Impove and document FIR filter

examples/Effects/Filter_FIR/Filter_FIR.ino

 19 - SCL
 22 - TX
 23 - LRCLK
-
 */
 
 //#include <arm_math.h>
 
 
 AudioInputI2S       audioInput;         // audio shield: mic or line-in
-// Use the fast FIR filter for left and right channels
-AudioFilterFIR       myFilterL(true);
-AudioFilterFIR       myFilterR(true);
+AudioFilterFIR      myFilterL;
+AudioFilterFIR      myFilterR;
 AudioOutputI2S      audioOutput;        // audio shield: headphones & line-out
 
 // Create Audio connections between the components
 
 
 void setup() {
-  
   Serial.begin(9600);
-  while (!Serial) ;
-  delay(3000);
+  delay(300);
 
-  pinMode(PASSTHRU_PIN,INPUT_PULLUP);
-  pinMode(FILTER_PIN,INPUT_PULLUP);
+  pinMode(PASSTHRU_PIN, INPUT_PULLUP);
+  pinMode(FILTER_PIN, INPUT_PULLUP);
 
-  // It doesn't work properly with any less than 8
+  // allocate memory for the audio library
   AudioMemory(8);
 
   audioShield.enable();
     Serial.println(") is grounded");
   }  
   // Initialize the filter
-  myFilterL.begin(fir_list[0].coeffs,fir_list[0].num_coeffs);
-  myFilterR.begin(fir_list[0].coeffs,fir_list[0].num_coeffs);  
+  myFilterL.begin(fir_list[0].coeffs, fir_list[0].num_coeffs);
+  myFilterR.begin(fir_list[0].coeffs, fir_list[0].num_coeffs);
   Serial.println("setup done");
 }
 
 // audio volume
 int volume = 0;
 unsigned long last_time = millis();
+
 void loop()
 {
   // Volume control
   int n = analogRead(15);
   if (n != volume) {
     volume = n;
-    audioShield.volume((float)n / 1023);
+    //uncomment this line if your audio shield has the volume pot
+    //audioShield.volume((float)n / 1023);
   }
   
   // update the two buttons
   
   // If the passthru button is pushed, save the current
   // filter index and then switch the filter to passthru
-  if(b_passthru.fallingEdge()) {
+  if (b_passthru.fallingEdge()) {
     old_idx = fir_idx;
-    myFilterL.begin(FIR_PASSTHRU,0);
-    myFilterR.begin(FIR_PASSTHRU,0);
+    myFilterL.begin(FIR_PASSTHRU, 0);
+    myFilterR.begin(FIR_PASSTHRU, 0);
   }
   
   // If passthru button is released, restore previous filter
-  if(b_passthru.risingEdge()) {
+  if (b_passthru.risingEdge()) {
     if(old_idx != -1) {
-      myFilterL.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
-      myFilterR.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+      myFilterL.begin(fir_list[fir_idx].coeffs, fir_list[fir_idx].num_coeffs);
+      myFilterR.begin(fir_list[fir_idx].coeffs, fir_list[fir_idx].num_coeffs);
     }
     old_idx = -1;
   }
   
   // switch to next filter in the list
-  if(b_filter.fallingEdge()) {
+  if (b_filter.fallingEdge()) {
     fir_idx++;
-    if(fir_list[fir_idx].num_coeffs == 0)fir_idx = 0;
-    myFilterL.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
-    myFilterR.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+    if (fir_list[fir_idx].num_coeffs == 0) fir_idx = 0;
+    myFilterL.begin(fir_list[fir_idx].coeffs, fir_list[fir_idx].num_coeffs);
+    myFilterR.begin(fir_list[fir_idx].coeffs, fir_list[fir_idx].num_coeffs);
   }
 
-
-if(1) {
-// With fast_fir
-// Proc = 18 (18),  Mem = 4 (6)
-
-  if(millis() - last_time >= 5000) {
+  // print information about resource usage
+  // Proc = 18 (18),  Mem = 4 (5)
+  if (millis() - last_time >= 2500) {
     Serial.print("Proc = ");
     Serial.print(AudioProcessorUsage());
     Serial.print(" (");    
     Serial.println(")");
     last_time = millis();
   }
-}
 
 
 }

examples/Effects/Filter_FIR/filters.cpp

 #include "filters.h"
 short low_pass[NUM_COEFFS] = {
-#include "lopass_4000_44100.h"
+#include "lopass_1000_44100.h"
 };
 
 short band_pass[NUM_COEFFS] = {

filter_fir.cpp

 
 #include "filter_fir.h"
 
-void AudioFilterFIR::begin(short *cp, int n_coeffs)
-{
-	// pointer to coefficients
-	coeff_p = cp;
-	// Initialize FIR instances for the left and right channels
-	if(coeff_p && (coeff_p != FIR_PASSTHRU)) {
-		arm_fir_init_q15(&l_fir_inst, n_coeffs, coeff_p,
-			&l_StateQ15[0], AUDIO_BLOCK_SAMPLES);
-	}
-}
 
-// This has the same effect as begin(NULL,0);
-void AudioFilterFIR::stop(void)
+void AudioFilterFIR::update(void)
 {
-	coeff_p = NULL;
-}
+	audio_block_t *block, *b_new;
 
+	block = receiveReadOnly();
+	if (!block) return;
 
-void AudioFilterFIR::update(void)
-{
-	audio_block_t *block,*b_new;
-  
 	// If there's no coefficient table, give up.  
-	if(coeff_p == NULL) return;
+	if (coeff_p == NULL) {
+		release(block);
+		return;
+	}
 
 	// do passthru
 	if (coeff_p == FIR_PASSTHRU) {
 		// Just passthrough
-		block = receiveReadOnly(0);
-		if(block) {
-			transmit(block,0);
-			release(block);
-		}
+		transmit(block);
+		release(block);
 		return;
 	}
-	// Left Channel
-	block = receiveReadOnly(0);
+
 	// get a block for the FIR output
 	b_new = allocate();
-	if (block && b_new) {
-		if (arm_fast) {
-			arm_fir_fast_q15(&l_fir_inst, (q15_t *)block->data,
-				(q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
-		} else {
-			arm_fir_q15(&l_fir_inst, (q15_t *)block->data,
-				(q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
-		}
-		transmit(b_new,0); // send the FIR output
+	if (b_new) {
+		arm_fir_fast_q15(&fir_inst, (q15_t *)block->data,
+			(q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
+		transmit(b_new); // send the FIR output
+		release(b_new);
 	}
-	if (block) release(block);
-	if (b_new) release(b_new);
+	release(block);
 }
 
 

filter_fir.h

 
 // Indicates that the code should just pass through the audio
 // without any filtering (as opposed to doing nothing at all)
-#define FIR_PASSTHRU ((short *) 1)
+#define FIR_PASSTHRU ((const short *) 1)
 
-#define FIR_MAX_COEFFS 120
+#define FIR_MAX_COEFFS 200
 
 class AudioFilterFIR : public AudioStream
 {
 public:
-	AudioFilterFIR(const boolean a_f): AudioStream(1,inputQueueArray),
-	  arm_fast(a_f), coeff_p(NULL) { 
+	AudioFilterFIR(void): AudioStream(1,inputQueueArray), coeff_p(NULL) {
+	}
+	void begin(const short *cp, int n_coeffs) {
+		coeff_p = cp;
+		// Initialize FIR instance (ARM DSP Math Library)
+		if (coeff_p && (coeff_p != FIR_PASSTHRU) && n_coeffs <= FIR_MAX_COEFFS) {
+			arm_fir_init_q15(&fir_inst, n_coeffs, (q15_t *)coeff_p,
+				&StateQ15[0], AUDIO_BLOCK_SAMPLES);
+		}
+	}
+	void end(void) {
+		coeff_p = NULL;
 	}
-	void begin(short *coeff_p,int n_coeffs);
 	virtual void update(void);
-	void stop(void);
-  
 private:
 	audio_block_t *inputQueueArray[1];
-	// arm state arrays and FIR instances for left and right channels
-	// the state arrays are defined to handle a maximum of MAX_COEFFS
-	// coefficients in a filter
-	q15_t l_StateQ15[AUDIO_BLOCK_SAMPLES + FIR_MAX_COEFFS];
 
-	// Whether to use the fast arm FIR code 
-	const boolean arm_fast;
 	// pointer to current coefficients or NULL or FIR_PASSTHRU
-	short *coeff_p;
-	arm_fir_instance_q15 l_fir_inst;
+	const short *coeff_p;
+
+	// ARM DSP Math library filter instance
+	arm_fir_instance_q15 fir_inst;
+	q15_t StateQ15[AUDIO_BLOCK_SAMPLES + FIR_MAX_COEFFS];
 };
 
 #endif

gui/list.html

 		<tr class=top><th>Port</th><th>Purpose</th></tr>
 		<tr class=odd><td align=center></td><td></td></tr>
 	</table>
-	<!--<h3>Parameters</h3>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Name</th><th>Type</th><th>Function</th></tr>
-		<tr class=odd><td align=center></td><td>Integer</td><td></td></tr>
-	</table>
-	<p>Extra description...  Section only present if object has params</p>-->
 	<h3>Functions</h3>
 	<p class=func><span class=keyword>begin</span>(delayBuffer, length, n_chorus);</p>
 	<p class=desc>blah blah blah blah
 		<tr class=top><th>Port</th><th>Purpose</th></tr>
 		<tr class=odd><td align=center></td><td></td></tr>
 	</table>
-	<!--<h3>Parameters</h3>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Name</th><th>Type</th><th>Function</th></tr>
-		<tr class=odd><td align=center></td><td>Integer</td><td></td></tr>
-	</table>
-	<p>Extra description...  Section only present if object has params</p>-->
 	<h3>Functions</h3>
 	<p class=func><span class=keyword>begin</span>(delayBuffer, length, offset, depth, delayRate);</p>
 	<p class=desc>blah blah blah blah
 </script>
 <script type="text/x-red" data-help-name="AudioFilterFIR">
 	<h3>Summary</h3>
-	<p>description</p>
+	<div>
+	<p>Finite impulse response filter, useful for all sorts of filtering.
+	</p>
+	<p align=center><img src="fir_filter.png"></p>
+	</div>
 	<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></td><td></td></tr>
-	</table>
-	<!--<h3>Parameters</h3>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Name</th><th>Type</th><th>Function</th></tr>
-		<tr class=odd><td align=center></td><td>Integer</td><td></td></tr>
+		<tr class=odd><td align=center>In 0</td><td>Signal to be filtered</td></tr>
+		<tr class=odd><td align=center>Out 0</td><td>Filtered Signal Output</td></tr>
 	</table>
-	<p>Extra description...  Section only present if object has params</p>-->
 	<h3>Functions</h3>
-	<p class=func><span class=keyword>function</span>(parm1, parm2);</p>
-	<p class=desc>blah blah blah blah
+	<p class=func><span class=keyword>begin</span>(array, length);</p>
+	<p class=desc>Initialize the filter.  The array must be 16 bit integers (the
+		filter's impulse response), and
+		length indicates the number of points in the array.  Array may also be
+		FIR_PASSTHRU (length = 0), to directly pass the input to output without
+		filtering.
+	</p>
+	<p class=func><span class=keyword>end</span>();</p>
+	<p class=desc>Turn the filter off.
 	</p>
 	<h3>Notes</h3>
-	<p></p>
+	<p>FIR filters requires more CPU time than Biquad (IIR), but they can
+		implement filters with better phase response.
+	</p>
+	<p>A 100 point filter requires 9% CPU time on Teensy 3.1.  The maximum
+		supported filter length is 200 points.
+	</p>
+	<p>The free
+		<a href="http://t-filter.appspot.com/fir/index.html" target="_blank"> TFilter Design Tool</a>
+		can be used to create the impulse response array.  Be sure to set the sampling
+		frequency to 44117 HZ (it defaults to only 2000 Hz) and the output type to "int" (16 bit).
+	</p>
+	<p>
+		If you use TFilter Design's "C/C++ array" option, it's output has "int" definition, which
+		is 32 bits on Teensy 3.1.  Edit "int" to "short" for an array of 16 bit numbers,
+		and add "const" to avoid consuming extra RAM.
+	</p>
 </script>
 <script type="text/x-red" data-template-name="AudioFilterFIR">
 	<div class="form-row">
 </script>
 <script type="text/x-red" data-help-name="AudioAnalyzePeak">
 	<h3>Summary</h3>
-	<p>description</p>
+	<p>Track the signal peak amplitude.  Very useful for simple
+		audio level response projects, and general troubleshooting.</p>
 	<h3>Audio Connections</h3>
 	<table class=doc align=center cellpadding=3>
 		<tr class=top><th>Port</th><th>Purpose</th></tr>
 <script type="text/x-red" data-help-name="AudioAnalyzePrint">
 	<h3>Summary</h3>
 	<p>Print raw audio data to the Arduino Serial Monitor.  This
-		object creates massive output quickly, and should not normall be used.</p>
+		object creates massive output quickly, and should not normally be used.</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></td><td></td></tr>
+		<tr class=odd><td align=center>In 0</td><td>Signal to print</td></tr>
 	</table>
-	<!--<h3>Parameters</h3>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Name</th><th>Type</th><th>Function</th></tr>
-		<tr class=odd><td align=center></td><td>Integer</td><td></td></tr>
-	</table>
-	<p>Extra description...  Section only present if object has params</p>-->
 	<h3>Functions</h3>
 	<p class=func><span class=keyword>name</span>(string);</p>
 	<p class=desc>blah blah blah blah
 	<p class=desc>blah blah blah blah
 	</p>
 	<h3>Notes</h3>
-	<p></p>
+	<p>This object doesn't work very well and probably should not be used.</p>
 </script>
 <script type="text/x-red" data-template-name="AudioAnalyzePrint">
 	<div class="form-row">