designVarSlopeFilter
Design variable slope lowpass or highpass IIR filter
Syntax
Description
[
specifies options using one or more B,A] =
designVarSlopeFilter(___,Name,Value)Name,Value pair
arguments.
Examples
Design Lowpass IIR Filter
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 = 10e3/(Fs/2); Fc2 = 16e3/(Fs/2);
Design the filter coefficients using the specified parameters.
[B1,A1] = designVarSlopeFilter(slope,Fc1,"Orientation","row"); [B2,A2] = designVarSlopeFilter(slope,Fc2,"Orientation","row");
Visualize your filter design.
fvtool([B1,A1],[B2,A2],"Fs",Fs,"FrequencyScale","Log"); legend("Fc = 10 kHz", ... "Fc = 16 kHz", ... "Location","SouthWest");
Process Audio Using Lowpass Filter
Design a second-order section (SOS) lowpass IIR filter using designVarSlopeFilter. Use your lowpass filter to process an audio signal.
Create audio file reader and audio device writer objects. Use the sample rate of the reader as the sample rate of the writer.
frameSize = 256; fileReader = dsp.AudioFileReader( ... "RockGuitar-16-44p1-stereo-72secs.wav", ... "SamplesPerFrame",frameSize); sampleRate = fileReader.SampleRate; deviceWriter = audioDeviceWriter( ... "SampleRate",sampleRate);
Play the audio signal through your device.
count = 0; while count < 2500 audio = fileReader(); deviceWriter(audio); count = count + 1; end reset(fileReader)
Design a lowpass filter with a 12 dB/octave slope and a 0.15 normalized cutoff frequency.
slope = 12; cutoff = 0.15; [B,A] = designVarSlopeFilter(slope,cutoff);
Visualize your filter design. To output filter coefficients suitable for fvtool, call designVarSlopeFilter again with the same design specifications but with Orientation set to "row".
[Bvisualize,Avisualize] = designVarSlopeFilter(slope,cutoff,"Orientation","row"); fvtool([Bvisualize,Avisualize],"Fs",sampleRate);
Create a biquad filter.
myFilter = dsp.BiquadFilter( ... "SOSMatrixSource","Input port", ... "ScaleValuesInputPort",false);
Create a spectrum analyzer 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.
count = 0; while count < 2500 originalSignal = fileReader(); filteredSignal = myFilter(originalSignal,B,A); scope([originalSignal(:,1),filteredSignal(:,1)]); deviceWriter(filteredSignal); count = count + 1; end
As a best practice, release your objects once done.
release(deviceWriter) release(fileReader) release(scope)
Design Highpass IIR Filter
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 specified parameters.
[B1,A1] = designVarSlopeFilter(slope1,Fc,"hi","Orientation","row"); [B2,A2] = designVarSlopeFilter(slope2,Fc,"hi","Orientation","row");
Visualize your filter design.
fvtool([B1,A1],[B2,A2],... "Fs",Fs,... "FrequencyScale","Log"); legend("slope = 18 dB/octave", ... "slope = 36 dB/octave", ... "Location","NorthWest")
Diminish Plosives from Speech Signal
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 object and a audioDeviceWriter 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.
fileReader = dsp.AudioFileReader('audioPlosives.wav'); deviceWriter = audioDeviceWriter('SampleRate',fileReader.SampleRate); while ~isDone(fileReader) audioIn = fileReader(); deviceWriter(audioIn); end release(deviceWriter) release(fileReader)
Design a highpass filter with a steep rolloff of all frequencies below 120 Hz. Use a dsp.BiquadFilter 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/(fileReader.SampleRate/2),"hi"); biquadFilter = dsp.BiquadFilter( ... "SOSMatrixSource","Input port", ... "ScaleValuesInputPort",false); crossFilt = crossoverFilter( ... "SampleRate",fileReader.SampleRate, ... "NumCrossovers",1, ... "CrossoverFrequencies",250, ... "CrossoverSlopes",48); dRCompressor = compressor( ... "Threshold",-35, ... "Ratio",10, ... "KneeWidth",20, ... "AttackTime",1e-4, ... "ReleaseTime",3e-1, ... "MakeUpGainMode","Property", ... "SampleRate",fileReader.SampleRate); scope = timescope( ... "SampleRate",fileReader.SampleRate, ... "TimeSpanSource","property","TimeSpan",3, ... "BufferLength",fileReader.SampleRate*3*2, ... "YLimits",[-1 1], ... "ShowGrid",true, ... "ShowLegend",true, ... "ChannelNames",{'Original','Processed'});
In an audio stream loop:
Read in a frame of the audio file.
Apply highpass filtering using your biquad filter.
Split the audio signal into two bands.
Apply dynamic range compression to the lower band.
Remix the channels.
Write the processed audio signal to your audio device for listening.
Visualize the processed and unprocessed signals on a time scope.
As a best practice, release your objects once done.
while ~isDone(fileReader) audioIn = fileReader(); audioIn = biquadFilter(audioIn,B,A); [band1,band2] = crossFilt(audioIn); band1compressed = dRCompressor(band1); audioOut = band1compressed + band2; deviceWriter(audioOut); scope([audioIn audioOut]) end
As a best practice, release your objects once done.
release(deviceWriter) release(fileReader) release(crossFilt) release(dRCompressor) release(scope)
Input Arguments
Output Arguments
References
[1] Orfanidis, Sophocles J. "High-Order Digital Parametric Equalizer Design." Journal of the Audio Engineering Society. Vol. 53, November 2005, pp. 1026–1046.