# Documentation

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# designVarSlopeFilter

Design variable slope lowpass or highpass IIR filter

## Syntax

[B,A] = designVarSlopeFilter(slope,Fc)
[B,A] = designVarSlopeFilter(slope,Fc,type)

## Description

example

[B,A] = designVarSlopeFilter(slope,Fc) designs a lowpass filter with the specified slope and cutoff frequency. B and A are matrices of numerator and denominator coefficients, with columns corresponding to cascaded second-order sections (SOS).

example

[B,A] = designVarSlopeFilter(slope,Fc,type) specifies the design type as a lowpass or highpass filter.

## Examples

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Design two second-order section (SOS) lowpass IIR filters using designVarSlopeFilter.

Specify the sampling frequency, slope, and normalized cutoff frequency for two lowpass IIR filters. The sampling frequency is in Hz. The slope is in dB/octave.

Fs = 48e3;

slope = 18;

Fc1 = 10000/(Fs/2);
Fc2 = 16000/(Fs/2);

Design the filter coefficients using the specified parameters.

[B1,A1] = designVarSlopeFilter(slope,Fc1);
[B2,A2] = designVarSlopeFilter(slope,Fc2);

Create filter matrices compatible with fvtool.

SOS1 = [B1',[ones(4,1),A1']];
SOS2 = [B2',[ones(4,1),A2']];

fvtool(SOS1,SOS2,'Fs',Fs);

legend('Fc = 10000 Hz',...
'Fc = 16000 Hz',...
'Location','SouthWest');

Design a second-order section (SOS) lowpass IIR filter using designVarSlopeFilter. Use your lowpass filter to process an audio signal.

Construct audio file reader and audio device writer System objects. Use the sample rate of the reader as the sample rate of the writer. Call setup to reduce the computational load of initialization in an audio stream loop.

frameSize = 256;

'RockGuitar-16-44p1-stereo-72secs.wav', ...
'SamplesPerFrame',frameSize);

deviceWriter = audioDeviceWriter( ...
'SampleRate',sampleRate);

setup(deviceWriter,ones(frameSize,2));

Play the audio signal through your device.

count = 0;
while count < 2500
deviceWriter(audio);
count = count+1;
end

Design a lowpass filter with a 12 dB/octave slope and a 0.15 normalized frequency cutoff.

[B,A] = designVarSlopeFilter(12,0.15);

SOS = [B',[ones(4,1),A']];
fvtool(SOS, ...
'Fs',sampleRate);

Construct a biquad filter System object.

'SOSMatrixSource','Input port', ...
'ScaleValuesInputPort',false);

Construct a spectrum analyzer System object to visualize the original audio signal and the audio signal passed through your lowpass filter.

scope = dsp.SpectrumAnalyzer( ...
'SampleRate',sampleRate, ...
'PlotAsTwoSidedSpectrum',false, ...
'FrequencyScale','Log', ...
'FrequencyResolutionMethod','WindowLength', ...
'WindowLength',frameSize, ...
'Title','Original and Equalized Signal', ...
'ShowLegend',true, ...
'ChannelNames',{'Original Signal','Filtered Signal'});

Play the filtered audio signal and visualize the original and filtered spectrums.

setup(scope,ones(frameSize,2));
count = 0;
while count < 2500
filteredSignal = myFilter(originalSignal,B,A);
scope([originalSignal(:,1),filteredSignal(:,1)]);
deviceWriter(filteredSignal);
count = count+1;
end

Design two second-order section (SOS) highpass IIR filters using designVarSlopeFilter.

Specify the sampling frequency in Hz, the slope in dB/octave, and the normalized cutoff frequency.

Fs = 48e3;
slope1 = 18;
slope2 = 36;
Fc = 4000/(Fs/2);

Design the filter coefficients using the specifed parameters.

[B1,A1] = designVarSlopeFilter(slope1,Fc,'hi');
[B2,A2] = designVarSlopeFilter(slope2,Fc,'hi');

Create filter matrices compatible with fvtool.

SOS1 = [B1',[ones(4,1),A1']];
SOS2 = [B2',[ones(4,1),A2']];

fvtool(SOS1,SOS2,...
'Fs',Fs,...
'FrequencyScale','Log');
legend('slope = 18 dB/octave',...
'slope = 36 dB/octave',...
'Location','NorthWest')

Plosives are consonant sounds resulting from a sudden release of airflow. They are most pronounced in words beginning with p, d, and g sounds. Plosives can be emphasized by the recording process and are often displeasurable to hear. In this example, you minimize the plosives of a speech signal by applying highpass filtering and low-band compression.

Create a dsp.AudioFileReader System object™ and a audioDeviceWriter System object™ to read an audio signal from a file and write an audio signal to a device. Play the unprocessed signal. Then release the file reader and device writer.

fullfile(matlabroot,'examples','audio','Plosives.wav'));
deviceWriter = audioDeviceWriter;

deviceWriter(audioIn);
end
release(deviceWriter)

Design a highpass filter with a steep rolloff of all frequencies below 120 Hz. Use a dsp.BiquadFilter System object to implement the highpass filter design. Create a crossover filter with one crossover at 250 Hz. The crossover filter enables you to separate the band of interest for processing. Create a dynamic range compressor to compress the dynamic range of plosive sounds. To apply no make-up gain, set the MakeUpGainMode to 'Property' and use the default 0 dB MakeUpGain property value. Create a time scope to visualize the processed and unprocessed audio signal.

[B,A] = designVarSlopeFilter(48,120/(44100/2),'hi');
'SOSMatrixSource','Input port', ...
'ScaleValuesInputPort',false);

crossFilt = crossoverFilter( ...
'NumCrossovers',1, ...
'CrossoverFrequencies',250, ...
'CrossoverSlopes',48);

dRCompressor = compressor( ...
'Threshold',-35, ...
'Ratio',10, ...
'KneeWidth',20, ...
'AttackTime',1e-4, ...
'ReleaseTime',3e-1, ...
'MakeUpGainMode','Property', ...

scope = dsp.TimeScope( ...
'TimeSpan',3, ...
'YLimits',[-1 1], ...
'ShowGrid',true, ...
'ShowLegend',true, ...
'ChannelNames',{'Original','Processed'});

In an audio stream loop:

1. Read in a frame of the audio file.

3. Split the audio signal into two bands.

4. Apply dynamic range compression to the lower band.

5. Remix the channels.

6. Write the processed audio signal to your audio device for listening.

7. Visualize the processed and unprocessed signals on a time scope.

As a best practice, release your objects once done.

[band1,band2] = crossFilt(audioIn);

band1compressed = dRCompressor(band1);

audioOut  = band1compressed + band2;

deviceWriter(audioOut);

scope([audioIn audioOut]);
end

release(deviceWriter)
release(scope)
release(crossFilt)
release(dRCompressor)

## Input Arguments

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Filter slope in dB/octave, specified as a real scalar in the range [0:6:48]. Values that are not multiples of 6 are rounded.

Normalized cutoff frequency, specified as a real scalar in the range 0 to 1, where 1 corresponds to the Nyquist frequency (π rad/sample).

Filter type, specified as 'lo' or 'hi'.

• 'lo'— Lowpass filter

• 'hi'— Highpass filter

## Output Arguments

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Numerator filter coefficients, returned as a 3-by-4 matrix. Each column of B corresponds to the numerator coefficients of a different second-order section of your cascaded IIR filter.

Denominator filter coefficients, returned as a 2-by-4 matrix. Each column of A corresponds to the denominator coefficients of a different second-order section of your cascaded IIR filter.

A does not include the leading unity coefficient for each section.