Basic FIR Filtering

Audio Plugins
8

Hello back. I have not forgot about you. Here it is my simple implementation of FIR filtering.
First I have defined some filter coefficients in header file of processor and delay line in length of number of coefficients.

 double filterCoefs[9] = { 0.0180524263969752,	0.0486396239380447,	0.122649723887587,
    		0.196847395914674,	0.227621659725439,	0.196847395914674,
    		0.122649723887587,	0.0486396239380447,	0.0180524263969752 };
 float delayLine[9] = {0, 0, 0, 0, 0, 0, 0, 0, 0};

After that, in cpp file in processBlock method:
AudioBuffer<FloatType> output = buffer;

for (int channel = 0; channel < getTotalNumInputChannels(); ++channel)
        		{

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

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

																			  // moving the samples in the delay line
				for (int i = 7; i >= 0; i--) { // delayLine being 9 values long
					delayLine[i + 1] = delayLine[i];
				}
				delayLine[0] = buffer.getSample(channel, sample);

				result = 0;
				for (int i = 0; i < 9; i++) { // for each tap
					result = result + delayLine[i] * filterCoefs[i]; // multiply     
				}
				output.setSample(channel, sample, result); // output
			}

		}

		buffer = output;
	}

Clearly it works for 9 coefficients only but it can easily be changed to different one, you will figure how easily. Try if that works and suites your needs, and if someone has some remarks please do say.

FIR Filter for decimation