TMDSLCDK6748: How to include previous samples for FIR filter DSPF_sp_fir_gen()

Part Number:TMDSLCDK6748

Hi all,

Can you help me figure out how to include previous samples in FIR filter. According to the TMS320C67x DSP Library Programmers Ref. Guide SPRU657C, the input array starts at x(-nh+1) and ends at x(nr-1).

To test the filter, I'm using the mcaspPlayBk.c from the StarterWare C6748_StarterWare_1_20_04_01. 

To construct the input array for the FIR filter, I'm increasing the buffer size from its default of 2000 samples to 2015 for a 16-tap filter. The problem I have (I think) is that I don't know how to insert (nh-1) previous samples to the head of buffer. Since the data comes in frames, I'm trying to save the last (nh-1) sample from the previous frame then using them in the "current" frame. But this isn't working.

Here is my code, which is modified version of mcaspPlayBk.c:

    while(1)
    {
        if(lastFullRxBuf != lastSentTxBuf)
        {
            /*
            ** Start the transmission from the link paramset. The param set
            ** 1 will be linked to param set at PAR_TX_START. So do not
            ** update paRAM set1.
            */
            parToSend =  PAR_TX_START + (parOffTxToSend % NUM_PAR);
            parOffTxToSend = (parOffTxToSend + 1) % NUM_PAR;
            parToLink  = PAR_TX_START + parOffTxToSend;

            lastSentTxBuf = (lastSentTxBuf + 1) % NUM_BUF;

            // is this necessary?
            memcpy(tempCharBuf, (void *)rxBufPtr[lastFullRxBuf], AUDIO_BUF_SIZE);
            /*
            **convert char array to short signed int array starting at NH
            */
            unpackMcASPBuf(tempCharBuf, NUM_SAMPLES_PER_AUDIO_BUF, NH, singleChannelInt);
            counter++; //for debugging
            /*
            ** insert previous samples to front of rx frame buffer for FIR filter
            */
            insertPrevSamples(singleChannelInt, previousSamples, NH);
            /*
            ** save trailing NH samples for next frame
            */
            saveTrailingSamples(singleChannelInt, NUM_SAMPLES_PER_AUDIO_BUF, previousSamples, NH);
            /*
            ** convert to sp float for FIR filter
            */
            intToFloat(singleChannelInt, NUM_SAMPLES_PER_AUDIO_BUF + NH, DSPInputSignal);
            /*
            **FIR filter
            */
            DSPF_sp_fir_gen(DSPInputSignal, filterCoefs, DSPOutputSignal, NH, NUM_SAMPLES_PER_AUDIO_BUF );
            /*
            ** convert back to int for txBuf
            */
            floatToInt(DSPOutputSignal, NUM_SAMPLES_PER_AUDIO_BUF, tempIntBuf);
            /*
            ** interleave back into txBuf short int to char conversion
            */
            packMcASPBuf(tempIntBuf, NUM_SAMPLES_PER_AUDIO_BUF, tempCharBuf);
            memcpy((void *)txBufPtr[lastSentTxBuf], tempCharBuf, AUDIO_BUF_SIZE);
            /*
            ** Send the buffer by setting the DMA params accordingly.
            ** Here the buffer to send and number of samples are passed as
            ** parameters. This is important, if only transmit section
            ** is to be used.
            */
            BufferTxDMAActivate(lastSentTxBuf, NUM_SAMPLES_PER_AUDIO_BUF,
      }
}

The functions are:

void insertPrevSamples(short signed int* x, short signed int* p, int np){ //p = previous
        int i=0;
        for(i= 0; i < np - 1 ; ++i)
                x[i] = p[i];
}

void saveTrailingSamples(short signed int* x, int nx, short signed int* p, int np){ //p = previous
        int i=0;
        for(i= 0; i < np - 1; i++)
                p[i] = x[nx + i];
}

void unpackMcASPBuf(unsigned char* x, int nx, int nh, short signed int* y){
    int i=0;
    int indx=0;
    for(i = nh; i < nx + nh; i++){
        y[i]  = x[indx+1] << 8;
        y[i] |= x[indx+0] << 0;
        indx += 8;
    }
}
void packMcASPBuf(short int* in, int size, unsigned char* out){
    int i=0;
    int indx=0;
    int byte0;
    int byte1;
    int byte4;
    int byte5;

    for(i=0; i< size/2; i++){
        out[indx+2] = 0;
        out[indx+3] = 0;
        out[indx+6] = 0;
        out[indx+7] = 0;

        byte0 = ( in[i] >> 0 );
        byte1 = ( in[i] >> 8 );
        byte4 = ( in[i] >> 16 );
        byte5 = ( in[i] >> 24 );

        out[indx+0] = (char)byte0;
        out[indx+1] = (char)byte1;

        out[indx+4] = (char)byte4;
        out[indx+5] = (char)byte5;
        indx += 8;
    }
}

Also included is a waveform showing the corrupted nh-1 samples at the head of the buffer. 

I hope this makes sense and sorry for the lengthy explanation and code.

thank you,

Scott