Change FIR filter (as suggested by rheslip) to filter one channel instead of two. Updated example, added documenting file

examples/fir_filter/fir_filter.ino

@@ -1,4 +1,8 @@
 /*
+c
+  - released
+b
+- Use FIR filters with fast_fft option
 
 The audio board uses the following pins.
  6 - MEMCS
@@ -25,14 +29,15 @@ The audio board uses the following pins.
 #include <Bounce.h>
 #include "filters.h"
 
-// If this pin is grounded the FIR filter is turned which
-// makes just pass through the audio
+// If this pin is grounded the FIR filter is turned off
+// which just passes the audio sraight through
 // Don't use any of the pins listed above
 #define PASSTHRU_PIN 1
 // If this pin goes low the next FIR filter in the list
 // is switched in.
 #define FILTER_PIN 0
 
+// debounce the passthru and filter switching pins
 Bounce b_passthru = Bounce(PASSTHRU_PIN,15);
 Bounce b_filter   = Bounce(FILTER_PIN,15);
 
@@ -41,16 +46,18 @@ const int myInput = AUDIO_INPUT_LINEIN;
 
 
 AudioInputI2S       audioInput;         // audio shield: mic or line-in
-AudioFilterFIR           myFilter;
+// Use the fast FIR filter for left and right channels
+AudioFilterFIR       myFilterL(true);
+AudioFilterFIR       myFilterR(true);
 AudioOutputI2S      audioOutput;        // audio shield: headphones & line-out
 
 // Create Audio connections between the components
-// Both channels of the audio input go to the FIR filter
-AudioConnection c1(audioInput, 0, myFilter, 0);
-AudioConnection c2(audioInput, 1, myFilter, 1);
-// both channels from the FIR filter go to the audio output
-AudioConnection c3(myFilter, 0, audioOutput, 0);
-AudioConnection c4(myFilter, 1, audioOutput, 1);
+// Route audio into the left and right filters
+AudioConnection c1(audioInput, 0, myFilterL, 0);
+AudioConnection c2(audioInput, 1, myFilterR, 0);
+// Route the output of the filters to their respective channels
+AudioConnection c3(myFilterL, 0, audioOutput, 0);
+AudioConnection c4(myFilterR, 0, audioOutput, 1);
 AudioControlSGTL5000 audioShield;
 
 
@@ -62,9 +69,9 @@ struct fir_filter {
 // index of current filter. Start with the low pass.
 int fir_idx = 0;
 struct fir_filter fir_list[] = {
-  low_pass , 100,    // low pass with cutoff at 1kHz and -60dB at 2kHz
-  band_pass, 100,    // bandpass 1200Hz - 1700Hz
-  NULL,      0
+  {low_pass , 100},    // low pass with cutoff at 1kHz and -60dB at 2kHz
+  {band_pass, 100},    // bandpass 1200Hz - 1700Hz
+  {NULL,      0}
 };
 
 
@@ -98,8 +105,8 @@ void setup() {
     Serial.println(") is grounded");
   }  
   // Initialize the filter
-  myFilter.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");
 }
 
@@ -108,7 +115,7 @@ int old_idx = -1;
 
 // audio volume
 int volume = 0;
-
+unsigned long last_time = millis();
 void loop()
 {
   // Volume control
@@ -122,18 +129,20 @@ void loop()
   b_passthru.update();
   b_filter.update();
   
-
-  
   // If the passthru button is pushed, save the current
   // filter index and then switch the filter to passthru
   if(b_passthru.fallingEdge()) {
     old_idx = fir_idx;
-    myFilter.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(old_idx != -1)myFilter.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+    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);
+    }
     old_idx = -1;
   }
   
@@ -141,9 +150,31 @@ void loop()
   if(b_filter.fallingEdge()) {
     fir_idx++;
     if(fir_list[fir_idx].num_coeffs == 0)fir_idx = 0;
-    myFilter.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);
+  }
+
+
+if(1) {
+// With fast_fir
+// Proc = 18 (18),  Mem = 4 (6)
+
+  if(millis() - last_time >= 5000) {
+    Serial.print("Proc = ");
+    Serial.print(AudioProcessorUsage());
+    Serial.print(" (");    
+    Serial.print(AudioProcessorUsageMax());
+    Serial.print("),  Mem = ");
+    Serial.print(AudioMemoryUsage());
+    Serial.print(" (");    
+    Serial.print(AudioMemoryUsageMax());
+    Serial.println(")");
+    last_time = millis();
   }
 }
 
 
+}
+
+
 

filter_fir.cpp

@@ -29,7 +29,6 @@ void AudioFilterFIR::begin(short *cp,int n_coeffs)
   // 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);
-    arm_fir_init_q15(&r_fir_inst, n_coeffs, coeff_p, &r_StateQ15[0], AUDIO_BLOCK_SAMPLES);
   }
 }
 
@@ -55,11 +54,6 @@ void AudioFilterFIR::update(void)
       transmit(block,0);
       release(block);
     }
-    block = receiveReadOnly(1);
-    if(block) {
-      transmit(block,1);
-      release(block);
-    }
     return;
   }
   // Left Channel
@@ -67,22 +61,15 @@ void AudioFilterFIR::update(void)
   // get a block for the FIR output
   b_new = allocate();
   if(block && b_new) {
-    arm_fir_q15(&l_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
+    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); 
     // send the FIR output to the left channel
     transmit(b_new,0);
   }
   if(block)release(block);
   if(b_new)release(b_new);
-
-  // Right Channel
-  block = receiveReadOnly(1);
-  b_new = allocate();
-  if(block && b_new) {
-    arm_fir_q15(&r_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
-    transmit(b_new,1);
-  }
-  if(block)release(block);
-  if(b_new)release(b_new);
 }
 
 

filter_fir.h

@@ -26,6 +26,10 @@
 #include "AudioStream.h"
 #include "arm_math.h"
 
+#define USE_FAST_FIR true
+#define USE_SLOW_FIR false
+
+
 // Maximum number of coefficients in a FIR filter
 // The audio breaks up with 128 coefficients so a
 // maximum of 150 is more than sufficient
@@ -39,12 +43,12 @@ class AudioFilterFIR :
 public AudioStream
 {
 public:
-  AudioFilterFIR(void): 
-  AudioStream(2,inputQueueArray), coeff_p(NULL)
+  AudioFilterFIR(const boolean a_f): 
+  AudioStream(2,inputQueueArray), arm_fast(a_f), coeff_p(NULL)
   { 
   }
 
-  void begin(short *coeff_p,int f_pin);
+  void begin(short *coeff_p,int n_coeffs);
   virtual void update(void);
   void stop(void);
   
@@ -54,11 +58,11 @@ private:
   // the state arrays are defined to handle a maximum of MAX_COEFFS
   // coefficients in a filter
   q15_t l_StateQ15[AUDIO_BLOCK_SAMPLES + MAX_COEFFS];
-  q15_t r_StateQ15[AUDIO_BLOCK_SAMPLES + MAX_COEFFS];
   arm_fir_instance_q15 l_fir_inst;
-  arm_fir_instance_q15 r_fir_inst;
   // pointer to current coefficients or NULL or FIR_PASSTHRU
   short *coeff_p;
+  // Whether to use the fast arm FIR code 
+  const boolean arm_fast;
 };
 
 #endif

filter_fir.md

@@ -0,0 +1,18 @@
+filter_fir 140418
+	Filters the audio stream using FIR coefficients supplied by the user.
+	The ARM	library has two q15 functions which perform an FIR filter. One uses
+	a 64-bit accumulator (arm_fir_q15) and the other uses a 32-bit accumulator
+	(arm_fir_fast_q15). When instantiating a filter a boolean argument specifies
+	which version to use. Specifying USE_FAST_FIR (defined as boolean true) uses
+	the fast code otherwise the slower code is used. For example:
+	AudioFilterFIR       myFilterL(USE_FAST_FIR);
+
+void begin(short *cp,int n_coeffs)
+	Starts the filter using coefficients at *cp and the number of coefficients
+	is n_coeffs. The special value FIR_PASSTHRU can be used in place of the
+	address of the coefficient array, in which case the function will just pass
+	audio samples through without filtering.
+
+void stop(void)
+	Stops the filter.
+