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<LI><A NAME="tex2html215"
<LI><A ID="tex2html215"
HREF="node5.html">Foreword</A>
<LI><A NAME="tex2html216"
<LI><A ID="tex2html216"
HREF="node6.html">Preface</A>
<LI><A NAME="tex2html217"
<LI><A ID="tex2html217"
HREF="node7.html">Sinusoids, amplitude and frequency</A>
<UL>
<LI><A NAME="tex2html218"
<LI><A ID="tex2html218"
HREF="node8.html">Measures of Amplitude</A>
<LI><A NAME="tex2html219"
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HREF="node9.html">Units of Amplitude</A>
<LI><A NAME="tex2html220"
<LI><A ID="tex2html220"
HREF="node10.html">Controlling Amplitude</A>
<LI><A NAME="tex2html221"
<LI><A ID="tex2html221"
HREF="node11.html">Frequency</A>
<LI><A NAME="tex2html222"
<LI><A ID="tex2html222"
HREF="node12.html">Synthesizing a sinusoid</A>
<LI><A NAME="tex2html223"
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<LI><A NAME="tex2html224"
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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="tex2html228"
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<LI><A NAME="tex2html229"
<LI><A ID="tex2html229"
HREF="node19.html">Constant amplitude scaler</A>
<LI><A NAME="tex2html230"
<LI><A ID="tex2html230"
HREF="node20.html">Amplitude control in decibels</A>
<LI><A NAME="tex2html231"
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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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<LI><A NAME="tex2html235"
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HREF="node26.html">Wavetables and samplers</A>
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<LI><A NAME="tex2html237"
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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="node35.html">Using a wavetable as a sampler</A>
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HREF="node71.html">Additive synthesis: Risset's bell</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="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>
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HREF="node111.html">Artificial reverberation</A>
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<LI><A ID="tex2html333"
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HREF="node127.html">Filters</A>
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HREF="node134.html">Non-recirculating filter, second form</A>
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HREF="node135.html">Elementary recirculating filter</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 ID="tex2html350"
HREF="node140.html">One-pole low-pass filter</A>
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HREF="node142.html">Shelving filter</A>
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<LI><A ID="tex2html353"
HREF="node143.html">Band-pass filter</A>
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<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>
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<LI><A ID="tex2html362"
HREF="node152.html">Subtractive synthesis</A>
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HREF="node159.html">Single sideband modulation</A>
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<H1><A NAME="SECTION00530000000000000000">
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Controlling Amplitude</A>
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@@ -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>
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@@ -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>
<DIV ALIGN="CENTER"><A ID="fig06.17"></A><A ID="7005"></A>
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<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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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>
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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>
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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.
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<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>.
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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.
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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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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.
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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.
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<H1><A NAME="SECTION0011100000000000000000">
<H1><A ID="SECTION0011100000000000000000">
Examples</A>
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<P>
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HREF="node118.html">Recirculating comb filter</A>
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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>
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HREF="node121.html">Order of execution in non-recirculating delay lines</A>
<LI><A NAME="tex2html2289"
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HREF="node122.html">Non-recirculating comb filter as octave doubler</A>
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HREF="node123.html">Time-varying complex comb filter: shakers</A>
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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.
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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.
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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>
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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,
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<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.
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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>
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@@ -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>
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@@ -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.
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<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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<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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<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-->
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<A ID="CHILD_LINKS"><STRONG>Subsections</STRONG></A>
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<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"
HREF="node131.html">Equalizing filters</A>
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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.
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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>
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HREF="node136.html">Compound filters</A>
<LI><A NAME="tex2html2553"
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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
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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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Real outputs from complex filters</A>
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@@ -246,36 +246,36 @@ either real-valued, or else appears in a pair with its complex conjugate.
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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">
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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"
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HREF="node142.html">Shelving filter</A>
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<H1><A NAME="SECTION00570000000000000000"></A>
<A NAME="sect1.fourier"></A>
<H1><A ID="SECTION00570000000000000000"></A>
<A ID="sect1.fourier"></A>
<BR>
Periodic Signals
</H1>
@@ -105,7 +105,7 @@ the smallest <IMG
WIDTH="12" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img135.png"
ALT="$\tau$"> if any at which a signal repeats is called the signal's
<A NAME="1175"></A><I>period</I>.
<A ID="1175"></A><I>period</I>.
In discussing periods of digital audio signals, we quickly run into the
difficulty of describing signals whose "period" isn't an integer, so that the
equation above doesn't make sense. For now we'll effectively
@@ -149,7 +149,7 @@ with frequencies <!-- MATH
SRC="img138.png"
ALT="$2 \pi / \omega$"> samples.
Such a sum is called a <I>Fourier Series</I>:
<A NAME="eq-fourierseries"></A>
<A ID="eq-fourierseries"></A>
<BR><P></P>
<DIV ALIGN="CENTER">
<!-- MATH
@@ -184,12 +184,12 @@ multiples of <IMG
WIDTH="14" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img27.png"
ALT="$\omega $">. They are called the
<A NAME="1187"></A><I>harmonics</I>
<A ID="1187"></A><I>harmonics</I>
of <IMG
WIDTH="14" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img27.png"
ALT="$\omega $">, which in turn is called the
<A NAME="1189"></A><I>fundamental</I>. In terms of pitch, the harmonics
<A ID="1189"></A><I>fundamental</I>. In terms of pitch, the harmonics
<!-- MATH
$\omega, 2 \omega, \ldots$
-->
@@ -205,7 +205,7 @@ pitches of the same scale; the third and sixth miss only by 2 cents and the
fifth misses by 14.
<P>
<A NAME="r-intervals"></A>Put another way, the frequency ratio 3:2 (a perfect fifth in Western
<A ID="r-intervals"></A>Put another way, the frequency ratio 3:2 (a perfect fifth in Western
terminology) is almost exactly
seven half-steps, 4:3 (a perfect fourth) is just as near to five half-steps,
and the ratios 5:4 and 6:5 (perfect major and minor thirds) are fairly close to
@@ -221,7 +221,7 @@ by the three component sinusoids.
<P>
<DIV ALIGN="CENTER"><A NAME="fig01.08"></A><A NAME="1196"></A>
<DIV ALIGN="CENTER"><A ID="fig01.08"></A><A ID="1196"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 1.8:</STRONG>
A Fourier series, showing three sinusoids and their sum. The
@@ -242,7 +242,7 @@ block diagram for doing this.
<P>
<DIV ALIGN="CENTER"><A NAME="fig01.09"></A><A NAME="1202"></A>
<DIV ALIGN="CENTER"><A ID="fig01.09"></A><A ID="1202"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 1.9:</STRONG>
Using many oscillators to synthesize a waveform with desired harmonic
@@ -256,43 +256,43 @@ amplitudes.</CAPTION>
<P>
This is an example of
<A NAME="1205"></A><I>additive synthesis</I>; more generally the term can be applied to networks
<A ID="1205"></A><I>additive synthesis</I>; more generally the term can be applied to networks
in which the frequencies of the oscillators are independently controllable.
The early days of computer music rang with the sound of additive synthesis.
<P>
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<A NAME="sect08.low-pass"></A>
<H2><A ID="SECTION001231000000000000000"></A>
<A ID="sect08.low-pass"></A>
<BR>
One-pole low-pass filter
</H2>
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.12"></A><A NAME="10320"></A>
<DIV ALIGN="CENTER"><A ID="fig08.12"></A><A ID="10320"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.12:</STRONG>
One-pole low-pass filter: (a) pole-zero diagram; (b)
@@ -164,36 +164,36 @@ cycles per second.
<P>
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<H2><A NAME="SECTION001232000000000000000"></A>
<A NAME="sect08.highpass"></A>
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<A ID="sect08.highpass"></A>
<BR>
One-pole, one-zero high-pass filter
</H2>
@@ -78,7 +78,7 @@ Section <A HREF="node78.html#sect5.waveshaping">5.3</A> almost always contain a
This is inaudible, but, since it specifies electrical power that is sent
to your speakers, its presence reduces the level of loudness you can
reach without distortion. Another name for a constant signal component is
<A NAME="10330"></A>"DC", meaning "direct current".
<A ID="10330"></A>"DC", meaning "direct current".
<P>
An easy and practical way to remove the zero-frequency component from an audio
@@ -135,7 +135,7 @@ in the previous example we plotted it all the way up to the sample rate,
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.13"></A><A NAME="10338"></A>
<DIV ALIGN="CENTER"><A ID="fig08.13"></A><A ID="10338"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.13:</STRONG>
One-pole, one-zero high-pass filter: (a) pole-zero diagram; (b)
@@ -150,36 +150,36 @@ frequency response (from zero to Nyquist frequency).</CAPTION>
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<H2><A NAME="SECTION001233000000000000000"></A>
<A NAME="sect8.shelving"></A>
<H2><A ID="SECTION001233000000000000000"></A>
<A ID="sect8.shelving"></A>
<BR>
Shelving filter
</H2>
@@ -96,7 +96,7 @@ diagrammed in Figure <A HREF="#fig08.14">8.14</A>.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.14"></A><A NAME="10346"></A>
<DIV ALIGN="CENTER"><A ID="fig08.14"></A><A ID="10346"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.14:</STRONG>
One-pole, one-zero shelving filter: (a) pole-zero diagram; (b)
@@ -211,36 +211,36 @@ as desired. For example, in the figure, <IMG
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<A NAME="sect8.twopolebandpass"></A>
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Band-pass filter
</H2>
@@ -146,7 +146,7 @@ The resulting pole-zero plot is as shown in Figure <A HREF="#fig08.15">8.15</A>.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.15"></A><A NAME="10361"></A>
<DIV ALIGN="CENTER"><A ID="fig08.15"></A><A ID="10361"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.15:</STRONG>
Two-pole band-pass filter: (a) pole-zero diagram; (b)
@@ -206,36 +206,36 @@ so that the gain drops to zero at angular frequencies <IMG
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Peaking and stop-band filter
</H2>
@@ -126,7 +126,7 @@ to the one-pole, one-zero high-pass filter above.
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<DIV ALIGN="CENTER"><A NAME="fig08.16"></A><A NAME="10371"></A>
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<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.16:</STRONG>
A peaking filter: (a) pole-zero diagram; (b)

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Butterworth filters</A>
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@@ -86,7 +86,7 @@ whose two bands are flatter and separated by a narrower transition region.
<P>
A procedure borrowed from the analog filtering world transforms real,
one-pole, one-zero filters to corresponding
<A NAME="10375"></A>
<A ID="10375"></A>
<I>Butterworth filters</I>,
which have narrower transition regions. This procedure is described clearly
and elegantly in the last chapter of [<A
@@ -259,7 +259,7 @@ which, after some algebra, gives an angular frequency equal to
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<DIV ALIGN="CENTER"><A NAME="fig08.17"></A><A NAME="10393"></A>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.17:</STRONG>
Replacing a real-valued pole or zero (shown as a solid dot) with an
@@ -300,7 +300,7 @@ becomes a shelving filter and then a high-pass one.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.18"></A><A NAME="10399"></A>
<DIV ALIGN="CENTER"><A ID="fig08.18"></A><A ID="10399"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.18:</STRONG>
Butterworth low-pass filter with three poles
@@ -338,36 +338,36 @@ unit gain at the Nyquist frequency, and the others for unit gain at DC. </CAPTIO
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Stretching the unit circle with rational functions</A>
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@@ -421,7 +421,7 @@ points of highest gain for the new filter.
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<DIV ALIGN="CENTER"><A NAME="fig08.19"></A><A NAME="10765"></A>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.19:</STRONG>
One-pole, one-zero low-pass filter: (a) pole-zero plot; (b)
@@ -444,36 +444,36 @@ result is a band-pass filter with center frequency <IMG
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Butterworth band-pass filter</A>
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@@ -402,7 +402,7 @@ are shown in Figure <A HREF="#fig08.20">8.20</A>.
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.20:</STRONG>
Butterworth band-pass filter: (a) pole-zero diagram; (b)
@@ -422,36 +422,36 @@ low-pass filter used.</CAPTION>
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Time-varying coefficients
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@@ -96,7 +96,7 @@ factor of two.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.21"></A><A NAME="10472"></A>
<DIV ALIGN="CENTER"><A ID="fig08.21"></A><A ID="10472"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.21:</STRONG>
Normalizing a recirculating elementary filter:
@@ -132,36 +132,36 @@ is needed.
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@@ -116,7 +116,7 @@ where <IMG
SRC="img977.png"
ALT="$e$"> is Euler's constant, about
2.718. The filter can be said to have a
<A NAME="10483"></A><I>settling time</I> of <IMG
<A ID="10483"></A><I>settling time</I> of <IMG
WIDTH="13" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img75.png"
ALT="$n$"> samples. In the figure, <IMG
@@ -131,7 +131,7 @@ over the cutoff frequency (in angular units).
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.22"></A><A NAME="10487"></A>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.22:</STRONG>
The impulse response of three elementary recirculating (one-pole)
@@ -223,43 +223,43 @@ where <IMG
WIDTH="11" HEIGHT="29" ALIGN="MIDDLE" BORDER="0"
SRC="img592.png"
ALT="$q$"> is the
<A NAME="10499"></A><I>quality</I> of the filter, defined as the center frequency divided by
<A ID="10499"></A><I>quality</I> of the filter, defined as the center frequency divided by
bandwidth. Resonant filters are often specified in terms of the center
frequency and "q" in place of bandwidth.
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@@ -106,47 +106,47 @@ patches from one to the other, but the two aren't truly compatible.
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All-pass filters</A>
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@@ -85,7 +85,7 @@ introduces a phase change of <IMG
SRC="img27.png"
ALT="$\omega $">.
Another class of filters, called
<A NAME="10502"></A><I>all-pass filters</I>,
<A ID="10502"></A><I>all-pass filters</I>,
can make phase changes which are more interesting functions of <IMG
WIDTH="14" HEIGHT="13" ALIGN="BOTTOM" BORDER="0"
SRC="img27.png"
@@ -171,7 +171,7 @@ upward or downward depending on the coefficient.
<P>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.23:</STRONG>
Phase response of all-pass filters with different pole locations <IMG
@@ -197,36 +197,36 @@ argument of the coefficient <IMG
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<H2><A NAME="SECTION001241000000000000000"></A>
<A NAME="sect08.subtractive"></A>
<H2><A ID="SECTION001241000000000000000"></A>
<A ID="sect08.subtractive"></A>
<BR>
Subtractive synthesis
</H2>
<P>
<A NAME="10517"></A><I>Subtractive synthesis</I> is the technique of using filters to shape the
<A ID="10517"></A><I>Subtractive synthesis</I> is the technique of using filters to shape the
spectral envelope of a sound, forming another sound, usually preserving
qualities of the original sound such as pitch, roughness, noisiness, or
graniness. The spectral envelope of the resulting sound is the product of the
@@ -82,7 +82,7 @@ filter, and result.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.24"></A><A NAME="10522"></A>
<DIV ALIGN="CENTER"><A ID="fig08.24"></A><A ID="10522"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.24:</STRONG>
Subtractive synthesis: (a) spectrum of input sound; (b) filter frequency
@@ -113,7 +113,7 @@ brass instruments over the life of a note.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.25"></A><A NAME="10528"></A>
<DIV ALIGN="CENTER"><A ID="fig08.25"></A><A ID="10528"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.25:</STRONG>
ADSR-controlled subtractive synthesis.</CAPTION>
@@ -127,36 +127,36 @@ ADSR-controlled subtractive synthesis.</CAPTION>
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<A NAME="sect8.envelopefollower"></A>
<H2><A ID="SECTION001242000000000000000"></A>
<A ID="sect8.envelopefollower"></A>
<BR>
Envelope following
</H2>
@@ -81,7 +81,7 @@ power over an interval of time long enough that its variations won't
show up in the power estimate, but short enough that changes in signal level
are quickly reported. A computation that provides a
time-varying power estimate of a signal is called an
<A NAME="10533"></A><I>envelope follower</I>.
<A ID="10533"></A><I>envelope follower</I>.
<P>
The output of a low-pass filter can be viewed as a moving average of its input.
@@ -133,7 +133,7 @@ the moving average.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.26"></A><A NAME="10539"></A>
<DIV ALIGN="CENTER"><A ID="fig08.26"></A><A ID="10539"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.26:</STRONG>
Envelope follower. The output is the average power of the input
@@ -232,7 +232,7 @@ estimates the average power.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.27"></A><A NAME="10551"></A>
<DIV ALIGN="CENTER"><A ID="fig08.27"></A><A ID="10551"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.27:</STRONG>
Envelope following from the spectral point of view: (a) an
@@ -260,36 +260,36 @@ getting a quick response and a smooth result.
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<A NAME="sect8.singlesideband"></A>
<H2><A ID="SECTION001243000000000000000"></A>
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<BR>
Single Sideband Modulation
</H2>
@@ -362,36 +362,36 @@ only positive frequencies.
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Examples</A>
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@@ -78,20 +78,20 @@ that require specially designed filters.
<P>
<BR><HR>
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Prefabricated low-, high-, and band-pass filters</A>
</H2>
@@ -84,7 +84,7 @@ We will need four new Pd objects:
WIDTH="47" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1014.png"
ALT="\fbox{\texttt{lop\~}}">:
<A NAME="10842"></A>one-pole low-pass filter. The left inlet takes a signal to be filtered, and
<A ID="10842"></A>one-pole low-pass filter. The left inlet takes a signal to be filtered, and
the right inlet takes control messages to set the
cutoff frequency of the filter. The filter is normalized so that the
gain is one at frequency 0.
@@ -97,7 +97,7 @@ gain is one at frequency 0.
WIDTH="53" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1016.png"
ALT="\fbox{ \texttt{hip\~}}">:
<A NAME="10843"></A>one-pole, one-zero high-pass filter, with the same inputs and outputs as
<A ID="10843"></A>one-pole, one-zero high-pass filter, with the same inputs and outputs as
<TT>lop~</TT>, normalized to have a gain of one at the Nyquist frequency.
<P>
@@ -108,7 +108,7 @@ gain is one at frequency 0.
WIDTH="44" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1017.png"
ALT="\fbox{ \texttt{bp\~}}">:
<A NAME="10844"></A>resonant filter. The middle inlet takes control messages to set the center
<A ID="10844"></A>resonant filter. The middle inlet takes control messages to set the center
frequency, and the right inlet to set "q".
<P>
@@ -119,12 +119,12 @@ frequency, and the right inlet to set "q".
WIDTH="69" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1019.png"
ALT="\fbox{ \texttt{noise\~}}">:
<A NAME="10845"></A>white noise generator. Each sample is an independent
<A ID="10845"></A>white noise generator. Each sample is an independent
pseudo-random number, uniformly distributed from -1 to 1.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.28"></A><A NAME="10600"></A>
<DIV ALIGN="CENTER"><A ID="fig08.28"></A><A ID="10600"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.28:</STRONG>
Using prefabricated filters in Pd: (a) a low-pass filter, with
@@ -146,36 +146,36 @@ frequency) component of a signal.
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<H2><A ID="SECTION001252000000000000000">
Prefabricated time-varying band-pass filter</A>
</H2>
@@ -82,7 +82,7 @@ using the <TT>vcf~</TT> object, introduced here:
WIDTH="53" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1022.png"
ALT="\fbox{ \texttt{vcf\~}}">:
<A NAME="10846"></A>a "voltage controlled" band-pass filter,
<A ID="10846"></A>a "voltage controlled" band-pass filter,
similar to <TT>bp~</TT>, but with a signal inlet to control center frequency.
Both <TT>bp~</TT> and <TT>vcf~</TT> are one-pole resonant filters as
developed in Section <A HREF="node143.html#sect8.twopolebandpass">8.3.4</A>; <TT>bp~</TT> outputs only
@@ -91,7 +91,7 @@ real and imaginary parts separately.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.29"></A><A NAME="10612"></A>
<DIV ALIGN="CENTER"><A ID="fig08.29"></A><A ID="10612"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.29:</STRONG>
The <TT>vcf~</TT> band-pass filter, with its center frequency
@@ -119,36 +119,36 @@ subtractive synthesis is demonstrated in example H05.filter.floyd.pd.
<P>
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Envelope followers</A>
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<A NAME="sect8.heterodyning"></A>
<A ID="sect8.heterodyning"></A>
<P>
Example H06.envelope.follower.pd shows a simple and self-explanatory realization of the
envelope follower described in Section <A HREF="node153.html#sect8.envelopefollower">8.4.2</A>. An
@@ -84,7 +84,7 @@ strength.
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.30"></A><A NAME="10622"></A>
<DIV ALIGN="CENTER"><A ID="fig08.30"></A><A ID="10622"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.30:</STRONG>
Analyzing the spectrum of a sound: (a) band-pass filtering a sampled
@@ -113,36 +113,36 @@ analysis, the subject of Chapter 9.
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Single sideband modulation</A>
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<P>
<DIV ALIGN="CENTER"><A NAME="fig08.31"></A><A NAME="10628"></A>
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<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.31:</STRONG>
Using an all-pass filter network to

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<H2><A NAME="SECTION00581000000000000000">
<H2><A ID="SECTION00581000000000000000">
Quick Introduction to Pd</A>
</H2>
<P>
Pd documents are called
<A NAME="1211"></A><I>patches</I>. They correspond roughly to the
<A ID="1211"></A><I>patches</I>. They correspond roughly to the
boxes in the abstract block diagrams shown earlier in this chapter, but in
detail they
are quite different, because Pd is an implementation
@@ -80,7 +80,7 @@ environment, not a specification language.
<P>
A Pd patch, such as the ones shown in
Figure <A HREF="#fig01.10">1.10</A>, consists of a collection of
<A NAME="1214"></A><I>boxes</I>
<A ID="1214"></A><I>boxes</I>
connected in a network.
The border of a box tells you how its text is interpreted and how the box
functions. In part (a) of the figure we see three types of boxes. From
@@ -88,7 +88,7 @@ top to bottom they are:
<UL>
<LI>a
<A NAME="1217"></A><A NAME="1218"></A><I>message box</I>.
<A ID="1217"></A><A ID="1218"></A><I>message box</I>.
Message boxes, with a flag-shaped border, interpret the text as a message to
send whenever the box is
activated (by an incoming message or with a pointing device). The message in this
@@ -97,7 +97,7 @@ case consists simply of the number "21".
<P>
</LI>
<LI>an
<A NAME="1220"></A><A NAME="1221"></A><I>object box</I>.
<A ID="1220"></A><A ID="1221"></A><I>object box</I>.
Object boxes have a rectangular border; they interpret the text to create
objects
when you load a patch. Object boxes may hold hundreds of different
@@ -106,18 +106,18 @@ signal processing modules to be introduced later--depending on the text
inside. In this example, the box holds an adder. In most Pd patches, the
majority of boxes are of type "object". The first word typed into an object
box specifies its
<A NAME="1223"></A><I>class</I>,
<A ID="1223"></A><I>class</I>,
which in this case is just "+". Any additional (blank-space-separated) words
appearing in the box are called
<A NAME="1225"></A><A NAME="1226"></A><I>creation arguments</I>,
<A ID="1225"></A><A ID="1226"></A><I>creation arguments</I>,
which specify the initial state of the object when it is created.
<P>
</LI>
<LI>a
<A NAME="1228"></A><A NAME="1229"></A><I>number box</I>.
<A ID="1228"></A><A ID="1229"></A><I>number box</I>.
Number boxes are a particular type of
<A NAME="1231"></A><A NAME="1232"></A><I>GUI box</I>. Others include push buttons and
<A ID="1231"></A><A ID="1232"></A><I>GUI box</I>. Others include push buttons and
toggle switches; these will come up later in the examples.
The number box
has a punched-card-shaped border, with a nick out of its top right corner.
@@ -137,7 +137,7 @@ from one box to the next; outputs of boxes are on the bottom and inputs on top.
<P>
<DIV ALIGN="CENTER"><A NAME="fig01.10"></A><A NAME="1238"></A>
<DIV ALIGN="CENTER"><A ID="fig01.10"></A><A ID="1238"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 1.10:</STRONG>
(a) three types of boxes in Pd (message, object, and GUI); (b) a simple patch to output a
@@ -153,7 +153,7 @@ sinusoid.</CAPTION>
Figure <A HREF="#fig01.10">1.10</A> (part b) shows a Pd patch which
makes a sinusoid with controllable frequency and amplitude. The connecting
patch lines are of two types here; the thin ones are for carrying sporadic
<A NAME="1242"></A><I>messages</I>,
<A ID="1242"></A><I>messages</I>,
and the thicker ones (connecting the oscillator, the multiplier, and the output
<TT>dac~</TT> object) carry digital audio signals. Since Pd is a real-time
program, the audio signals flow in a continuous stream. On the other hand, the
@@ -173,36 +173,36 @@ and "<TT>osc~</TT>".
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<H2><A NAME="SECTION001255000000000000000">
<H2><A ID="SECTION001255000000000000000">
Using elementary filters directly: shelving and peaking</A>
</H2>
<P>
<DIV ALIGN="CENTER"><A NAME="fig08.32"></A><A NAME="10641"></A>
<DIV ALIGN="CENTER"><A ID="fig08.32"></A><A ID="10641"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.32:</STRONG>
Building filters from elementary, raw ones: (a) shelving; (b) peaking.
@@ -114,7 +114,7 @@ First we introduce the six Pd objects that realize elementary filters:
WIDTH="69" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1030.png"
ALT="\fbox{ \texttt{rpole\~}}">:
<A NAME="10847"></A><A NAME="10848"></A><A NAME="10849"></A>elementary filters with real-valued coefficients operating on real-valued
<A ID="10847"></A><A ID="10848"></A><A ID="10849"></A>elementary filters with real-valued coefficients operating on real-valued
signals. The three implement non-recirculating filters of the first and
second types, and the recirculating filter. They all have one inlet, at
right, to supply the coefficient that sets the location of the zero or pole.
@@ -144,7 +144,7 @@ audio signals. No stability check is performed.
WIDTH="69" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1036.png"
ALT="\fbox{ \texttt{cpole\~}}">:
<A NAME="10850"></A><A NAME="10851"></A><A NAME="10852"></A>elementary filters with complex-valued coefficients, operating on complex-valued
<A ID="10850"></A><A ID="10851"></A><A ID="10852"></A>elementary filters with complex-valued coefficients, operating on complex-valued
signals, corresponding to the real-valued ones above. Instead of two inlets and
one outlet, each of these filters has four inlets (real and imaginary part of
the signal to filter, and real and imaginary part of the coefficient) and
@@ -175,36 +175,36 @@ easy reuse.
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Making and using all-pass filters</A>
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<DIV ALIGN="CENTER"><A NAME="fig08.33"></A><A NAME="10672"></A>
<DIV ALIGN="CENTER"><A ID="fig08.33"></A><A ID="10672"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 8.33:</STRONG>
All-pass filters: (a) making an all-pass filter from elementary

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@@ -122,36 +122,36 @@ Hertz). Where would you place the pole? What is the value of "q"?
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Fourier analysis and resynthesis
</H1>
@@ -78,7 +78,7 @@ amplitude and phase of a sinusoidal component of a signal (Page
<A HREF="node158.html#sect8.heterodyning"><IMG ALIGN="BOTTOM" BORDER="1" ALT="[*]"
SRC="crossref.png"></A>). In this chapter we will refine this technique
into what is called
<A NAME="12298"></A><I>Fourier analysis</I>.
<A ID="12298"></A><I>Fourier analysis</I>.
In its simplest form, Fourier analysis takes as input any periodic
signal (of period <IMG
WIDTH="18" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
@@ -104,63 +104,63 @@ Finally, we will develop some standard applications such as the phase vocoder.
<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="tex2html2999"
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<LI><A NAME="tex2html3001"
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<BR>
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<UL>
<LI><A NAME="tex2html3003"
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<LI><A NAME="tex2html3004"
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<BR>
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HREF="node172.html">Fourier analysis and reconstruction of audio signals</A>
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<LI><A NAME="tex2html3008"
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<LI><A NAME="tex2html3009"
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HREF="node174.html">Timbre stamping (classical vocoder)</A>
</UL>
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Fourier analysis of periodic signals</A>
</H1>
@@ -325,7 +325,7 @@ averaging over a period:
<BR CLEAR="ALL">
<P></P>
This is such a useful formula that it gets its own notation. The
<A NAME="12325"></A><I>Fourier transform</I>
<A ID="12325"></A><I>Fourier transform</I>
of a signal <IMG
WIDTH="36" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img669.png"
@@ -412,47 +412,47 @@ frequency associated with the period <IMG
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Fourier transform as additive synthesis
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@@ -298,36 +298,36 @@ from its Fourier transform <IMG
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Properties of Fourier transforms</A>
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@@ -79,14 +79,14 @@ result to any other sinusoid.
<P>
<BR><HR>
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Fourier transform of DC</A>
</H2>
@@ -330,7 +330,7 @@ where <IMG
WIDTH="49" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img1090.png"
ALT="${D_N}(k)$">, known as the
<A NAME="12394"></A><I>Dirichlet kernel</I>,
<A ID="12394"></A><I>Dirichlet kernel</I>,
is defined as
<BR><P></P>
<DIV ALIGN="CENTER">
@@ -430,7 +430,7 @@ the <IMG
WIDTH="41" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img1092.png"
ALT="$k=0$">. This is surrounded by one-unit-wide
<A NAME="12409"></A><I>sidelobes</I>,
<A ID="12409"></A><I>sidelobes</I>,
alternating in sign and gradually decreasing in magnitude as <IMG
WIDTH="12" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img58.png"
@@ -461,7 +461,7 @@ radians over the peak from <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.01"></A><A NAME="12413"></A>
<DIV ALIGN="CENTER"><A ID="fig09.01"></A><A ID="12413"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.1:</STRONG>
The Fourier transform of a signal consisting of all ones. Here
@@ -480,7 +480,7 @@ page and the imaginary axis pointing away from it.</CAPTION>
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.02"></A><A NAME="12418"></A>
<DIV ALIGN="CENTER"><A ID="fig09.02"></A><A ID="12418"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.2:</STRONG>
The Dirichlet kernel, for <IMG
@@ -497,36 +497,36 @@ The Dirichlet kernel, for <IMG
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Shifts and phase changes
</H2>
@@ -246,7 +246,7 @@ transform of <IMG
<P></P>
(The third line is just the second one with the terms summed in a
different order). We therefore get the Time Shift Formula for Fourier
Transforms: <A NAME="12441"></A>
Transforms: <A ID="12441"></A>
<BR><P></P>
<DIV ALIGN="CENTER">
<!-- MATH
@@ -430,7 +430,7 @@ The Fourier transform is:
<BR CLEAR="ALL">
<P></P>
We therefore get the Phase Shift Formula for Fourier Transforms:
<A NAME="12464"></A>
<A ID="12464"></A>
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<!-- MATH
@@ -454,36 +454,36 @@ We therefore get the Phase Shift Formula for Fourier Transforms:
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How to find and run the examples</A>
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@@ -72,12 +72,12 @@ How to find and run the examples</A>
To run the patches, you must first download, install, and run Pd.
Instructions for
doing this appear in Pd's on-line HTML documentation, which you can find at
<A NAME="tex2html11"
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This book should appear at
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Fourier transform of a sinusoid</A>
</H2>
@@ -146,7 +146,7 @@ where <IMG
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.03"></A><A NAME="12484"></A>
<DIV ALIGN="CENTER"><A ID="fig09.03"></A><A ID="12484"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.3:</STRONG>
Fourier transforms of complex sinusoids, with <IMG
@@ -187,7 +187,7 @@ in Figure <A HREF="#fig09.03">9.3</A> (part a).
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.04"></A><A NAME="12490"></A>
<DIV ALIGN="CENTER"><A ID="fig09.04"></A><A ID="12490"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.4:</STRONG>
A complex sinusoid with frequency <!-- MATH
@@ -239,36 +239,36 @@ of frequencies.
<P>
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<H1><A ID="SECTION001330000000000000000">
Fourier analysis of non-periodic signals</A>
</H1>
@@ -110,12 +110,12 @@ Given a signal <IMG
ALT="$N-1$">,
the technique is to envelope the signal before doing the Fourier analysis.
The envelope shape is known as a
<A NAME="12499"></A><I>window function</I>.
<A ID="12499"></A><I>window function</I>.
Given a window function <IMG
WIDTH="33" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img1123.png"
ALT="$w[n]$">, the
<A NAME="12501"></A><I>windowed Fourier transform</I>
<A ID="12501"></A><I>windowed Fourier transform</I>
is:
<BR><P></P>
<DIV ALIGN="CENTER">
@@ -136,7 +136,7 @@ is:
<P></P>
Much ink has been spilled over the design of suitable window functions for
particular situations, but here we will consider the simplest one, named the
<A NAME="12504"></A><A NAME="12505"></A><I>Hann</I>
<A ID="12504"></A><A ID="12505"></A><I>Hann</I>
window function (the name is sometimes corrupted to "Hanning" in DSP circles).
The Hann window is:
<BR><P></P>
@@ -302,7 +302,7 @@ Dirichlet kernel components are also shown separately.
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.05"></A><A NAME="12536"></A>
<DIV ALIGN="CENTER"><A ID="fig09.05"></A><A ID="12536"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.5:</STRONG>
The magnitude M(k) of the Fourier transform of the Hann
@@ -450,7 +450,7 @@ Reducing the amplitude of the sidelobes reduces this interference.
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.06"></A><A NAME="12558"></A>
<DIV ALIGN="CENTER"><A ID="fig09.06"></A><A ID="12558"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.6:</STRONG>
The Hann-windowed Fourier transform of a signal with two
@@ -507,36 +507,36 @@ four-period rule may be relaxed to three or even slightly less.
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<H1><A NAME="SECTION001340000000000000000">
<H1><A ID="SECTION001340000000000000000">
Fourier analysis and reconstruction of audio signals</A>
</H1>
@@ -90,7 +90,7 @@ modified transform.
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.07"></A><A NAME="12565"></A>
<DIV ALIGN="CENTER"><A ID="fig09.07"></A><A ID="12565"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.7:</STRONG>
Sliding-window analysis and resynthesis of an audio signal using
@@ -107,7 +107,7 @@ Fourier transform with a desired frequency response.</CAPTION>
Figure <A HREF="#fig09.07">9.7</A> shows how to carry out a Fourier analysis, modification,
and reconstruction of an audio signal. The first step is to divide the
signal into
<A NAME="12569"></A><I>windows</I>,
<A ID="12569"></A><I>windows</I>,
which are segments of the signal, of <IMG
WIDTH="18" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img3.png"
@@ -137,7 +137,7 @@ Let <IMG
WIDTH="18" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img25.png"
ALT="$H$"> denote the
<A NAME="12571"></A><I>hop size</I>,
<A ID="12571"></A><I>hop size</I>,
the number of samples each window is advanced past the
previous window. Then for each <!-- MATH
$m = \ldots, 0, 1, \ldots$
@@ -310,47 +310,47 @@ to vary smoothly between the input and some transformed version of it.
<P>
<BR><HR>
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<H2><A NAME="SECTION001341000000000000000">
<H2><A ID="SECTION001341000000000000000">
Narrow-band companding</A>
</H2>
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.08"></A><A NAME="12579"></A>
<DIV ALIGN="CENTER"><A ID="fig09.08"></A><A ID="12579"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.8:</STRONG>
Block diagram for narrow-band noise suppression by companding.</CAPTION>
@@ -84,12 +84,12 @@ Block diagram for narrow-band noise suppression by companding.</CAPTION>
<P>
A
<A NAME="12582"></A><I>compander</I>
<A ID="12582"></A><I>compander</I>
is a tool that amplifies a signal with a variable gain, depending on the
signal's measured amplitude. The term is a contraction of "compressor" and
"expander". A compressor's gain decreases as the input level increases, so
that the
<A NAME="12584"></A><I>dynamic range</I>,
<A ID="12584"></A><I>dynamic range</I>,
that is, the overall variation in signal level, is reduced. An expander does
the reverse, increasing the dynamic range. Frequently the gain depends not only
on the immediate signal level but on its history; for instance the rate of change
@@ -132,7 +132,7 @@ simply a function of the magnitude of the current sample: <IMG
The patch diagrammed in Figure <A HREF="#fig09.08">9.8</A> shows one very useful
application of companding, called
a
<A NAME="12587"></A><I>noise gate</I>. Here the gain <IMG
<A ID="12587"></A><I>noise gate</I>. Here the gain <IMG
WIDTH="34" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img1159.png"
ALT="$g[m]$"> depends on the channel amplitude
@@ -257,36 +257,36 @@ undistorted.
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<H2><A NAME="SECTION001342000000000000000">
<H2><A ID="SECTION001342000000000000000">
Timbre stamping (classical vocoder)</A>
</H2>
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.09"></A><A NAME="12598"></A>
<DIV ALIGN="CENTER"><A ID="fig09.09"></A><A ID="12598"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.9:</STRONG>
Block diagram for timbre stamping (AKA "vocoding'').</CAPTION>
@@ -87,11 +87,11 @@ filter capable of making one sound take on the evolving spectral envelope of
another. This is
widely known in electronic music circles as
a
<A NAME="12601"></A><I>vocoder</I>,
<A ID="12601"></A><I>vocoder</I>,
named, not quite accurately, after the original
Bell Laboratories vocal analysis/synthesis device. The
technique described here is more accurately called
<A NAME="12603"></A><I>timbre stamping</I>.
<A ID="12603"></A><I>timbre stamping</I>.
Two input signals are used, one to be filtered, and the other to control the
filter via its time-varying spectral envelope. The windowed Fourier
transform is used both on the control signal input to estimate its spectral
@@ -136,36 +136,36 @@ suitable limiting function is applied to the gain before using it.
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<H1><A NAME="SECTION001350000000000000000"></A>
<A NAME="sect9.phase"></A>
<H1><A ID="SECTION001350000000000000000"></A>
<A ID="sect9.phase"></A>
<BR>
Phase
</H1>
@@ -82,7 +82,7 @@ time requires analyzing and reconstructing the phase.
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.10"></A><A NAME="12612"></A>
<DIV ALIGN="CENTER"><A ID="fig09.10"></A><A ID="12612"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.10:</STRONG>
Phase in windowed Fourier analysis: (a) a complex sinusoid analyzed
@@ -268,7 +268,7 @@ from somewhere else and can't be assumed to be in line with our wishes.
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.11"></A><A NAME="12631"></A>
<DIV ALIGN="CENTER"><A ID="fig09.11"></A><A ID="12631"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.11:</STRONG>
Propagating phases in resynthesis. Each phase, such as that of
@@ -286,7 +286,7 @@ input phases.</CAPTION>
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.12"></A><A NAME="12636"></A>
<DIV ALIGN="CENTER"><A ID="fig09.12"></A><A ID="12636"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.12:</STRONG>
Phases of one channel of the analysis windows and two successive
@@ -522,45 +522,45 @@ S[m, k] \; = \;
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<A NAME="sect9.phaserelationship"></A>
<H2><A ID="SECTION001351000000000000000"></A>
<A ID="sect9.phaserelationship"></A>
<BR>
Phase relationships between channels
</H2>
@@ -121,7 +121,7 @@ can alter the equation for <IMG
ALT="$S[m, k]$"> above so that whenever there happens to be
a solution to the over-constrained system of equations, the reconstruction
algorithm homes in on the solution. This approach is called
<A NAME="12657"></A><I>phase locking</I>
<A ID="12657"></A><I>phase locking</I>
[<A
HREF="node202.html#r-puckette95b">Puc95b</A>], and has the virtue of simplicity although more
sophisticated techniques are available [<A
@@ -374,36 +374,36 @@ same phase, as long as the phase relationships between the measured spectra
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Phase bashing
</H1>
@@ -88,7 +88,7 @@ corrected using Fourier analysis and resynthesis [<A
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.13"></A><A NAME="12680"></A>
<DIV ALIGN="CENTER"><A ID="fig09.13"></A><A ID="12680"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.13:</STRONG>
Phase-bashing a recorded sound (here, a sinusoid with rising
@@ -130,36 +130,36 @@ powerful tools for vocal and other imitative synthesis.
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Examples</A>
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<H2><A NAME="SECTION001371000000000000000">
<H2><A ID="SECTION001371000000000000000">
Fourier analysis and resynthesis in Pd</A>
</H2>
@@ -75,7 +75,7 @@ Fourier transform of an audio signal using the <TT>fft~</TT> object:
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.14"></A><A NAME="12858"></A>
<DIV ALIGN="CENTER"><A ID="fig09.14"></A><A ID="12858"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure:</STRONG>
Fourier analysis in Pd: (a) the <!-- MATH
@@ -103,7 +103,7 @@ Hann windowing function.</CAPTION>
WIDTH="47" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1216.png"
ALT="\fbox{\texttt{fft\~}}">:
<A NAME="12898"></A><A NAME="12696"></A>Fast Fourier transform. The two inlets take audio signals representing the real
<A ID="12898"></A><A ID="12696"></A>Fast Fourier transform. The two inlets take audio signals representing the real
and imaginary parts of a complex-valued signal. The window size <IMG
WIDTH="18" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img3.png"
@@ -134,7 +134,7 @@ subwindow which in turn is shown in part (c). New objects are:
WIDTH="56" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1218.png"
ALT="\fbox{\texttt{rfft\~}}">:
<A NAME="12899"></A>real Fast Fourier transform. The imaginary part of the input is assumed to
<A ID="12899"></A>real Fast Fourier transform. The imaginary part of the input is assumed to
be zero. Only the first <IMG
WIDTH="60" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img1220.png"
@@ -150,7 +150,7 @@ are determined by symmetry). This takes half the computation time of the
WIDTH="106" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1221.png"
ALT="\fbox{\texttt{tabreceive\~}}">:
<A NAME="12900"></A>repeatedly outputs the contents of a wavetable. Each
<A ID="12900"></A>repeatedly outputs the contents of a wavetable. Each
block of computation outputs the same first <IMG
WIDTH="18" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img3.png"
@@ -169,7 +169,7 @@ graphing.
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.15"></A><A NAME="12707"></A>
<DIV ALIGN="CENTER"><A ID="fig09.15"></A><A ID="12707"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.15:</STRONG>
Fourier analysis and resynthesis, using <TT>block~</TT> to specify an
@@ -196,7 +196,7 @@ We need one new object for the inverse Fourier transform:
WIDTH="64" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1224.png"
ALT="\fbox{\texttt{rifft\~}}">:
<A NAME="12901"></A>real inverse Fast Fourier transform. Using the first <IMG
<A ID="12901"></A>real inverse Fast Fourier transform. Using the first <IMG
WIDTH="60" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img1220.png"
ALT="$N/2+1$"> points of its
@@ -248,36 +248,36 @@ Blocking delay does not depend on overlap, only on block sizes.
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Examples
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@@ -74,20 +74,20 @@ Examples
<P>
<BR><HR>
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<H2><A NAME="SECTION001372000000000000000">
<H2><A ID="SECTION001372000000000000000">
Narrow-band companding: noise suppression</A>
</H2>
<P>
<DIV ALIGN="CENTER"><A NAME="fig09.16"></A><A NAME="12719"></A>
<DIV ALIGN="CENTER"><A ID="fig09.16"></A><A ID="12719"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.16:</STRONG>
Noise suppression as an example of narrow-band companding: (a)
@@ -164,7 +164,7 @@ figure. To compute the mask we are using two new new objects:
WIDTH="56" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1231.png"
ALT="\fbox{\texttt{bang\~}}">:
<A NAME="12903"></A>send a bang in advance of each block of computation. The bang appears at the
<A ID="12903"></A>send a bang in advance of each block of computation. The bang appears at the
logical time of the first sample in each block (the earliest logical time whose
control computation affects that block and not the previous one), following
the scheme shown
@@ -178,7 +178,7 @@ in Figure <A HREF="node42.html#fig03.02">3.2</A>.
WIDTH="81" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1233.png"
ALT="\fbox{\texttt{tabsend\~}}">:
<A NAME="12904"></A>the companion object for <TT>tabreceive~</TT>, repeatedly copies its input to
<A ID="12904"></A>the companion object for <TT>tabreceive~</TT>, repeatedly copies its input to
the contents of a table, affecting up to the first <IMG
WIDTH="18" HEIGHT="14" ALIGN="BOTTOM" BORDER="0"
SRC="img3.png"
@@ -229,36 +229,36 @@ from any other one.)
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Timbre stamp ("vocoder")</A>
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<DIV ALIGN="CENTER"><A NAME="fig09.17"></A><A NAME="12742"></A>
<DIV ALIGN="CENTER"><A ID="fig09.17"></A><A ID="12742"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.17:</STRONG>
Timbre stamp.</CAPTION>
@@ -132,36 +132,36 @@ unchanged, and values in between give a smooth interpolation between the two.
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<H2><A NAME="SECTION001374000000000000000">
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Phase vocoder time bender</A>
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<DIV ALIGN="CENTER"><A NAME="fig09.18"></A><A NAME="12750"></A>
<DIV ALIGN="CENTER"><A ID="fig09.18"></A><A ID="12750"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 9.18:</STRONG>
Phase vocoder for time stretching and contraction.</CAPTION>
@@ -99,7 +99,7 @@ the location in the recorded sound we hear. Two new objects are used:
WIDTH="81" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1241.png"
ALT="\fbox{\texttt{lrshift\~}}">:
<A NAME="12905"></A>shift a block left or right (according to its creation argument). If the
<A ID="12905"></A>shift a block left or right (according to its creation argument). If the
argument is positive, each block of the output is the input shifted that number
of spaces to the right, filling zeros in as needed on the left. A negative
argument shifts to the left, filling zeros in at the right.
@@ -112,7 +112,7 @@ argument shifts to the left, filling zeros in at the right.
WIDTH="87" HEIGHT="39" ALIGN="MIDDLE" BORDER="0"
SRC="img1243.png"
ALT="\fbox{\texttt{q8\_rsqrt\~}}">:
<A NAME="12906"></A>quick and approximate reciprocal square root. Outputs the reciprocal of the
<A ID="12906"></A>quick and approximate reciprocal square root. Outputs the reciprocal of the
square root of its input, good to about a part in 256, using much less
computation than a full-precision square root and reciprocal would.
@@ -213,36 +213,36 @@ can be made using daisy-chained cross-fades.
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Exercises</A>
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@@ -154,36 +154,36 @@ is then channels 1 through 4)?
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<A NAME="chapter-waveforms"></A>
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<BR>
Classical waveforms
</H1>
@@ -82,12 +82,12 @@ controllable endpoints. This approach is historically at least as important
as the others, and was dominant during the
analog synthesizer period, approximately 1965-1985.
For lack of a better name, we'll use the term
<A NAME="14228"></A><I>classical waveforms</I>
<A ID="14228"></A><I>classical waveforms</I>
to denote waveforms composed of line segments.
<P>
<DIV ALIGN="CENTER"><A NAME="fig10.01"></A><A NAME="14232"></A>
<DIV ALIGN="CENTER"><A ID="fig10.01"></A><A ID="14232"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 10.1:</STRONG>
Classical waveforms: (a) the sawtooth, (b) the triangle, and (c)
@@ -102,7 +102,7 @@ shown as functions of a continuous variable (not sampled).</CAPTION>
<P>
They include the
<A NAME="14235"></A><A NAME="14236"></A><A NAME="14237"></A><I>sawtooth</I>, <I>triangle</I>, and <I>rectangle</I> waves pictured in Figure
<A ID="14235"></A><A ID="14236"></A><A ID="14237"></A><I>sawtooth</I>, <I>triangle</I>, and <I>rectangle</I> waves pictured in Figure
<A HREF="#fig10.01">10.1</A>, among many other possibilities. The salient features of
classical waveforms are either discontinuous jumps (changes in value) or
corners (changes in slope). In the figure, the sawtooth and rectangle waves
@@ -150,87 +150,87 @@ foldover is produced, and careful handling will help us control it.
<P>
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HREF="node190.html">Parabolic wave</A>
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@@ -471,45 +471,45 @@ of the two.
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<H2><A NAME="SECTION001411000000000000000">
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Sawtooth waves and symmetry</A>
</H2>
<P>
<DIV ALIGN="CENTER"><A NAME="fig10.02"></A><A NAME="14265"></A>
<DIV ALIGN="CENTER"><A ID="fig10.02"></A><A ID="14265"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 10.2:</STRONG>
Using a symmetry relation to extract even and odd harmonics from a
@@ -97,7 +97,7 @@ waveform with slope zero except at the discontinuities. The discontinuities
coming from the original sawtooth wave jump in the same direction (negative to
positive), but those coming from the shifted one are negated and jump from
positive to negative. The result is a
<A NAME="14269"></A><I>square wave</I>,
<A ID="14269"></A><I>square wave</I>,
a particular rectangle wave in which the two component segments
have the same duration.
@@ -112,36 +112,36 @@ waveform.
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Dissecting classical waveforms</A>
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@@ -79,7 +79,7 @@ waveform having only jumps (discontinuities in value) but no corners
amplitudes. We then develop the idea further, showing how to generate waveforms
with corners (either in addition to, or instead of, jumps) using
another elementary waveform we'll call the
<A NAME="14273"></A><I>parabolic wave</I>.
<A ID="14273"></A><I>parabolic wave</I>.
<P>
Suppose first that a waveform of period <IMG
@@ -121,7 +121,7 @@ a jump from a lower to a higher value.
<P>
<DIV ALIGN="CENTER"><A NAME="fig10.03"></A><A NAME="14282"></A>
<DIV ALIGN="CENTER"><A ID="fig10.03"></A><A ID="14282"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 10.3:</STRONG>
Dissecting a waveform: (a) the original waveform with two
@@ -242,7 +242,7 @@ this to happen in general the jumps must add to zero:
<P>
<DIV ALIGN="CENTER"><A NAME="fig10.04"></A><A NAME="14301"></A>
<DIV ALIGN="CENTER"><A ID="fig10.04"></A><A ID="14301"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 10.4:</STRONG>
The parabolic wave.</CAPTION>
@@ -328,7 +328,7 @@ An example is shown graphically in Figure <A HREF="#fig10.05">10.5</A>.
<P>
<DIV ALIGN="CENTER"><A NAME="fig10.05"></A><A NAME="14320"></A>
<DIV ALIGN="CENTER"><A ID="fig10.05"></A><A ID="14320"></A>
<TABLE>
<CAPTION ALIGN="BOTTOM"><STRONG>Figure 10.5:</STRONG>
Decomposing a triangle wave (part a) into two parabolic waves
@@ -367,36 +367,36 @@ connect with itself.
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<H1><A NAME="SECTION001430000000000000000"></A>
<A NAME="sect10.spectra"></A>
<H1><A ID="SECTION001430000000000000000"></A>
<A ID="sect10.spectra"></A>
<BR>
Fourier series of the elementary waveforms
</H1>
@@ -125,7 +125,7 @@ of a signal <IMG
WIDTH="31" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img80.png"
ALT="$x[n]$"> with its
<A NAME="14332"></A><I>first difference</I>,
<A ID="14332"></A><I>first difference</I>,
defined as <IMG
WIDTH="105" HEIGHT="32" ALIGN="MIDDLE" BORDER="0"
SRC="img1303.png"
@@ -248,51 +248,51 @@ Now we plug this result in to evaluate:
<P></P>
<BR><HR>
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HREF="node189.html">Sawtooth wave</A>
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<H2><A NAME="SECTION001431000000000000000">
<H2><A ID="SECTION001431000000000000000">
Sawtooth wave</A>
</H2>
First we apply this to the sawtooth wave <IMG
@@ -107,7 +107,7 @@ Ignoring the constant offset of <IMG
WIDTH="30" HEIGHT="35" ALIGN="MIDDLE" BORDER="0"
SRC="img1314.png"
ALT="$-{1 \over N}$">, this gives an
<A NAME="14349"></A><I>impulse</I>,
<A ID="14349"></A><I>impulse</I>,
zero everywhere except one sample per cycle. The summation in
the Fourier transform only has one term, and we get:
<BR><P></P>
@@ -305,36 +305,36 @@ s[n] \approx {1 \over \pi} \left [
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