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Lucas Cordiviola 2022-04-12 23:32:40 -03:00
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<LI><A ID="tex2html215"
HREF="node5.html">Foreword</A>
<LI><A NAME="tex2html216"
<LI><A ID="tex2html216"
HREF="node6.html">Preface</A>
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<LI><A ID="tex2html217"
HREF="node7.html">Sinusoids, amplitude and frequency</A>
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<LI><A NAME="tex2html218"
<LI><A ID="tex2html218"
HREF="node8.html">Measures of Amplitude</A>
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HREF="node10.html">Controlling Amplitude</A>
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HREF="node11.html">Frequency</A>
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HREF="node16.html">Quick Introduction to Pd</A>
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HREF="node17.html">How to find and run the examples</A>
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<LI><A NAME="tex2html229"
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HREF="node19.html">Constant amplitude scaler</A>
<LI><A NAME="tex2html230"
<LI><A ID="tex2html230"
HREF="node20.html">Amplitude control in decibels</A>
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HREF="node21.html">Smoothed amplitude control with an envelope generator</A>
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HREF="node24.html">More additive synthesis</A>
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HREF="node27.html">The Wavetable Oscillator</A>
<LI><A NAME="tex2html238"
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HREF="node28.html">Sampling</A>
<LI><A NAME="tex2html239"
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HREF="node29.html">Enveloping samplers</A>
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HREF="node34.html">Wavetable lookup in general</A>
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HREF="node35.html">Using a wavetable as a sampler</A>
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HREF="node99.html">Simple formant generator</A>
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HREF="node103.html">Exercises</A>
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HREF="node104.html">Time shifts and delays</A>
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HREF="node108.html">Delay networks</A>
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HREF="node109.html">Recirculating delay networks</A>
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HREF="node110.html">Power conservation and complex delay networks</A>
<LI><A NAME="tex2html321"
<LI><A ID="tex2html321"
HREF="node111.html">Artificial reverberation</A>
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<LI><A NAME="tex2html322"
<LI><A ID="tex2html322"
HREF="node112.html">Controlling reverberators</A>
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HREF="node120.html">Order of execution and lower limits on delay times</A>
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<LI><A ID="tex2html332"
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<LI><A ID="tex2html333"
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<LI><A NAME="tex2html334"
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<LI><A ID="tex2html336"
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HREF="node127.html">Filters</A>
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<LI><A ID="tex2html339"
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HREF="node133.html">Elementary non-recirculating filter</A>
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HREF="node134.html">Non-recirculating filter, second form</A>
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<LI><A ID="tex2html345"
HREF="node135.html">Elementary recirculating filter</A>
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<LI><A ID="tex2html346"
HREF="node136.html">Compound filters</A>
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<LI><A ID="tex2html347"
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<LI><A NAME="tex2html348"
<LI><A ID="tex2html348"
HREF="node138.html">Two recirculating filters for the price of one</A>
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<LI><A NAME="tex2html349"
<LI><A ID="tex2html349"
HREF="node139.html">Designing filters</A>
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<LI><A NAME="tex2html350"
<LI><A ID="tex2html350"
HREF="node140.html">One-pole low-pass filter</A>
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<LI><A ID="tex2html351"
HREF="node141.html">One-pole, one-zero high-pass filter</A>
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<LI><A ID="tex2html352"
HREF="node142.html">Shelving filter</A>
<LI><A NAME="tex2html353"
<LI><A ID="tex2html353"
HREF="node143.html">Band-pass filter</A>
<LI><A NAME="tex2html354"
<LI><A ID="tex2html354"
HREF="node144.html">Peaking and stop-band filter</A>
<LI><A NAME="tex2html355"
<LI><A ID="tex2html355"
HREF="node145.html">Butterworth filters</A>
<LI><A NAME="tex2html356"
<LI><A ID="tex2html356"
HREF="node146.html">Stretching the unit circle with rational functions</A>
<LI><A NAME="tex2html357"
<LI><A ID="tex2html357"
HREF="node147.html">Butterworth band-pass filter</A>
<LI><A NAME="tex2html358"
<LI><A ID="tex2html358"
HREF="node148.html">Time-varying coefficients</A>
<LI><A NAME="tex2html359"
<LI><A ID="tex2html359"
HREF="node149.html">Impulse responses of recirculating filters</A>
<LI><A NAME="tex2html360"
<LI><A ID="tex2html360"
HREF="node150.html">All-pass filters</A>
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<LI><A NAME="tex2html361"
<LI><A ID="tex2html361"
HREF="node151.html">Applications</A>
<UL>
<LI><A NAME="tex2html362"
<LI><A ID="tex2html362"
HREF="node152.html">Subtractive synthesis</A>
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<LI><A ID="tex2html365"
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HREF="node159.html">Single sideband modulation</A>
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<LI><A ID="tex2html370"
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<H1><A NAME="SECTION00530000000000000000">
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Controlling Amplitude</A>
</H1>
@ -82,7 +82,7 @@ nominal amplitude <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig01.04"></A><A NAME="1090"></A>
<DIV ALIGN="CENTER"><A ID="fig01.04"></A><A ID="1090"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 1.4:</STRONG>
The relationship between "MIDI" pitch and frequency in cycles per
@ -166,36 +166,36 @@ considered later in Chapter <A HREF="node75.html#chapter-modulation">5</A>.
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Two-cosine carrier signal</A>
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@ -86,7 +86,7 @@ and <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig06.16"></A><A NAME="6994"></A>
<DIV ALIGN="CENTER"><A ID="fig06.16"></A><A ID="6994"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 6.17:</STRONG>
Cross-fading between sinusoids to make movable
@ -169,36 +169,36 @@ never avoid getting phase cancellations where they overlap.
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The PAF generator</A>
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<DIV ALIGN="CENTER"><A NAME="fig06.17"></A><A NAME="7005"></A>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 6.18:</STRONG>
The phase-aligned formant (PAF) synthesis algorithm.</CAPTION>
@ -126,7 +126,7 @@ control object is needed:
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<DIV ALIGN="CENTER"><A NAME="fig06.18"></A><A NAME="7047"></A>
<DIV ALIGN="CENTER"><A ID="fig06.18"></A><A ID="7047"></A>
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<CAPTION ALIGN="BOTTOM"><STRONG>Figure 6.19:</STRONG>
Filling in the wavetable for Figure <A HREF="#fig06.17">6.18</A>.</CAPTION>
@ -145,7 +145,7 @@ Filling in the wavetable for Figure <A HREF="#fig06.17">6.18</A>.</CAPTION>
WIDTH="57" HEIGHT="41" ALIGN="MIDDLE" BORDER="0"
SRC="img623.png"
ALT="\fbox{ $\mathrm{until}$\ }"> :
<A NAME="7054"></A>When the left, "start" inlet is banged, output sequential bangs (with no
<A ID="7054"></A>When the left, "start" inlet is banged, output sequential bangs (with no
elapsed time between them) iteratively, until the right, "stop" inlet is
banged. The stopping "bang" message must originate somehow from the
<TT>until</TT> object's outlet; otherwise, the outlet will send "bang" messages
@ -231,36 +231,36 @@ the fundamental dropping, not rising, in amplitude as the string decays.
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<BR>
Time shifts and delays
</H1>
@ -83,7 +83,7 @@ But now suppose you played it at 5:00 and 5:00:01 on the same day (on two
different playback systems, since the music lasts much longer than one second).
Now the sound is much different. The difference, whatever it is,
clearly resides in neither of the two individual sounds, but rather in the
<A NAME="7751"></A><I>interference</I>
<A ID="7751"></A><I>interference</I>
between the two. This interference can be perceived in at least four different
ways:
<DL>
@ -138,96 +138,96 @@ section of this chapter we will develop the additional background needed.
<P>
<BR><HR>
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HREF="node113.html">Variable and fractional shifts</A>
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HREF="node114.html">Fidelity of interpolating delay lines</A>
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<LI><A NAME="tex2html2111"
<LI><A ID="tex2html2111"
HREF="node120.html">Order of execution and lower limits on delay times</A>
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HREF="node121.html">Order of execution in non-recirculating delay lines</A>
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HREF="node122.html">Non-recirculating comb filter as octave doubler</A>
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<H1><A NAME="SECTION001110000000000000000"></A>
<A NAME="sect7.complex"></A>
<H1><A ID="SECTION001110000000000000000"></A>
<A ID="sect7.complex"></A>
<BR>
Complex numbers
</H1>
<P>
Complex
<A NAME="7761"></A>numbers are written as:
<A ID="7761"></A>numbers are written as:
<BR><P></P>
<DIV ALIGN="CENTER">
<!-- MATH
@ -122,9 +122,9 @@ line) to graph it, as shown in Figure <A HREF="#fig07.01">7.1</A>. The quantiti
WIDTH="10" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img21.png"
ALT="$b$"> are called the
<A NAME="7764"></A><I>real</I>
<A ID="7764"></A><I>real</I>
and
<A NAME="7766"></A><I>imaginary parts</I> of <IMG
<A ID="7766"></A><I>imaginary parts</I> of <IMG
WIDTH="15" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img20.png"
ALT="$Z$">, written as:
@ -165,7 +165,7 @@ b = \mathrm{im}(Z)
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.01"></A><A NAME="7772"></A>
<DIV ALIGN="CENTER"><A ID="fig07.01"></A><A ID="7772"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.1:</STRONG>
A number, <IMG
@ -191,7 +191,7 @@ If <IMG
WIDTH="15" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img20.png"
ALT="$Z$"> is a complex number, its
<A NAME="7775"></A><A NAME="7776"></A><I>magnitude</I> (or <I>absolute value</I>),
<A ID="7775"></A><A ID="7776"></A><I>magnitude</I> (or <I>absolute value</I>),
written as <IMG
WIDTH="24" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img629.png"
@ -218,7 +218,7 @@ point <IMG
<BR CLEAR="ALL">
<P></P>
and its
<A NAME="7781"></A><I>argument</I>,
<A ID="7781"></A><I>argument</I>,
written as <IMG
WIDTH="38" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img632.png"
@ -306,8 +306,8 @@ A complex number may be written in terms of its real and imaginary parts
WIDTH="76" HEIGHT="30" ALIGN="MIDDLE" BORDER="0"
SRC="img637.png"
ALT="$Z=a+bi$"> (this is called
<A NAME="7785"></A><I>rectangular form</I>), or alternatively in
<A NAME="7787"></A><I>polar form</I>,
<A ID="7785"></A><I>rectangular form</I>), or alternatively in
<A ID="7787"></A><I>polar form</I>,
in terms of <IMG
WIDTH="11" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img467.png"
@ -511,45 +511,45 @@ imaginary parts.
<P>
<BR><HR>
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Complex sinusoids</A>
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@ -104,7 +104,7 @@ arguments and magnitudes.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.02"></A><A NAME="7829"></A>
<DIV ALIGN="CENTER"><A ID="fig07.02"></A><A ID="7829"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.2:</STRONG>
The powers of a complex number <IMG
@ -360,7 +360,7 @@ sinusoid of this form, <IMG
WIDTH="15" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img23.png"
ALT="$A$"> is called the
<A NAME="7846"></A>
<A ID="7846"></A>
<I>complex amplitude</I>.
<P>
@ -426,36 +426,36 @@ end of whatever we're doing.
<P>
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<H1><A NAME="SECTION001120000000000000000"></A>
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<H1><A ID="SECTION001120000000000000000"></A>
<A ID="sect7.phase"></A>
<BR>
Time shifts and phase changes
</H1>
@ -152,7 +152,7 @@ further property that, if you time shift a sinusoid of frequency <IMG
result is another sinusoid of the same frequency; time shifting never
introduces frequencies that weren't present in the signal before it was
shifted. This property, called
<A NAME="7855"></A><I>time invariance</I>,
<A ID="7855"></A><I>time invariance</I>,
makes it easy to analyze the effects of time shifts--and linear combinations
of them--by considering separately what the operations do on individual
sinusoids.
@ -234,36 +234,36 @@ since the sinusoid advances <IMG
<P>
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<H1><A NAME="SECTION001130000000000000000"></A>
<A NAME="sect7.network"></A>
<H1><A ID="SECTION001130000000000000000"></A>
<A ID="sect7.network"></A>
<BR>
Delay networks
</H1>
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.03"></A><A NAME="7867"></A>
<DIV ALIGN="CENTER"><A ID="fig07.03"></A><A ID="7867"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.3:</STRONG>
A delay network. Here we add the incoming signal to a delayed
@ -95,7 +95,7 @@ successive moments in time, then time shifting the signal by <IMG
SRC="img28.png"
ALT="$d$"> samples
corresponds to a
<A NAME="7870"></A>
<A ID="7870"></A>
<I>delay</I>
of <IMG
WIDTH="31" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
@ -105,7 +105,7 @@ of <IMG
SRC="img36.png"
ALT="$R$"> is the sample rate.
Figure <A HREF="#fig07.03">7.3</A> shows one example of a
<A NAME="7873"></A><I>linear delay network</I>:
<A ID="7873"></A><I>linear delay network</I>:
an assembly of delay units, possibly with amplitude
scaling operations, combined using addition and subtraction. The output
is a linear function of the input, in the sense that adding two signals at the
@ -117,13 +117,13 @@ so that the gains and delay times do not change with time.
<P>
In general there are two ways of thinking about delay networks. We can think
in the
<A NAME="7875"></A><I>time domain</I>,
<A ID="7875"></A><I>time domain</I>,
in which we draw waveforms as functions of time (or of the index <IMG
WIDTH="13" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img75.png"
ALT="$n$">), and
consider delays as time shifts. Alternatively we may think in the
<A NAME="7877"></A><I>frequency domain</I>,
<A ID="7877"></A><I>frequency domain</I>,
in which we dose the input with a complex sinusoid (so that its output is a
sinusoid at the same frequency) and report the amplitude and/or phase change
wrought by the network, as a function of the frequency. We'll now look at the
@ -131,7 +131,7 @@ delay network of Figure <A HREF="#fig07.03">7.3</A> in each of the two ways in t
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.04"></A><A NAME="8340"></A>
<DIV ALIGN="CENTER"><A ID="fig07.04"></A><A ID="8340"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.4:</STRONG>
The time domain view of the delay network of Figure <A HREF="#fig07.03">7.3</A>.
@ -155,7 +155,7 @@ delayed copy of itself.
<P>
A frequently used test function is an
<A NAME="7886"></A><I>impulse</I>,
<A ID="7886"></A><I>impulse</I>,
which is a pulse lasting only one sample. The utility of this is that, if we
know the output of the network for an impulse, we can find the output for
any other digital audio signal--because any signal <IMG
@ -271,7 +271,7 @@ combined additively.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.05"></A><A NAME="8344"></A>
<DIV ALIGN="CENTER"><A ID="fig07.05"></A><A ID="8344"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.5:</STRONG>
Analysis, in the complex plane, of the frequency-domain behavior of
@ -483,19 +483,19 @@ The quantity <IMG
WIDTH="26" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img30.png"
ALT="$\vert H\vert$"> is called the
<A NAME="7917"></A><I>gain</I>
<A ID="7917"></A><I>gain</I>
of the delay network at the angular frequency <IMG
WIDTH="14" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img27.png"
ALT="$\omega $">, and is graphed in
Figure <A HREF="#fig07.06">7.6</A>. The frequency-dependent gain of a delay network (that
is, the gain as a function of frequency) is called the network's
<A NAME="7920"></A><I>frequency response</I>.
<A ID="7920"></A><I>frequency response</I>.
<P>
Since the network has greater gain at some frequencies
than at others, it may be considered as a
<A NAME="7922"></A><I>filter</I>
<A ID="7922"></A><I>filter</I>
that can be used to separate certain components of a sound from others.
Because of the shape of this particular gain expression as a function of
<IMG
@ -503,11 +503,11 @@ Because of the shape of this particular gain expression as a function of
SRC="img27.png"
ALT="$\omega $">, this kind of delay network
is called a (non-recirculating)
<A NAME="7924"></A><I>comb filter</I>.
<A ID="7924"></A><I>comb filter</I>.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.06"></A><A NAME="8359"></A>
<DIV ALIGN="CENTER"><A ID="fig07.06"></A><A ID="8359"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.6:</STRONG>
Gain of the delay network of Figure <A HREF="#fig07.03">7.3</A>, shown as a function
@ -596,36 +596,36 @@ over the entire range of possible delay times.
<P>
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<H1><A NAME="SECTION001140000000000000000"></A>
<A NAME="sect7.recirculatingcomb"></A>
<H1><A ID="SECTION001140000000000000000"></A>
<A ID="sect7.recirculatingcomb"></A>
<BR>
Recirculating delay networks
</H1>
@ -80,7 +80,7 @@ the network to engender yet others.
<P>
The simplest example of a recirculating network is the
<A NAME="7935"></A>
<A ID="7935"></A>
<I>recirculating comb filter</I>
whose block diagram is shown in Figure <A HREF="#fig07.07">7.7</A>. As with the
earlier, simple comb filter, the input signal is sent down a delay line whose
@ -97,7 +97,7 @@ multiplied by a number <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.07"></A><A NAME="7940"></A>
<DIV ALIGN="CENTER"><A ID="fig07.07"></A><A ID="7940"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.7:</STRONG>
Block diagram for a recirculating comb filter. Here <IMG
@ -128,11 +128,11 @@ gain <IMG
SRC="img29.png"
ALT="$g$">. In general, a delay network's output given an impulse as input is
called the network's
<A NAME="7944"></A><I>impulse response</I>.
<A ID="7944"></A><I>impulse response</I>.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.08"></A><A NAME="7948"></A>
<DIV ALIGN="CENTER"><A ID="fig07.08"></A><A ID="7948"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.8:</STRONG>
Time-domain analysis of the recirculating comb filter, using
@ -154,7 +154,7 @@ a larger magnitude than the previous one. Instead of falling exponentially
as they do in the figure, they would grow exponentially. A recirculating
network whose output eventually falls toward zero after its input terminates
is called
<A NAME="7951"></A><I>stable</I>;
<A ID="7951"></A><I>stable</I>;
one whose output grows without bound is called <I>unstable</I>.
<P>
@ -420,7 +420,7 @@ frequency response <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.09"></A><A NAME="7976"></A>
<DIV ALIGN="CENTER"><A ID="fig07.09"></A><A ID="7976"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.9:</STRONG>
Diagram in the complex plane for approximating the output gain <IMG
@ -449,7 +449,7 @@ is chosen to give a gain about 3 dB below the peak.</CAPTION>
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.10"></A><A NAME="7981"></A>
<DIV ALIGN="CENTER"><A ID="fig07.10"></A><A ID="7981"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.10:</STRONG>
Frequency response of the recirculating comb filter with <IMG
@ -671,36 +671,36 @@ filters in Chapter <A HREF="node127.html#chapter-filter">8</A>.
<P>
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<H1><A NAME="SECTION00540000000000000000">
<H1><A ID="SECTION00540000000000000000">
Frequency</A>
</H1>
@ -78,7 +78,7 @@ the musical interval between them.
<P>
The Western musical scale divides the
<A NAME="1095"></A><I>octave</I> (the musical interval associated with a ratio of 2:1) into
<A ID="1095"></A><I>octave</I> (the musical interval associated with a ratio of 2:1) into
twelve equal sub-intervals, each of which therefore corresponds to a ratio
of <!-- MATH
${2 ^ {1/{12}}}$
@ -87,12 +87,12 @@ of <!-- MATH
WIDTH="37" HEIGHT="17" ALIGN="BOTTOM" BORDER="0"
SRC="img110.png"
ALT="${2 ^ {1/{12}}}$">. For historical reasons this sub-interval is called a
<A NAME="1098"></A><I>half-step</I>.
<A ID="1098"></A><I>half-step</I>.
A convenient logarithmic scale for pitch is simply to
count the number of half-steps from a reference pitch--allowing fractions to
permit us to specify pitches which don't fall on a note of the Western scale.
The most commonly used logarithmic pitch scale is
<A NAME="1100"></A>"MIDI pitch", in which the pitch 69 is assigned to a frequency of 440 cycles
<A ID="1100"></A>"MIDI pitch", in which the pitch 69 is assigned to a frequency of 440 cycles
per second--the A above middle C. To convert between a MIDI pitch <IMG
WIDTH="17" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img111.png"
@ -102,10 +102,10 @@ in cycles per second <IMG
WIDTH="13" HEIGHT="30" ALIGN="MIDDLE" BORDER="0"
SRC="img112.png"
ALT="$f$">, apply the
<A NAME="1101"></A>Pitch/Frequency Conversion formulas:
<A ID="1101"></A>Pitch/Frequency Conversion formulas:
<P>
<A NAME="eq-pitchmidi"></A>
<A ID="eq-pitchmidi"></A>
<BR><P></P>
<DIV ALIGN="CENTER">
<!-- MATH
@ -162,7 +162,7 @@ description of MIDI, its capabilities and limitations, see
<P>
A half-step comes to a ratio of about 1.059 to 1, or about a six percent
increase in frequency. Half-steps are further divided into <A NAME="1107"></A><I>cents</I>, each cent being one hundredth of a half-step. As a rule of
increase in frequency. Half-steps are further divided into <A ID="1107"></A><I>cents</I>, each cent being one hundredth of a half-step. As a rule of
thumb, it might take about three cents to make a discernible change in the
pitch of a musical tone. At middle C this comes to a difference of about 1/2
cycle per second. A graph of frequency as a function of MIDI pitch, over a
@ -171,36 +171,36 @@ two-octave range, is shown in Figure <A HREF="node10.html#fig01.04">1.4</A>.
<P>
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<H1><A NAME="SECTION001150000000000000000">
<H1><A ID="SECTION001150000000000000000">
Power conservation and complex delay networks</A>
</H1>
@ -90,7 +90,7 @@ the gain, suitably defined, is exactly one.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.11"></A><A NAME="8003"></A>
<DIV ALIGN="CENTER"><A ID="fig07.11"></A><A ID="8003"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.11:</STRONG>
First fundamental building block for unitary delay networks:
@ -163,7 +163,7 @@ where <IMG
It turns out that a wide range of interesting delay networks has the property
that the total power output equals the total power input;
they are called
<A NAME="8013"></A><I>unitary</I>. To start with, we can put any number of delays in parallel, as
<A ID="8013"></A><I>unitary</I>. To start with, we can put any number of delays in parallel, as
shown in Figure <A HREF="#fig07.11">7.11</A>. Whatever the total power of the inputs,
the total power of the outputs has to equal it.
@ -233,7 +233,7 @@ of a collection of signals must must be preserved by rotation.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.12"></A><A NAME="8024"></A>
<DIV ALIGN="CENTER"><A ID="fig07.12"></A><A ID="8024"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.12:</STRONG>
Second fundamental building block for unitary delay networks:
@ -329,7 +329,7 @@ s = \sin(\theta)
<BR CLEAR="ALL">
<P></P>
for an
<A NAME="8036"></A><I>angle of rotation</I> <IMG
<A ID="8036"></A><I>angle of rotation</I> <IMG
WIDTH="11" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img634.png"
ALT="$\theta$">.
@ -423,7 +423,7 @@ unit magnitude and argument <IMG
<P>
If we perform a rotation on a pair of signals and then invert one (but not the
other) of them, the result is a
<A NAME="8050"></A>
<A ID="8050"></A>
<I>reflection</I>.
This also preserves total signal power, since we can invert any or all of a
collection of signals without changing the total power. In two dimensions, a
@ -486,7 +486,7 @@ a) because each signal need only be multiplied by the one quantity <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.13"></A><A NAME="8389"></A>
<DIV ALIGN="CENTER"><A ID="fig07.13"></A><A ID="8389"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.13:</STRONG>
Details about rotation (and reflection) matrix operations: (a)
@ -535,7 +535,7 @@ recirculating networks that still enjoy flat frequency responses.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.14"></A><A NAME="8069"></A>
<DIV ALIGN="CENTER"><A ID="fig07.14"></A><A ID="8069"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.14:</STRONG>
Flat frequency response in recirculating networks: (a) in general,
@ -655,7 +655,7 @@ let the transformation <IMG
WIDTH="20" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img31.png"
ALT="$W$"> output, the result is the well-known
<A NAME="8086"></A><A NAME="8087"></A><I>all-pass filter</I>.
<A ID="8086"></A><A ID="8087"></A><I>all-pass filter</I>.
With some juggling, and letting <!-- MATH
$c = \cos(\theta)$
-->
@ -670,36 +670,36 @@ of which we will visit later in this book.
<P>
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<H1><A NAME="SECTION001160000000000000000">
<H1><A ID="SECTION001160000000000000000">
Artificial reverberation</A>
</H1>
@ -101,13 +101,13 @@ is not designed correctly.
To make an artificial reverberator using a delay network, we must
fill two competing demands simultaneously. First, the delay lines must
be long enough to prevent
<A NAME="8091"></A><I>coloration</I> in the output as a result of comb filtering.
<A ID="8091"></A><I>coloration</I> in the output as a result of comb filtering.
(Even if we move beyond the simple comb filter
of Section <A HREF="node109.html#sect7.recirculatingcomb">7.4</A>, the frequency response will tend to
have peaks and
valleys whose spacing varies inversely with total delay time.) On the other
hand, we should not hear individual echoes; the
<A NAME="8094"></A><I>echo density</I> should ideally be at least one thousand per second.
<A ID="8094"></A><I>echo density</I> should ideally be at least one thousand per second.
<P>
In pursuit of these aims, we assemble some number of delay lines and
@ -138,7 +138,7 @@ and 80 milliseconds. The figure shows three such stages.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.15"></A><A NAME="8099"></A>
<DIV ALIGN="CENTER"><A ID="fig07.15"></A><A ID="8099"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.15:</STRONG>
Reverberator design using power-preserving transformations and
@ -273,45 +273,45 @@ and lengthy tuning by trial, error, and critical listening.
<P>
<BR><HR>
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<H2><A NAME="SECTION001161000000000000000">
<H2><A ID="SECTION001161000000000000000">
Controlling reverberators</A>
</H2>
@ -88,7 +88,7 @@ moments of stable pitch.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.16"></A><A NAME="8114"></A>
<DIV ALIGN="CENTER"><A ID="fig07.16"></A><A ID="8114"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.16:</STRONG>
Controlling a reverberator to capture sounds selectively: (a)
@ -135,36 +135,36 @@ decay.
<P>
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<H1><A NAME="SECTION001170000000000000000"></A>
<A NAME="sect7-fractional"></A>
<H1><A ID="SECTION001170000000000000000"></A>
<A ID="sect7-fractional"></A>
<BR>
Variable and fractional shifts
</H1>
@ -181,7 +181,7 @@ result from the changes.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.17"></A><A NAME="8124"></A>
<DIV ALIGN="CENTER"><A ID="fig07.17"></A><A ID="8124"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.17:</STRONG>
A variable length delay line, whose output is the input from some
@ -350,7 +350,7 @@ t[n] = y[n] - y[n-1] = 1 - (d[n] - d[n-1])
</DIV>
<BR CLEAR="ALL">
<P></P>
<A NAME="eq.momentarydel"></A>If <IMG
<A ID="eq.momentarydel"></A>If <IMG
WIDTH="30" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img787.png"
ALT="$d[n]$"> does not change with <IMG
@ -372,7 +372,7 @@ downward, and if <IMG
<P>
This is called the
<A NAME="8132"></A><I>Doppler effect</I>, and it occurs in nature as well.
<A ID="8132"></A><I>Doppler effect</I>, and it occurs in nature as well.
The air that sound travels through can sometimes be thought of as a delay
line. Changing the length of the delay line corresponds to moving the
listener toward or away from a stationary sound source; the Doppler effect
@ -391,36 +391,36 @@ diagonal region.
<P>
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<H1><A NAME="SECTION001180000000000000000">
<H1><A ID="SECTION001180000000000000000">
Fidelity of interpolating delay lines</A>
</H1>
@ -200,7 +200,7 @@ itself, however, the gain is zero.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.18"></A><A NAME="8143"></A>
<DIV ALIGN="CENTER"><A ID="fig07.18"></A><A ID="8143"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.18:</STRONG>
Gain of a four-point interpolating delay line with a delay halfway
@ -221,36 +221,36 @@ the way up to 20000 Hertz.
<P>
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<H1><A NAME="SECTION001190000000000000000"></A>
<A NAME="sect7.pitchshift"></A>
<H1><A ID="SECTION001190000000000000000"></A>
<A ID="sect7.pitchshift"></A>
<BR>
Pitch shifting
</H1>
@ -87,7 +87,7 @@ in the strip.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.19"></A><A NAME="8151"></A>
<DIV ALIGN="CENTER"><A ID="fig07.19"></A><A ID="8151"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.19:</STRONG>
Vibrato using a variable delay line. Since the pitch shift
@ -161,7 +161,7 @@ This ranges in value between <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.20"></A><A NAME="8159"></A>
<DIV ALIGN="CENTER"><A ID="fig07.20"></A><A ID="8159"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.20:</STRONG>
Piecewise linear delay functions to maintain a constant transposition
@ -187,7 +187,7 @@ to avoid discontinuities.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.21"></A><A NAME="8165"></A>
<DIV ALIGN="CENTER"><A ID="fig07.21"></A><A ID="8165"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.21:</STRONG>
Using a variable delay line as a pitch shifter. The sawtooth wave
@ -230,7 +230,7 @@ pitch shifter since it is essentially free. The quantity <IMG
SRC="img208.png"
ALT="$s$"> is sometimes
called the
<A NAME="8171"></A><I>window size</I>. It corresponds roughly to the sample length in a
<A ID="8171"></A><I>window size</I>. It corresponds roughly to the sample length in a
looping sampler (Section <A HREF="node28.html#sect2.sampling">2.2</A>).
<P>
@ -330,7 +330,7 @@ Figure <A HREF="#fig07.22">7.22</A>.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.22"></A><A NAME="8183"></A>
<DIV ALIGN="CENTER"><A ID="fig07.22"></A><A ID="8183"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.22:</STRONG>
The pitch shifter's delay reading pattern using two delay lines,
@ -373,7 +373,7 @@ f = {{(t - 1) R} \over s}
</DIV>
<BR CLEAR="ALL">
<P></P>
<A NAME="sect7.psformula"></A>The window size <IMG
<A ID="sect7.psformula"></A>The window size <IMG
WIDTH="10" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img208.png"
ALT="$s$"> should be chosen small enough, if possible, so that the
@ -445,36 +445,36 @@ artifacts such as difference tones.
<P>
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<H1><A NAME="SECTION0011100000000000000000">
<H1><A ID="SECTION0011100000000000000000">
Examples</A>
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<P>
<BR><HR>
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<A NAME="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
<A ID="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
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<LI><A NAME="tex2html2284"
<LI><A ID="tex2html2284"
HREF="node117.html">Fixed, noninterpolating delay line</A>
<LI><A NAME="tex2html2285"
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HREF="node118.html">Recirculating comb filter</A>
<LI><A NAME="tex2html2286"
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HREF="node119.html">Variable delay line</A>
<LI><A NAME="tex2html2287"
<LI><A ID="tex2html2287"
HREF="node120.html">Order of execution and lower limits on delay times</A>
<LI><A NAME="tex2html2288"
<LI><A ID="tex2html2288"
HREF="node121.html">Order of execution in non-recirculating delay lines</A>
<LI><A NAME="tex2html2289"
<LI><A ID="tex2html2289"
HREF="node122.html">Non-recirculating comb filter as octave doubler</A>
<LI><A NAME="tex2html2290"
<LI><A ID="tex2html2290"
HREF="node123.html">Time-varying complex comb filter: shakers</A>
<LI><A NAME="tex2html2291"
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HREF="node124.html">Reverberator</A>
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<H2><A NAME="SECTION0011101000000000000000">
<H2><A ID="SECTION0011101000000000000000">
Fixed, noninterpolating delay line</A>
</H2>
@ -75,7 +75,7 @@ an input signal. Two new objects are needed:
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.23"></A><A NAME="8194"></A>
<DIV ALIGN="CENTER"><A ID="fig07.23"></A><A ID="8194"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.23:</STRONG>
Example patch G01.delay.pd, showing a noninterpolating delay with
@ -95,7 +95,7 @@ a delay time controlled in milliseconds.</CAPTION>
WIDTH="94" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img819.png"
ALT="\fbox{ \texttt{delwrite\~}}">:
<A NAME="8424"></A>define and write to a delay line. The first creation argument gives the name of the
<A ID="8424"></A>define and write to a delay line. The first creation argument gives the name of the
delay line (and two delay lines may not share the same name). The
second creation argument is the length of the delay line in milliseconds. The
inlet takes an audio signal and writes it continuously into the delay line.
@ -108,7 +108,7 @@ inlet takes an audio signal and writes it continuously into the delay line.
WIDTH="86" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img821.png"
ALT="\fbox{ \texttt{delread\~}}">:
<A NAME="8425"></A>read from (or "tap") a delay line. The first creation argument gives the name
<A ID="8425"></A>read from (or "tap") a delay line. The first creation argument gives the name
of the delay line (which should agree with the name of the corresponding
<TT>delwrite~</TT> object; this is how Pd knows which <TT>delwrite~</TT> to
associate with the <TT>delread~</TT> object). The
@ -130,36 +130,36 @@ possible when the delay time changes.
<P>
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<H2><A NAME="SECTION0011102000000000000000">
<H2><A ID="SECTION0011102000000000000000">
Recirculating comb filter</A>
</H2>
@ -81,7 +81,7 @@ of Section <A HREF="node109.html#sect7.recirculatingcomb">7.4</A>.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.24"></A><A NAME="8208"></A>
<DIV ALIGN="CENTER"><A ID="fig07.24"></A><A ID="8208"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.24:</STRONG>
Recirculating delay (still noninterpolating).</CAPTION>

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<H2><A NAME="SECTION0011103000000000000000">
<H2><A ID="SECTION0011103000000000000000">
Variable delay line</A>
</H2>
@ -76,7 +76,7 @@ new object is introduced here:
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.25"></A><A NAME="8215"></A>
<DIV ALIGN="CENTER"><A ID="fig07.25"></A><A ID="8215"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.25:</STRONG>
The flanger: an interpolating, variable delay line.</CAPTION>
@ -95,7 +95,7 @@ The flanger: an interpolating, variable delay line.</CAPTION>
WIDTH="44" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img825.png"
ALT="\fbox{ \texttt{vd\~}}">:
<A NAME="8426"></A>Read from a delay line, with a time-varying delay time. As with the
<A ID="8426"></A>Read from a delay line, with a time-varying delay time. As with the
<TT>delread~</TT> object, this reads from a delay line whose name is specified
as a creation argument. Instead of using a second argument and/or
control messages to specify the delay time, for the <TT>vd~</TT> object the delay in milliseconds is specified by an incoming audio signal.
@ -136,36 +136,36 @@ characteristic way.
<P>
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<H1><A NAME="SECTION00550000000000000000"></A>
<A NAME="sect1.synth"></A>
<H1><A ID="SECTION00550000000000000000"></A>
<A ID="sect1.synth"></A>
<BR>
Synthesizing a sinusoid
</H1>
@ -75,10 +75,10 @@ Synthesizing a sinusoid
In most widely used audio synthesis and processing packages (Csound, Max/MSP,
and Pd, for instance), the audio operations are specified as
networks of
<A NAME="1112"></A><I>unit generators</I>[<A
<A ID="1112"></A><I>unit generators</I>[<A
HREF="node202.html#r-mathews69">Mat69</A>] which pass audio signals among themselves. The user of
the software package specifies the network, sometimes called a
<A NAME="1115"></A><I>patch</I>,
<A ID="1115"></A><I>patch</I>,
which essentially corresponds to the synthesis algorithm to be used, and then
worries about how to control the various unit generators in time. In this
section, we'll use abstract block diagrams to describe patches, but in the
@ -91,14 +91,14 @@ details.
To show how to produce a sinusoid with time-varying amplitude we'll need to
introduce two unit generators. First we need a pure sinusoid which is
made with an
<A NAME="1118"></A><I>oscillator</I>. Figure <A HREF="#fig01.05">1.5</A> (part a) shows a pictorial
<A ID="1118"></A><I>oscillator</I>. Figure <A HREF="#fig01.05">1.5</A> (part a) shows a pictorial
representation of
a sinusoidal oscillator as an icon. The input is a frequency (in cycles per second), and
the output is a sinusoid of peak amplitude one.
<P>
<DIV ALIGN="CENTER"><A NAME="fig01.05"></A><A NAME="1123"></A>
<DIV ALIGN="CENTER"><A ID="fig01.05"></A><A ID="1123"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 1.5:</STRONG>
Block diagrams for (a) a sinusoidal oscillator; (b) controlling the
@ -131,7 +131,7 @@ negative in value.
<P>
<DIV ALIGN="CENTER"><A NAME="fig01.06"></A><A NAME="1351"></A>
<DIV ALIGN="CENTER"><A ID="fig01.06"></A><A ID="1351"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 1.6:</STRONG>
Two amplitude functions (parts a, c), and (parts b, d), the result of
@ -183,7 +183,7 @@ Suitable amplitude control functions <IMG
WIDTH="30" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img2.png"
ALT="$y[n]$"> may be made using an
<A NAME="1134"></A><I>envelope generator</I>.
<A ID="1134"></A><I>envelope generator</I>.
Figure <A HREF="#fig01.07">1.7</A> shows a network in which an envelope generator is used to
control the amplitude of an oscillator. Envelope generators vary widely in
design, but we will focus on the simplest
@ -252,7 +252,7 @@ Envelope generators are described in more detail in Section <A HREF="node59.html
<P>
<DIV ALIGN="CENTER"><A NAME="fig01.07"></A><A NAME="1148"></A>
<DIV ALIGN="CENTER"><A ID="fig01.07"></A><A ID="1148"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 1.7:</STRONG>
Using an envelope generator to control amplitude.</CAPTION>
@ -266,36 +266,36 @@ Using an envelope generator to control amplitude.</CAPTION>
<P>
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<H2><A NAME="SECTION0011104000000000000000">
<H2><A ID="SECTION0011104000000000000000">
Order of execution and lower limits on delay times</A>
</H2>
@ -94,7 +94,7 @@ delay operations might, for example).
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.26"></A><A NAME="8233"></A>
<DIV ALIGN="CENTER"><A ID="fig07.26"></A><A ID="8233"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.26:</STRONG>
Order of execution of tilde objects in Pd: (a), an acyclic network.
@ -249,7 +249,7 @@ a new object:
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.27"></A><A NAME="8407"></A>
<DIV ALIGN="CENTER"><A ID="fig07.27"></A><A ID="8407"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.27:</STRONG>
A patch using block size control to lower the loop delay below
@ -277,7 +277,7 @@ with a <TT>block~</TT> object and a recirculating delay network.</CAPTION>
WIDTH="78" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img834.png"
ALT="\fbox{ \texttt{switch\~}}">:
<A NAME="8427"></A><A NAME="8428"></A>Set the local block size of the patch window the object sits in. Block sizes
<A ID="8427"></A><A ID="8428"></A>Set the local block size of the patch window the object sits in. Block sizes
are normally powers of two. The <TT>switch~</TT> object, in addition, can be used to
turn audio computation within the window on and off, using control messages.
Additional creation arguments can set the local sample rate and specify
@ -298,7 +298,7 @@ minimum achievable delay is one sample instead of the default 64.
<P>
Putting a pulse (or other excitation signal) into a recirculating comb filter
to make a pitch is sometimes called
<A NAME="8260"></A><I>Karplus-Strong synthesis</I>,
<A ID="8260"></A><I>Karplus-Strong synthesis</I>,
having been described in a paper by them [<A
HREF="node202.html#r-karplus83">KS83</A>], although the idea
seems to be older. It shows up for example in Paul Lansky's 1979 piece,
@ -307,36 +307,36 @@ seems to be older. It shows up for example in Paul Lansky's 1979 piece,
<P>
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<H2><A NAME="SECTION0011105000000000000000">
<H2><A ID="SECTION0011105000000000000000">
Order of execution in non-recirculating delay lines</A>
</H2>
@ -84,7 +84,7 @@ G05.execution.order.pd (Figure <A HREF="#fig07.28">7.28</A>).
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.28"></A><A NAME="8410"></A>
<DIV ALIGN="CENTER"><A ID="fig07.28"></A><A ID="8410"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.28:</STRONG>
Using subpatches to ensure that delay lines are written before they
@ -125,36 +125,36 @@ delays below the 64 sample block size.
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Non-recirculating comb filter as octave doubler</A>
</H2>
@ -83,7 +83,7 @@ would have got by using speed change to do the transposition.
<P>
<DIV ALIGN="CENTER"><A NAME="fig07.29"></A><A NAME="8278"></A>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.29:</STRONG>
An "octave doubler" uses pitch information (obtained using
@ -165,36 +165,36 @@ up-shifting for best results.)
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Time-varying complex comb filter: shakers</A>
</H2>
@ -79,7 +79,7 @@ fourth "tap" is the original, un-delayed signal.
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<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.30:</STRONG>
A "shaker", a four-tap comb filter with randomly varying gains
@ -135,36 +135,36 @@ subject of Chapter 8).
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Reverberator</A>
</H2>
@ -81,7 +81,7 @@ rotations of <IMG
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<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.31:</STRONG>
An artificial reverberator.</CAPTION>
@ -110,7 +110,7 @@ reinserted into them, so the reverberation lasts perpetually.
<P>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.32:</STRONG>
The echo generator used in the reverberator.</CAPTION>
@ -141,36 +141,36 @@ sonic qualities described as "presence", "warmth", "clarity", and so on.
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Pitch shifter</A>
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@ -77,7 +77,7 @@ minimum plus a window size (the "window" control.)
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<DIV ALIGN="CENTER"><A NAME="fig07.33"></A><A NAME="8317"></A>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 7.33:</STRONG>
A pitch shifter using two variable taps into a delay
@ -141,36 +141,36 @@ envelopes and summed.
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Exercises</A>
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@ -174,36 +174,36 @@ function of angular frequency <IMG
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<BR>
Filters
</H1>
@ -79,11 +79,11 @@ variably depending on frequency. When the delay times used are very short,
the most important properties of a delay network become its frequency and
phase response. A delay network that is designed specifically for its
frequency or phase response is called a
<A NAME="10048"></A><I>filter</I>.
<A ID="10048"></A><I>filter</I>.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.01"></A><A NAME="10052"></A>
<DIV ALIGN="CENTER"><A ID="fig08.01"></A><A ID="10052"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.1:</STRONG>
Representations of a filter: (a) in a block diagram; (b) a graph of its
@ -116,7 +116,7 @@ depends on <IMG
WIDTH="40" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img853.png"
ALT="$H(\omega)$">, which is called the
<A NAME="10057"></A><I>transfer function</I>
<A ID="10057"></A><I>transfer function</I>
of the filter.
<P>
@ -168,128 +168,128 @@ when its parameters change quickly with time.
<P>
<BR><HR>
<!--Table of Child-Links-->
<A NAME="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
<A ID="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
<UL>
<LI><A NAME="tex2html2443"
<LI><A ID="tex2html2443"
HREF="node128.html">Taxonomy of filters</A>
<UL>
<LI><A NAME="tex2html2444"
<LI><A ID="tex2html2444"
HREF="node129.html">Low-pass and high-pass filters</A>
<LI><A NAME="tex2html2445"
<LI><A ID="tex2html2445"
HREF="node130.html">Band-pass and stop-band filters</A>
<LI><A NAME="tex2html2446"
<LI><A ID="tex2html2446"
HREF="node131.html">Equalizing filters</A>
</UL>
<BR>
<LI><A NAME="tex2html2447"
<LI><A ID="tex2html2447"
HREF="node132.html">Elementary filters</A>
<UL>
<LI><A NAME="tex2html2448"
<LI><A ID="tex2html2448"
HREF="node133.html">Elementary non-recirculating filter</A>
<LI><A NAME="tex2html2449"
<LI><A ID="tex2html2449"
HREF="node134.html">Non-recirculating filter, second form</A>
<LI><A NAME="tex2html2450"
<LI><A ID="tex2html2450"
HREF="node135.html">Elementary recirculating filter</A>
<LI><A NAME="tex2html2451"
<LI><A ID="tex2html2451"
HREF="node136.html">Compound filters</A>
<LI><A NAME="tex2html2452"
<LI><A ID="tex2html2452"
HREF="node137.html">Real outputs from complex filters</A>
<LI><A NAME="tex2html2453"
<LI><A ID="tex2html2453"
HREF="node138.html">Two recirculating filters for the price of one</A>
</UL>
<BR>
<LI><A NAME="tex2html2454"
<LI><A ID="tex2html2454"
HREF="node139.html">Designing filters</A>
<UL>
<LI><A NAME="tex2html2455"
<LI><A ID="tex2html2455"
HREF="node140.html">One-pole low-pass filter</A>
<LI><A NAME="tex2html2456"
<LI><A ID="tex2html2456"
HREF="node141.html">One-pole, one-zero high-pass filter</A>
<LI><A NAME="tex2html2457"
<LI><A ID="tex2html2457"
HREF="node142.html">Shelving filter</A>
<LI><A NAME="tex2html2458"
<LI><A ID="tex2html2458"
HREF="node143.html">Band-pass filter</A>
<LI><A NAME="tex2html2459"
<LI><A ID="tex2html2459"
HREF="node144.html">Peaking and stop-band filter</A>
<LI><A NAME="tex2html2460"
<LI><A ID="tex2html2460"
HREF="node145.html">Butterworth filters</A>
<LI><A NAME="tex2html2461"
<LI><A ID="tex2html2461"
HREF="node146.html">Stretching the unit circle with rational functions</A>
<LI><A NAME="tex2html2462"
<LI><A ID="tex2html2462"
HREF="node147.html">Butterworth band-pass filter</A>
<LI><A NAME="tex2html2463"
<LI><A ID="tex2html2463"
HREF="node148.html">Time-varying coefficients</A>
<LI><A NAME="tex2html2464"
<LI><A ID="tex2html2464"
HREF="node149.html">Impulse responses of recirculating filters</A>
<LI><A NAME="tex2html2465"
<LI><A ID="tex2html2465"
HREF="node150.html">All-pass filters</A>
</UL>
<BR>
<LI><A NAME="tex2html2466"
<LI><A ID="tex2html2466"
HREF="node151.html">Applications</A>
<UL>
<LI><A NAME="tex2html2467"
<LI><A ID="tex2html2467"
HREF="node152.html">Subtractive synthesis</A>
<LI><A NAME="tex2html2468"
<LI><A ID="tex2html2468"
HREF="node153.html">Envelope following</A>
<LI><A NAME="tex2html2469"
<LI><A ID="tex2html2469"
HREF="node154.html">Single Sideband Modulation</A>
</UL>
<BR>
<LI><A NAME="tex2html2470"
<LI><A ID="tex2html2470"
HREF="node155.html">Examples</A>
<UL>
<LI><A NAME="tex2html2471"
<LI><A ID="tex2html2471"
HREF="node156.html">Prefabricated low-, high-, and band-pass filters</A>
<LI><A NAME="tex2html2472"
<LI><A ID="tex2html2472"
HREF="node157.html">Prefabricated time-varying band-pass filter</A>
<LI><A NAME="tex2html2473"
<LI><A ID="tex2html2473"
HREF="node158.html">Envelope followers</A>
<LI><A NAME="tex2html2474"
<LI><A ID="tex2html2474"
HREF="node159.html">Single sideband modulation</A>
<LI><A NAME="tex2html2475"
<LI><A ID="tex2html2475"
HREF="node160.html">Using elementary filters directly: shelving and peaking</A>
<LI><A NAME="tex2html2476"
<LI><A ID="tex2html2476"
HREF="node161.html">Making and using all-pass filters</A>
</UL>
<BR>
<LI><A NAME="tex2html2477"
<LI><A ID="tex2html2477"
HREF="node162.html">Exercises</A>
</UL>
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<H1><A NAME="SECTION001210000000000000000">
<H1><A ID="SECTION001210000000000000000">
Taxonomy of filters</A>
</H1>
@ -80,14 +80,14 @@ applications of filters in electronic music.
<P>
<BR><HR>
<!--Table of Child-Links-->
<A NAME="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
<A ID="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
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<LI><A NAME="tex2html2492"
<LI><A ID="tex2html2492"
HREF="node129.html">Low-pass and high-pass filters</A>
<LI><A NAME="tex2html2493"
<LI><A ID="tex2html2493"
HREF="node130.html">Band-pass and stop-band filters</A>
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<LI><A ID="tex2html2494"
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<H2><A NAME="SECTION001211000000000000000">
<H2><A ID="SECTION001211000000000000000">
Low-pass and high-pass filters</A>
</H2>
@ -73,13 +73,13 @@ Low-pass and high-pass filters</A>
By far the most frequent purpose for using a filter is extracting either
the low-frequency or the high-frequency portion of an audio signal, attenuating
the rest. This is accomplished using a
<A NAME="10062"></A><A NAME="10063"></A><I>low-pass</I> or
<A NAME="10065"></A><A NAME="10066"></A><I>high-pass</I>
<A ID="10062"></A><A ID="10063"></A><I>low-pass</I> or
<A ID="10065"></A><A ID="10066"></A><I>high-pass</I>
filter.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.02"></A><A NAME="10070"></A>
<DIV ALIGN="CENTER"><A ID="fig08.02"></A><A ID="10070"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.2:</STRONG>
Terminology for describing the frequency response of low-pass and
@ -104,7 +104,7 @@ and computation time we put into it, the closer we can get.
Figure <A HREF="#fig08.02">8.2</A> shows the frequency response of a low-pass
filter. Frequency is divided into three bands, labeled on the
horizontal axis. The
<A NAME="10074"></A><I>passband</I>
<A ID="10074"></A><I>passband</I>
is the region (frequency band) where the filter should pass its input through
to its output with unit gain.
For a low-pass filter (as shown), the passband reaches from a frequency of
@ -113,18 +113,18 @@ would appear on the right-hand side of the graph and would extend from the
frequency limit up to the highest frequency possible. Any
realizable filter's passband will be only approximately flat;
the deviation from flatness is called the
<A NAME="10076"></A><I>ripple</I>,
<A ID="10076"></A><I>ripple</I>,
and is often specified by giving the ratio between the highest and lowest gain
in the passband, expressed in decibels. The ideal low-pass or high-pass filter
would have a ripple of 0 dB.
<P>
The
<A NAME="10078"></A><I>stopband</I>
<A ID="10078"></A><I>stopband</I>
of a low-pass or high-pass filter is the frequency
band over which the filter is intended not to transmit its input.
The
<A NAME="10080"></A><I>stopband attenuation</I>
<A ID="10080"></A><I>stopband attenuation</I>
is the difference, in decibels, between the lowest gain in the passband
and the highest gain in the stopband. Ideally this would
be infinite; the higher the better.
@ -134,42 +134,42 @@ Finally, a realizable filter, whose frequency response is always a
continuous function of frequency, must have a frequency
band over which the gain drops from the passband gain to the stopband
gain; this is called the
<A NAME="10082"></A><I>transition band</I>.
<A ID="10082"></A><I>transition band</I>.
The thinner this band can be made, the more nearly ideal the filter.
<P>
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<H1><A NAME="SECTION00560000000000000000"></A>
<A NAME="sect1.combine"></A>
<H1><A ID="SECTION00560000000000000000"></A>
<A ID="sect1.combine"></A>
<BR>
Superposing Signals
</H1>
@ -153,7 +153,7 @@ If we fix a window from <IMG
SRC="img128.png"
ALT="$N+M-1$"> as usual, we can write out the
mean power of the sum of two signals:
<A NAME="eq-meanpowersum"></A>
<A ID="eq-meanpowersum"></A>
<BR><P></P>
<DIV ALIGN="CENTER">
<!-- MATH
@ -174,7 +174,7 @@ P \{x[n] + y[n]\} = P \{x[n]\} + P \{y[n]\}
<BR CLEAR="ALL">
<P></P>
where we have introduced the
<A NAME="1161"></A><I>covariance</I> of two signals:
<A ID="1161"></A><I>covariance</I> of two signals:
<BR><P></P>
<DIV ALIGN="CENTER">
<!-- MATH
@ -276,36 +276,36 @@ amplitude <IMG
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<H2><A NAME="SECTION001212000000000000000"></A>
<A NAME="sect8.bandpassdef"></A>
<H2><A ID="SECTION001212000000000000000"></A>
<A ID="sect8.bandpassdef"></A>
<BR>
Band-pass and stop-band filters
</H2>
<P>
A
<A NAME="10086"></A><A NAME="10087"></A><I>band-pass filter</I>
<A ID="10086"></A><A ID="10087"></A><I>band-pass filter</I>
admits frequencies within a given band, rejecting frequencies below it and
above it. Figure <A HREF="#fig08.03">8.3</A> shows the frequency response of a band-pass
filter, with the key parameters labelled. A stop-band filter
@ -82,7 +82,7 @@ frequencies outside it.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.03"></A><A NAME="10092"></A>
<DIV ALIGN="CENTER"><A ID="fig08.03"></A><A ID="10092"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.3:</STRONG>
Terminology for describing the frequency response of band-pass and
@ -99,9 +99,9 @@ contiguous stopband surrounded by two passbands.</CAPTION>
<P>
In practice, a simpler language is often used for describing bandpass filters,
as shown in Figure <A HREF="#fig08.04">8.4</A>. Here there are only two parameters: a
<A NAME="10096"></A><I>center frequency</I>
<A ID="10096"></A><I>center frequency</I>
and a
<A NAME="10098"></A><I>bandwidth</I>.
<A ID="10098"></A><I>bandwidth</I>.
The passband is considered to be the region where the filter has at least half
the power gain as at the peak (i.e., the gain is within 3 decibels of its
maximum). The bandwidth is the width, in frequency units, of the passband.
@ -110,7 +110,7 @@ midpoint of the passband.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.04"></A><A NAME="10102"></A>
<DIV ALIGN="CENTER"><A ID="fig08.04"></A><A ID="10102"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.4:</STRONG>
A simplified view of a band-pass filter, showing bandwidth and
@ -125,36 +125,36 @@ center frequency.</CAPTION>
<P>
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<H2><A NAME="SECTION001213000000000000000"></A>
<A NAME="sect8.equalizing"></A>
<H2><A ID="SECTION001213000000000000000"></A>
<A ID="sect8.equalizing"></A>
<BR>
Equalizing filters
</H2>
<P>
In some applications, such as
<A NAME="10107"></A><I>equalization</I>,
<A ID="10107"></A><I>equalization</I>,
the goal isn't to pass signals of certain frequencies while stopping others
altogether, but to make controllable adjustments, boosting or attenuating
a signal, over a frequency range, by a desired gain. Two filter types
are useful for this. First, a
<A NAME="10109"></A><A NAME="10110"></A><I>shelving filter</I>
<A ID="10109"></A><A ID="10110"></A><I>shelving filter</I>
(Figure <A HREF="#fig08.05">8.5</A>) is used for selectively boosting or reducing either the
low or high end of the frequency range. Below a selectable crossover frequency,
the filter tends toward a low-frequency gain, and above it it tends toward a
@ -86,7 +86,7 @@ and high-frequency gain can all be adjusted independently.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.05"></A><A NAME="10115"></A>
<DIV ALIGN="CENTER"><A ID="fig08.05"></A><A ID="10115"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.5:</STRONG>
A shelving filter, showing low and high frequency gain, and
@ -100,7 +100,7 @@ crossover frequency.</CAPTION>
<P>
Second, a
<A NAME="10118"></A><A NAME="10119"></A><I>peaking filter</I>
<A ID="10118"></A><A ID="10119"></A><I>peaking filter</I>
(Figure <A HREF="#fig08.06">8.6</A>) is capable of boosting or attenuating signals within
a range of frequencies. The center frequency and bandwidth (which together
control the range of frequencies affected), and the in-band and out-of-band
@ -108,7 +108,7 @@ gains are separately adjustible.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.06"></A><A NAME="10124"></A>
<DIV ALIGN="CENTER"><A ID="fig08.06"></A><A ID="10124"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.6:</STRONG>
A peaking filter, with controllable center frequency, bandwidth,
@ -128,36 +128,36 @@ adjust bands in between.
<P>
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<H1><A NAME="SECTION001220000000000000000">
<H1><A ID="SECTION001220000000000000000">
Elementary filters</A>
</H1>
@ -103,55 +103,55 @@ filters--with complex-valued gains.
<P>
<BR><HR>
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<A ID="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
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HREF="node133.html">Elementary non-recirculating filter</A>
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HREF="node135.html">Elementary recirculating filter</A>
<LI><A NAME="tex2html2552"
<LI><A ID="tex2html2552"
HREF="node136.html">Compound filters</A>
<LI><A NAME="tex2html2553"
<LI><A ID="tex2html2553"
HREF="node137.html">Real outputs from complex filters</A>
<LI><A NAME="tex2html2554"
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<H2><A NAME="SECTION001221000000000000000"></A>
<A NAME="sect8.nonrecirculating"></A>
<H2><A ID="SECTION001221000000000000000"></A>
<A ID="sect8.nonrecirculating"></A>
<BR>
Elementary non-recirculating filter
</H2>
@ -75,7 +75,7 @@ Elementary non-recirculating filter
The non-recirculating comb
filter may be generalized to yield the design shown in Figure <A HREF="#fig08.07">8.7</A>.
This is the
<A NAME="10133"></A><A NAME="10134"></A><I>elementary non-recirculating filter</I>,
<A ID="10133"></A><A ID="10134"></A><I>elementary non-recirculating filter</I>,
of the first form. Its single, complex-valued parameter <IMG
WIDTH="16" HEIGHT="30" ALIGN="MIDDLE" BORDER="0"
SRC="img42.png"
@ -84,7 +84,7 @@ complex gain of the delayed signal subtracted from the original one.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.07"></A><A NAME="10680"></A>
<DIV ALIGN="CENTER"><A ID="fig08.07"></A><A ID="10680"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.7:</STRONG>
A delay network with a single-sample delay and a complex
@ -250,7 +250,7 @@ plane, which is equal to the distance from <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.08"></A><A NAME="10688"></A>
<DIV ALIGN="CENTER"><A ID="fig08.08"></A><A ID="10688"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.8:</STRONG>
Diagram for calculating the frequency response of the
@ -300,7 +300,7 @@ function for three different values of <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.09"></A><A NAME="10689"></A>
<DIV ALIGN="CENTER"><A ID="fig08.09"></A><A ID="10689"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.9:</STRONG>
Frequency response of the elementary non-recirculating filter
@ -322,36 +322,36 @@ same argument (-2 radians), but with varying absolute value (magnitude) <IMG
<P>
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<H2><A NAME="SECTION001222000000000000000"></A>
<A NAME="sect8.secondform"></A>
<H2><A ID="SECTION001222000000000000000"></A>
<A ID="sect8.secondform"></A>
<BR>
Non-recirculating filter, second form
</H2>
@ -80,7 +80,7 @@ Instead of multiplying the delay output by <IMG
SRC="img42.png"
ALT="$Q$"> we multiply the direct signal
by its
<A NAME="10166"></A><I>complex conjugate</I>
<A ID="10166"></A><I>complex conjugate</I>
<IMG
WIDTH="16" HEIGHT="36" ALIGN="MIDDLE" BORDER="0"
SRC="img872.png"
@ -180,7 +180,7 @@ occasionally lead us to prefer the second form.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.10"></A><A NAME="10180"></A>
<DIV ALIGN="CENTER"><A ID="fig08.10"></A><A ID="10180"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.10:</STRONG>
The elementary non-recirculating filter, second form.</CAPTION>
@ -194,36 +194,36 @@ The elementary non-recirculating filter, second form.</CAPTION>
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<H2><A NAME="SECTION001223000000000000000"></A>
<A NAME="sect8.recirculating"></A>
<H2><A ID="SECTION001223000000000000000"></A>
<A ID="sect8.recirculating"></A>
<BR>
Elementary recirculating filter
</H2>
<P>
The
<A NAME="10185"></A><A NAME="10186"></A><I>elementary recirculating filter</I> is the recirculating comb filter of
<A ID="10185"></A><A ID="10186"></A><I>elementary recirculating filter</I> is the recirculating comb filter of
Figure <A HREF="node109.html#fig07.07">7.7</A> with a
complex-valued feedback gain <IMG
WIDTH="15" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
@ -133,7 +133,7 @@ output grows exponentially as the delayed sample recirculates.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.11"></A><A NAME="10197"></A>
<DIV ALIGN="CENTER"><A ID="fig08.11"></A><A ID="10197"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.11:</STRONG>
The elementary recirculating filter: (a) block diagram; (b)
@ -158,36 +158,36 @@ working out the impulse response of the combined network).
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<H2><A NAME="SECTION001224000000000000000">
<H2><A ID="SECTION001224000000000000000">
Compound filters</A>
</H2>
<P>
We can use the recirculating and non-recirculating filters developed here to
create a
<A NAME="10201"></A><A NAME="10202"></A><I>compound filter</I> by putting several elementary ones in series. If the parameters
<A ID="10201"></A><A ID="10202"></A><I>compound filter</I> by putting several elementary ones in series. If the parameters
of the non-recirculating ones (of the first type) are <!-- MATH
${Q_1}, \ldots, {Q_j}$
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<H2><A ID="SECTION001225000000000000000">
Real outputs from complex filters</A>
</H2>
@ -246,36 +246,36 @@ either real-valued, or else appears in a pair with its complex conjugate.
<P>
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Two recirculating filters for the price of one</A>
</H2>
@ -297,36 +297,36 @@ explicitly.
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<H1><A NAME="SECTION001230000000000000000">
<H1><A ID="SECTION001230000000000000000">
Designing filters</A>
</H1>
@ -109,7 +109,7 @@ and each of the <IMG
SRC="img899.png"
ALT="$P_i$"> with an "x" (a "pole"); their names are borrowed
from the field of complex analysis. A plot showing the poles and zeroes
associated with a filter is unimaginatively called a <A NAME="10312"></A><I>pole-zero plot</I>.
associated with a filter is unimaginatively called a <A ID="10312"></A><I>pole-zero plot</I>.
<P>
When <IMG
@ -151,65 +151,65 @@ processing.
<P>
<BR><HR>
<!--Table of Child-Links-->
<A NAME="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
<A ID="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
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HREF="node141.html">One-pole, one-zero high-pass filter</A>
<LI><A NAME="tex2html2653"
<LI><A ID="tex2html2653"
HREF="node142.html">Shelving filter</A>
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<LI><A NAME="tex2html2659"
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