FIR Filtering slight issue

Audio Plugins
1

Hi,

I’m implementing an FIR filter (should be super simple!). I’m writing my own because being the genius I am I couldn’t figure out how to use an array of arbitrary coefficients in the dsp FIR method. Either way, I just want something that works for the time being - I’m not too worried about efficiency.

With all that said, my method works ALMOST correctly but I get occasionally 1-sample-off glitches that are periodic, so I think the loops are wrong, or the way I’m wrapping the pointer to the current sample in the circular buffer isn’t quite right, but I can’t work out the exact issue. It’d be wonderful if someone would take a look and see if they notice the issue.

            float result = 0; // initialisation to 0 of the result

	// For each sample
	for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
	{

		bp = buffer_pointer[channel];  //bp is now the current index in the circular buffer

		delayLine[channel][buffer_pointer[channel]] = channelData[sample]; //insert current input into circular buffer

		result = 0;
		for (int i = 0; i < filter_length; i++) { // for each tap
			result = result + delayLine[channel][bp] * filterCoefs[i]; // multiply     
			bp--;
			if (bp < 0)
				bp += filter_length;
		}

		buffer_pointer[channel] = (buffer_pointer[channel] + 1) % filter_length; //update buffer pointer

		channelData[sample] = result;
	}

Where filter_length is the number of coefficients in the filter array.

Thanks for the help!