Finite Impulse Response Filters

Finite impulse response (FIR) filters allow the use of arbitrary filters with a finite number of taps. This library does not provide FIR filter design tools, but allows for coefficients to be imported from other design tools, such as SciPy/filter_design.

FIR Direct

The direct FIR implements the filter as a convolution in the time domain. This library uses FIR filter_fir_s32 implementation from lib_xcore_math to run on xcore. More information on implementation can be found in XCORE Math Library documentation.

class audio_dsp.dsp.fir.fir_direct(fs: float, n_chans: int, coeffs_path: Path, Q_sig: int = 27)

An FIR filter, implemented in direct form in the time domain.

When the filter coefficients are converted to fixed point, if there will be leading zeros, a left shift is applied to the coefficients in order to use the full dynamic range of the VPU. A subsequent right shift is applied to the accumulator after the convolution to return to the same gain.

Parameters:

fsint

Sampling frequency in Hz.

n_chansint

Number of channels the block runs on.

coeffs_pathPath

Path to a file containing the coefficients, in a format supported by np.loadtxt.

Q_sig: int, optional

Q format of the signal, number of bits after the decimal point. Defaults to Q4.27.

Attributes:

fsint

Sampling frequency in Hz.

n_chansint

Number of channels the block runs on.

Q_sig: int

Q format of the signal, number of bits after the decimal point.

coeffsnp.ndarray

Array of the FIR coefficients in floating point format.

coeffs_intlist

Array of the FIR coefficients in fixed point int32 format.

shiftint

Right shift to be applied to the fixed point convolution result. This compensates for any left shift applied to the coefficients.

n_tapsint

Number of taps in the filter.

buffernp.ndarray

Buffer of previous inputs for the convlution in floating point format.

buffer_intlist

Buffer of previous inputs for the convlution in fixed point format.

buffer_idxlist

List of the floating point buffer head for each channel.

buffer_idx_intlist

List of the fixed point point buffer head for each channel.

process(sample: float, channel: int = 0) → float

Update the buffer with the current sample and convolve with the filter coefficients, using floating point math.

Parameters:

samplefloat

The input sample to be processed.

channelint

The channel index to process the sample on.

Returns:

float

The processed output sample.

reset_state() → None

Reset all the delay line values to zero.

check_coeff_scaling()

Check the coefficient scaling is optimal.

If there will be leading zeros, calculate a shift to use the full dynamic range of the VPU