FIR direct

FIR Filter Implementation Using Octave GNU Tool and C Language

1. Abbreviation

FIR – Finite Impulse Response
IIR – Infinite Impulse Response
DFT – Discrete Fourier Transformation
IDFT – Inverse Discrete Fourier Transformation
FFT – Fast Fourier Transformation
ADC – Analog Digital Converter

2. Introduction

A FIR filter is a linear digital filter with a finite impulse response and whose z-transform has only nulls. Such a filter is unrealizable in analog technology, which makes it unique in the digital field. The advantage of the FIR filter is always stable and easy to implement with a linear phase response.
Referring to
[1]: L. Rabiner, B. Gold ” Theory and application of digital signal processing“
[2]: A. Oppenheim, R. Schafer ” Discrete-Time Signal Processing“,
here I will give the main properties of the FIR filter, which we will use later.
  1. The z-transform of the FIR filter has the form:
    FIR formula1
    where h(0), h(1), h(2), …, h(N-1) – the pulse response of this filter with a N length.
  2. Conditions for the linearity of the phase response of the FIR filter:
    h(k) = h(N-1-k) where 0 ≤ k ≤ N-1 (2)
    h(k) = – h(N-1-k) where 0 ≤ k ≤ N-1 (3)
    For example, (2) is the pulse response of a low-pass filter, and (3) is the pulse response of a high-pass filter, bandpass filter, or differentiator.
  3. FIR filter delay with linear phase:
    FIR formula4
  4. The nulls of the FIR filter with a linear phase response locate on the unit circle or must satisfy the inversion condition:
    FIR formula5 1
    is a null of the FIR filter then
    FIR formula5 2 (5)
    is the null of the FIR filter too!
  5. For the design of FIR filters used by Parks and McClellan algorithm using Chebyshev theory and Remez exchange method. On the internet you can easily find a calculator for the FIR filter coefficients by this method.
    For example:

3. FIR Filter in recursive Form

This method is rather theoretical in nature and I have not met the use of this implementation in practice. The z-transform of the FIR filter can be represented as follows (see [1]):
FIR formula6
FIR formula6 1– DFT (Discrete Fourier Transformation) from the impulse response of the FIR filter:

h (0), h(1),…,h (N-1), so this implementation is called a frequency-sampled structure.
The implementation of the FIR filter according to this scheme is as follows:
FIR recursive form

                                    Figure 3-1: FIR Filter in recursive Form

  1. The poles of this filter
    FIR formula6 2
    are compensated by nulls
    FIR formula6 3
  2. All poles location on the unit circle, which can lead to instability of the filter, since all calculations have finite accuracy
  3. The filter is used the complex numbers, which requires double the computing power, since calculations must be done separately for the imaginary and real parts of complex values
  4. This filter takes a simple form in the case of a narrow-band filter, since in this case most of the values of H(k) are zero
  5. I do not provide an code implementation of the filter realization in demo SW

4. FIR Filter in direct Form

The variant is used mainly in practical applications. This form is described in (1) and the implementation is showed in Figure 4-1:

FIR direct 1

                            Figure 4-1: FIR Filter in direct Form

  1. The realization has N-1 delays and requires to save N-1 input samples: x(n-1), … , x(n-N+1)
  2. I present a demo SW with the method implementation. See the functions fir_coeff_float(…)/ fir_coeff_integer(…) from file: fir_filter.c/h
  3. Using the symmetry of the impulse response (2) and (3), it is possible to reduce the number of multiplications:
    • Variant 1: h(k) = h(N-1-k). If N is an even number then we can write:
      FIR formula7
      If N is an odd number then:
      FIR formula8 1
    • Variant 2: h(k) = – h(N-1-k). If N is an even number then we can write:
      FIR formula9
      If N is an odd number then:
      FIR formula10 1

5. FIR Filter in cascade Form

As the filter order increases (but the FIR filter always has a high order) there is a high sensitivity of the filter nulls from quantization or rounding of the filter coefficients. In this case, using a cascading implementation of the FIR filter can help.
To preserve the linearity of the phase characteristic of the filter, the inverse nulls (5) should be combined together. In this case, we get a 4-order block:

FIR formula11

For real inverse nulls, we get a 2-order block:

FIR formula12

For nulls on the unit circle, two options are possible. This is a 2-order block for complex nulls and a first-order block for real null 1 or -1:

FIR formula13 14

Thus, the FIR filter can be implemented in a cascade form, using the 1, 2, 4 – order blocks.

FIR cascade

                     Figure 5-1: FIR Filter in cascade Form

  • See the [1]. The FIR filter in direct form gives a good result and usually the cascading form does not need in practice. This is due to the fact that the nulls of the FIR filter are located at a sufficient distance from each other and quantization of the coefficients does not lead to a deterioration in the characteristics of the filter. Therefore, I hope that you will not need the cascading form of your implementation in practice
  • You can implement a cascading form by repeatedly using the fir_coeff_float(…)/ fir_coeff_integer(…) functions from the file fir_filter.c/h

6. FIR Filter using Convolution with the Overlap-Add Method

Sometimes there is a long data sequences of input samples and we need to quickly pass the samples in the filter. The calculations must be performed in separate data blocks.
The filter response can be calculated using the convolution:
FIR formula15
Let the input x (n) sequence have length M (x (0), x (1),…, x (M-1)) and the impulse response of the filter h(n) has length N (h (0), h (1),…, h(N-1)), then the convolution of these two sequences will have length M+N-1, and the result can be calculated by the formula:

FIR formula16 1


To calculate the output long data sequence, we divide the input data into blocks of length M. We will calculate the convolution for each individual block. To combine the results of these calculations, there are two methods: overlap-add method and overlap-save method (see [1], [2]). I will focus here only on the overlap-add method. In this case, the filter response to individual blocks is calculated, and the overlap between these reactions is summed up. See the Figure 6-1.




          Figure 6-1: FIR Filter. Convolution with the Overlap-Add Method

The calculation of convolution for each data block takes quite a long time, so the fast convolution algorithm is often used. This method is based on the DFT and IDFT algorithms, which can be effectively implemented based on the FFT algorithm. It is known (see [1] or [2]) that the signal spectra y (n), x (n) and h (n) in (16) can be calculated by the formula:
FIR formula17
where Y(k), X(k), H(k) are DFT from signals:
FIR formula17 1
Next, I will describe a fast convolution algorithm that can be used with the Overlap-Add Method to calculate the filtered signal of the Long Data Sequences:
  • Add zeros to the impulse response: h (0), h(1),…, h(N-1), 0,…, 0 so that the array size is N + M – 1, where M is the size of the input signal block. Then using FFT algorithm, the DFT is calculated: H(k)
  • Take a block, for example i-block of input data of length M. We add zeros to this data:
    FIR formula17 2,
    to get an array of length N+M-1, where N is the size of the impulse response of the FIR filter. Using FFT algotithm, we calculate the DFT: Xi(k)
  • Multiply the Yi(k)=Xi(k)H(k). Calculate the IDFT from the Yi(k) and get the yi(n)
    To get the output signal, calculate the sum for all yi(n) (see the Figure 6-1):
    FIR formula17 3
The algorithm is implemented in my article “Fast Fourier Transform (FFT) and C Implementation Using the Octave GNU Tool“.

7. M-Source Code FIR_Filter.m for “Octave“ Gnu Tool

For example I designed a Low Pass FIR filter on the internet site with the following parameters:
Sampling frequency Fs=12 kHz
Pass Band: 0 Hz …1 kHz with ripple 2 dB
Stop Band: 2 kHz…6 kHz with ripple -40 dB
Filter Order: 17
Next, to analyze the resulting filter, I wrote the fir_filter.m code for the Octave GNU Tool:
  • Normalized the filter coefficients so that H(0Hz)=1. See the transformation the fir_coeff_original to fir_coeff;
  • Coefficients quantization of the FIR filter from float format to integer format (signed int16) with the factor:
    FIR formula17 4
    using the my function: FirFilterQuantization(…). See the transformation the fir_coeff to fir_coeff_q.
    Note: the value 17 is chosen solely as an example, which does not give an integer overflow for this FIR filter with these coefficients. Also, of course, it is necessary to take into account the bit size of the input signal x(n), which is often determined by the ADC: 8-14 bits;
  • Build the Magnitude and Phase Responses using my function FirFilterResponse(…) or standard Octave functions: freqz(…)/freqz_plot(…). See the Figure 7-1
  • Prepare the few test signals: step signal, sinusoidal signal 500Hz (pass band space) and sinusoidal signal 2kHz (stop band space). Check the FIR filter response on the signals using my function FirFilter_FftConvolutionOverlapAdd(…). The function is realized with the Overlap-Add Convolution Method. See the Figure 7-2
  • Generate the help files with data arrays for the C-Code using my functions IntegerTestSignals2file(…), IntegerParamVector2file(…), FloatParamVector2file(…). The data can be copied to C-Code: FIR filters coefficients, test signals and so on;

FIR 17order AmplitudeResponse

FIR 17order PhaseResponse

              Figure 7-1: Low Pass FIR Filter. Magnitude and Phase Responses

Figure 7-1 notes:
  • Low Pass FIR Filter. Sampling Frequency is 12kHz. Pass band is 0…1kHz. Stop Band 2kHz…6kHz
  • The filter has linear phase response. Phase jumps of 180 degrees are explained by a bug in the function unwrap() of the Octave Gnu. Really the phase response is linear
  • Magnitude and Phase Responses was done after quantization coefficients of the FIR filter from float to integer format
FIR 17order StepResponse

FIR 17order Sinus500HzResponse

FIR 17order Sinus2kHzResponse

                        Figure 7-2: FIR Filter. Signal Responses for Low Pass Filter.
                        The blue signal is filter input and the red signal is filter output.


Figure 7-2 notes:
  • Despite the linear phase characteristic, the filter response on the step signal has two spikes (about 7%). This interference occurs because the pulse response (FIR) is limited in time, as opposed to the pulse response of an ideal low-pass filter
  • The signal with a frequency of 500Hz (pass band space) passes through the filter only with a delay (N-1)/2=(17-1)/2=8 samples
  • The signal with a frequency of 2 kHz (stop band space) is suppressed by -40 dB (in 100 times)

8. C-Source Code: fir_filter.c/h

The file has the follow definitions:
1. Constants:
#define FIR_ORDER N – Order filter definitions
#define COEFF_SHIFT_FACTOR M – The definition is supported the FIR filter with the integer coefficients. The factor is
FIR formula17 5
Support the fir_filter_integer(…) function
const float fir_coeff_float[FIR_ORDER] = {…} – FIR filter coefficients in float format. Support the fir_filter_float(…) function
const sint16 fir_coeff_integer[FIR_ORDER] = {…} – FIR filter coefficients in int16 format. Support the fir_filter_integer(…) function
2. Functions
void fir_filter_initialization(void) – initialize / clear the filter history (x(n-1),…x(n-N+1))
sint16 fir_filter_float(sint16 x_input) – calculate the output signal y(n) using the input signal x(n)=x_input. The function is used the float FIR coefficients
sint16 fir_filter_integer(sint16 x_input) – calculate the output signal y(n) using the input signal x(n)=x_input. The function is used the integer FIR coefficients

9. Download the FIR_Filter.m and fir_filter.c/h

You can download the files with the button:


10. Literature / References

[1] L.Rabiner, B.Gold “Theory and application of digital signal processing“, Prentice-Hall, Ing Englewood Cliffs, New Jercy 1975
[2] A.Oppenheim, R.Schafer “Discrete-Time Signal Processing“, published by Pearson Education, Inc, publishing as Prentice Hall, 1999