Finish AudioFilterStateVariable (signal control of corner frequency)

filter_variable.cpp

@@ -27,14 +27,23 @@
 #include "filter_variable.h"
 #include "utility/dspinst.h"
 
-
 // State Variable Filter (Chamberlin) with 2X oversampling
 // http://www.musicdsp.org/showArchiveComment.php?ArchiveID=92
 
+// The fast 32x32 with rshift32 discards 2 bits, which probably
+// never matter.
 //#define MULT(a, b) (int32_t)(((int64_t)(a) * (b)) >> 30)
 #define MULT(a, b) (multiply_32x32_rshift32_rounded(a, b) << 2)
 
-static bool first = true;
+// It's very unlikely anyone could hear any difference, but if you
+// care deeply about numerical precision in seldom-used cases,
+// uncomment this to improve the control signal accuracy
+//#define IMPROVE_HIGH_FREQUENCY_ACCURACY
+
+// This increases the exponential approximation accuracy from
+// about 0.341% error to only 0.012% error, which probably makes
+// no audible difference.
+//#define IMPROVE_EXPONENTIAL_ACCURACY
 
 void AudioFilterStateVariable::update_fixed(const int16_t *in,
 	int16_t *lp, int16_t *bp, int16_t *hp)
@@ -50,7 +59,6 @@ void AudioFilterStateVariable::update_fixed(const int16_t *in,
 	inputprev = state_inputprev;
 	lowpass = state_lowpass;
 	bandpass = state_bandpass;
-
 	do {
 		input = (*in++) << 12;
 		lowpass = lowpass + MULT(fmult, bandpass);
@@ -69,33 +77,88 @@ void AudioFilterStateVariable::update_fixed(const int16_t *in,
 		*lp++ = lowpasstmp;
 		*bp++ = bandpasstmp;
 		*hp++ = highpasstmp;
-#if 0
-		if (first) {
-			Serial.print(input >> 12);
-			Serial.print(", ");
-			Serial.print(lowpasstmp);
-			Serial.print(", ");
-			Serial.print(bandpasstmp);
-			Serial.print(", ");
-			Serial.print(highpasstmp);
-			Serial.println();
-		}
-#endif
 	} while (in < end);
-
 	state_inputprev = inputprev;
 	state_lowpass = lowpass;
 	state_bandpass = bandpass;
-	first = false;
 }
 
+
 void AudioFilterStateVariable::update_variable(const int16_t *in,
 	const int16_t *ctl, int16_t *lp, int16_t *bp, int16_t *hp)
 {
-	// TODO: implement control signal modulation of corner frequency
-	update_fixed(in, lp, bp, hp);
+	const int16_t *end = in + AUDIO_BLOCK_SAMPLES;
+	int32_t input, inputprev, control;
+	int32_t lowpass, bandpass, highpass;
+	int32_t lowpasstmp, bandpasstmp, highpasstmp;
+	int32_t fcenter, fmult, damp, octavemult;
+	int32_t n;
+
+	fcenter = setting_fcenter;
+	octavemult = setting_octavemult;
+	damp = setting_damp;
+	inputprev = state_inputprev;
+	lowpass = state_lowpass;
+	bandpass = state_bandpass;
+	do {
+		// compute fmult using control input, fcenter and octavemult
+		control = *ctl++;          // signal is always 15 fractional bits
+		control *= octavemult;     // octavemult range: 0 to 28671 (12 frac bits)
+		n = control & 0x7FFFFFF;   // 27 fractional control bits
+		#ifdef IMPROVE_EXPONENTIAL_ACCURACY
+		int32_t x = n << 3;
+		n = multiply_accumulate_32x32_rshift32_rounded(536870912, x, 1494202713);
+		int32_t sq = multiply_32x32_rshift32_rounded(x, x);
+		n = multiply_accumulate_32x32_rshift32_rounded(n, sq, 1934101615);
+		n = n + (multiply_32x32_rshift32_rounded(sq,
+			multiply_32x32_rshift32_rounded(x, 1358044250)) << 1);
+		n = n << 1;
+		#else
+		n = (n + 134217728) << 3;
+		n = multiply_32x32_rshift32_rounded(n, n);
+		n = multiply_32x32_rshift32_rounded(n, 715827883) << 3;
+		n = n + 715827882;
+		#endif
+		n = n >> (6 - (control >> 27)); // 4 integer control bits
+		fmult = multiply_32x32_rshift32_rounded(fcenter, n);
+		if (fmult > 5378279) fmult = 5378279;
+		fmult = fmult << 8;
+		// fmult is within 0.4% accuracy for all but the top 2 octaves
+		// of the audio band.  This math improves accuracy above 5 kHz.
+		// Without this, the filter still works fine for processing
+		// high frequencies, but the filter's corner frequency response
+		// can end up about 6% higher than requested.
+		#ifdef IMPROVE_HIGH_FREQUENCY_ACCURACY
+		fmult = (multiply_32x32_rshift32_rounded(fmult, 2145892402) +
+			multiply_32x32_rshift32_rounded(
+			multiply_32x32_rshift32_rounded(fmult, fmult),
+			multiply_32x32_rshift32_rounded(fmult, -1383276101))) << 1;
+		#endif
+		// now do the state variable filter as normal, using fmult
+		input = (*in++) << 12;
+		lowpass = lowpass + MULT(fmult, bandpass);
+		highpass = ((input + inputprev)>>1) - lowpass - MULT(damp, bandpass);
+		inputprev = input;
+		bandpass = bandpass + MULT(fmult, highpass);
+		lowpasstmp = lowpass;
+		bandpasstmp = bandpass;
+		highpasstmp = highpass;
+		lowpass = lowpass + MULT(fmult, bandpass);
+		highpass = input - lowpass - MULT(damp, bandpass);
+		bandpass = bandpass + MULT(fmult, highpass);
+		lowpasstmp = signed_saturate_rshift(lowpass+lowpasstmp, 16, 13);
+		bandpasstmp = signed_saturate_rshift(bandpass+bandpasstmp, 16, 13);
+		highpasstmp = signed_saturate_rshift(highpass+highpasstmp, 16, 13);
+		*lp++ = lowpasstmp;
+		*bp++ = bandpasstmp;
+		*hp++ = highpasstmp;
+	} while (in < end);
+	state_inputprev = inputprev;
+	state_lowpass = lowpass;
+	state_bandpass = bandpass;
 }
 
+
 void AudioFilterStateVariable::update(void)
 {
 	audio_block_t *input_block=NULL, *control_block=NULL;

filter_variable.h

@@ -34,6 +34,7 @@ class AudioFilterStateVariable: public AudioStream
 public:
 	AudioFilterStateVariable() : AudioStream(2, inputQueueArray) {
 		frequency(1000);
+		octaveControl(1.0); // default values
 		resonance(0.707);
 		state_inputprev = 0;
 		state_lowpass = 0;
@@ -42,6 +43,10 @@ public:
 	void frequency(float freq) {
 		if (freq < 20.0) freq = 20.0;
 		else if (freq > AUDIO_SAMPLE_RATE_EXACT/2.5) freq = AUDIO_SAMPLE_RATE_EXACT/2.5;
+		setting_fcenter = (freq * (3.141592654/(AUDIO_SAMPLE_RATE_EXACT*2.0)))
+			* 2147483647.0;
+		// TODO: should we use an approximation when freq is not a const,
+		// so the sinf() function isn't linked?
 		setting_fmult = sinf(freq * (3.141592654/(AUDIO_SAMPLE_RATE_EXACT*2.0)))
 			* 2147483647.0;
 	}
@@ -51,13 +56,23 @@ public:
 		// TODO: allow lower Q when frequency is lower
 		setting_damp = (1.0 / q) * 1073741824.0;
 	}
+	void octaveControl(float n) {
+		// filter's corner frequency is Fcenter * 2^(control * N)
+		// where "control" ranges from -1.0 to +1.0
+		// and "N" allows the frequency to change from 0 to 7 octaves
+		if (n < 0.0) n = 0.0;
+		else if (n > 6.9999) n = 6.9999;
+		setting_octavemult = n * 4096.0;
+	}
 	virtual void update(void);
 private:
 	void update_fixed(const int16_t *in,
 		int16_t *lp, int16_t *bp, int16_t *hp);
 	void update_variable(const int16_t *in, const int16_t *ctl,
 		int16_t *lp, int16_t *bp, int16_t *hp);
+	int32_t setting_fcenter;
 	int32_t setting_fmult;
+	int32_t setting_octavemult;
 	int32_t setting_damp;
 	int32_t state_inputprev;
 	int32_t state_lowpass;

gui/list.html

@@ -1341,6 +1341,73 @@
 
 
 
+<script type="text/javascript">
+	RED.nodes.registerType('AudioFilterStateVariable',{
+	shortName: "filter",
+		inputs:2,
+		outputs:3,
+		category: 'filter-function',
+		color:"#E6E0F8",
+		icon: "arrow-in.png"
+	});
+</script>
+<script type="text/x-red" data-help-name="AudioFilterStateVariable">
+	<h3>Summary</h3>
+	<p>A State Variable (Chamberlin) Filter with 12 dB/octave roll-off,
+		adjustable resonance, and optional signal control of corner
+		frequency.</p>
+	<h3>Audio Connections</h3>
+	<table class=doc align=center cellpadding=3>
+		<tr class=top><th>Port</th><th>Purpose</th></tr>
+		<tr class=odd><td align=center>In 0</td><td>Signal to Filter</td></tr>
+		<tr class=odd><td align=center>In 1</td><td>Frequency Control</td></tr>
+		<tr class=odd><td align=center>Out 0</td><td>Low Pass Output</td></tr>
+		<tr class=odd><td align=center>Out 1</td><td>Band Pass Output</td></tr>
+		<tr class=odd><td align=center>Out 2</td><td>High Pass Output</td></tr>
+	</table>
+	<h3>Functions</h3>
+	<p class=func><span class=keyword>frequency</span>(freq);</p>
+	<p class=desc>Set the filter's corner frequency.  When a signal is
+		connected to the control input, the filter will implement this
+		frequency when the signal is zero.
+	</p>
+	<p class=func><span class=keyword>resonance</span>(Q);</p>
+	<p class=desc>Set the filter's resonance.  Q ranges from 0.7 to 5.0.
+		Resonance greater than 0.707 will amplify the signal near the
+		corner frequency.  You must attenuate the signal before input
+		to this filter, to prevent clipping.
+	</p>
+	<p class=func><span class=keyword>octaveControl</span>(octaves);</p>
+	<p class=desc>Set how much (in octaves) the control signal can alter
+		the filter's corner freqency.  Range is 0 to 7 octaves.  For
+		example, when set to 2.5, a full scale positive signal (1.0) will
+		shift the filter frequency up 2.5 octaves, and a full scale negative
+		signal will shift it down 2.5 octaves.
+	</p>
+	<h3>Notes</h3>
+	<p>
+		When controlled by a signal, the equation for the filter
+		frequency is:
+	</p>
+	<p>
+		F = Fcenter * 2^<sup>(signal * octaves)</sup>
+		<br><small>If anyone knows how to do HTML equations, please
+		help me improve this.....</small>
+	</p>
+	<p>When operating with signal control of corner frequency, this
+		object uses approximately 4% of the CPU time on Teensy 3.1.
+	</p>
+</script>
+<script type="text/x-red" data-template-name="AudioFilterFIR">
+	<div class="form-row">
+		<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
+		<input type="text" id="node-input-name" placeholder="Name">
+	</div>
+</script>
+
+
+
+
 <script type="text/javascript">
 	RED.nodes.registerType('AudioPeak',{
 	shortName: "peak",

keywords.txt

@@ -44,6 +44,7 @@ updateCoefs	KEYWORD2
 frequency	KEYWORD2
 amplitude	KEYWORD2
 resonance	KEYWORD2
+octaveControl	KEYWORD2
 modify	KEYWORD2
 output	KEYWORD2
 trigger	KEYWORD2