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    <title>Filter design on Digital Signal Processing: Tips, Tricks and more</title>
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    <description>Recent content in Filter design on Digital Signal Processing: Tips, Tricks and more</description>
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    <lastBuildDate>Thu, 28 Jun 2018 16:38:43 +0200</lastBuildDate>
    
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      <title>Lowpass filter with arbitrary steepness</title>
      <link>https://dspblog.audio-dsp.de/post/iirlowpassfilterarbitrary/</link>
      <pubDate>Thu, 28 Jun 2018 16:38:43 +0200</pubDate>
      
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      <description>&lt;p&gt;Lowpass filter design (Butterworth) for speech and audio is always connected with the bilinear transform and its distortions near Nyquist. Furthermore, we are limited to integer order and the corresponding 6dB / Oct per order steepness. Read on, and you will discover a solution for both problems by using the frequency domain least squares design routine for IIR filters. A Python Jupyter notebook is provided to play around with it.&lt;/p&gt;</description>
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