBs */ r9 = fix f9 by r1; /* convert back to fixed point */ r8 = fix f8 by r1; DM(Left_Channel_Out) = r9; /* send result to AD1819 DACs */ DM(Right_Channel_Out) = r8; jump end_filter; no_filter: r1 = 31; /* scale the result back up to MSBs */ r0 = fix f0 by r1; /* convert back to fixed point */ r2 = fix f2 by r1; DM(Left_Channel_Out) = r0; /* send result to AD1819 DACs */ DM(Right_Channel_Out) = r2; end_filter: rts (db); r4 = 0; /* if selecting slower sample rates, this flag clear instruction is required */ dm(RX_left_flag) = r4; /* clear RX_left_flag since we have processed incoming data */ .endseg;

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FIR on ADSP-21160 - Author Unknown - Sep 28 1:03:00 2001