Manually merged Pete's pull request: Correct a calculation in flange;remove static class from chorus and flange

https://github.com/PaulStoffregen/Audio/pull/7

effect_chorus.cpp

@@ -26,19 +26,8 @@
 
 //                A u d i o E f f e c t C h o r u s
 // Written by Pete (El Supremo) Jan 2014
+// 140219 - correct storage class (not static)
 
-// circular addressing indices for left and right channels
-short AudioEffectChorus::l_circ_idx;
-short AudioEffectChorus::r_circ_idx;
-
-short * AudioEffectChorus::l_delayline = NULL;
-short * AudioEffectChorus::r_delayline = NULL;
-int AudioEffectChorus::delay_length;
-// An initial value of zero indicates passthru
-int AudioEffectChorus::num_chorus = 0;
-
-
-// All three must be valid.
 boolean AudioEffectChorus::begin(short *delayline,int d_length,int n_chorus)
 {
 Serial.print("AudioEffectChorus.begin(Chorus delay line length = ");
@@ -47,11 +36,11 @@ Serial.print(", n_chorus = ");
 Serial.print(n_chorus);
 Serial.println(")");
 
-l_delayline = NULL;
-r_delayline = NULL;
-delay_length = 0;
-l_circ_idx = 0;
-r_circ_idx = 0;
+  l_delayline = NULL;
+  r_delayline = NULL;
+  delay_length = 0;
+  l_circ_idx = 0;
+  r_circ_idx = 0;
 
   if(delayline == NULL) {
     return(false);
@@ -71,25 +60,12 @@ r_circ_idx = 0;
   return(true);
 }
 
-// This has the same effect as begin(NULL,0);
-void AudioEffectChorus::stop(void)
-{
-
-}
-
 void AudioEffectChorus::modify(int n_chorus)
 {
   num_chorus = n_chorus;
 }
 
-int iabs(int x)
-{
-  if(x < 0)return(-x);
-  return(x);
-}
-//static int d_count = 0;
-
-int last_idx = 0;
+//int last_idx = 0;
 void AudioEffectChorus::update(void)
 {
   audio_block_t *block;

effect_chorus.h

@@ -29,14 +29,17 @@
 
 //                A u d i o E f f e c t C h o r u s
 // Written by Pete (El Supremo) Jan 2014
+// 140219 - correct storage class (not static)
+
+#define CHORUS_DELAY_PASSTHRU -1
 
 class AudioEffectChorus : 
 public AudioStream
 {
 public:
-  AudioEffectChorus(void): 
-  AudioStream(2,inputQueueArray) { 
-  }
+  AudioEffectChorus(void):
+  AudioStream(2,inputQueueArray), num_chorus(2)
+  { }
 
   boolean begin(short *delayline,int delay_length,int n_chorus);
   virtual void update(void);
@@ -45,12 +48,12 @@ public:
   
 private:
   audio_block_t *inputQueueArray[2];
-  static short *l_delayline;
-  static short *r_delayline;
-  static short l_circ_idx;
-  static short r_circ_idx;
-  static int num_chorus;
-  static int delay_length;
+  short *l_delayline;
+  short *r_delayline;
+  short l_circ_idx;
+  short r_circ_idx;
+  int num_chorus;
+  int delay_length;
 };
 
 #endif

effect_flange.cpp

@@ -29,23 +29,24 @@
 // 140207 - fix calculation of delay_rate_incr which is expressed as
 //			a fraction of 2*PI
 // 140207 - cosmetic fix to begin()
+// 140219 - correct the calculation of "frac"
 
 // circular addressing indices for left and right channels
-short AudioEffectFlange::l_circ_idx;
-short AudioEffectFlange::r_circ_idx;
+//short AudioEffectFlange::l_circ_idx;
+//short AudioEffectFlange::r_circ_idx;
 
-short * AudioEffectFlange::l_delayline = NULL;
-short * AudioEffectFlange::r_delayline = NULL;
+//short * AudioEffectFlange::l_delayline = NULL;
+//short * AudioEffectFlange::r_delayline = NULL;
 
 // User-supplied offset for the delayed sample
 // but start with passthru
-int AudioEffectFlange::delay_offset_idx = DELAY_PASSTHRU;
-int AudioEffectFlange::delay_length;
+//int AudioEffectFlange::delay_offset_idx = FLANGE_DELAY_PASSTHRU;
+//int AudioEffectFlange::delay_length;
 
-int AudioEffectFlange::delay_depth;
-int AudioEffectFlange::delay_rate_incr;
-unsigned int AudioEffectFlange::l_delay_rate_index;
-unsigned int AudioEffectFlange::r_delay_rate_index;
+//int AudioEffectFlange::delay_depth;
+//int AudioEffectFlange::delay_rate_incr;
+//unsigned int AudioEffectFlange::l_delay_rate_index;
+//unsigned int AudioEffectFlange::r_delay_rate_index;
 // fails if the user provides unreasonable values but will
 // coerce them and go ahead anyway. e.g. if the delay offset
 // is >= CHORUS_DELAY_LENGTH, the code will force it to
@@ -132,7 +133,7 @@ void AudioEffectFlange::update(void)
   if(r_delayline == NULL)return; 
 
   // do passthru
-  if(delay_offset_idx == DELAY_PASSTHRU) {
+  if(delay_offset_idx == FLANGE_DELAY_PASSTHRU) {
     // Just passthrough
     block = receiveWritable(0);
     if(block) {
@@ -169,15 +170,30 @@ void AudioEffectFlange::update(void)
   if(block) {
     bp = block->data;
     for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+      // increment the index into the circular delay line buffer
       l_circ_idx++;
+      // wrap the index around if necessary
       if(l_circ_idx >= delay_length) {
         l_circ_idx = 0;
       }
+      // store the current sample in the delay line
       l_delayline[l_circ_idx] = *bp;
-      idx = arm_sin_q15( (q15_t)((l_delay_rate_index >> 16) & 0x7fff));
-      idx = (idx * delay_depth) >> 15;
+      // The argument to the arm_sin_q15 function is NOT in radians. It is
+      // actually, in effect, the fraction remaining after the division
+      // of radians/(2*PI) which is then expressed as a positive Q15
+      // fraction in the interval [0 , +1) - this is l_delay_rate_index. 
+      // l_delay_rate_index should probably be called l_delay_rate_phase
+      // (sorry about that!)
+      // It is a Q31 positive number of which the high order 16 bits are
+      // used when calculating the sine. idx will have a value in the
+      // interval [-1 , +1)
+      frac = arm_sin_q15( (q15_t)((l_delay_rate_index >> 16) & 0x7fff));
+      // multiply the sin by the delay depth
+      idx = (frac * delay_depth) >> 15;
 //Serial.println(idx);
+      // Calculate the offset into the buffer
       idx = l_circ_idx - (delay_offset_idx + idx);
+      // and adjust idx to point into the circular buffer
       if(idx < 0) {
         idx += delay_length;
       }
@@ -185,21 +201,28 @@ void AudioEffectFlange::update(void)
         idx -= delay_length;
       }
 
+      // Here we interpolate between two indices but if the sine was negative
+      // then we interpolate between idx and idx-1, otherwise the
+      // interpolation is between idx and idx+1
       if(frac < 0)
         idx1 = idx - 1;
       else
         idx1 = idx + 1;
+      // adjust idx1 in the circular buffer
       if(idx1 < 0) {
         idx1 += delay_length;
       }
       if(idx1 >= delay_length) {
         idx1 -= delay_length;
       }
+      // Do the interpolation
       frac = (l_delay_rate_index >> 1) &0x7fff;
       frac = (( (int)(l_delayline[idx1] - l_delayline[idx])*frac) >> 15);
 
+//frac = 0;
       *bp++ = (l_delayline[l_circ_idx]
-                + l_delayline[idx] + frac               
+                + l_delayline[idx] + frac
+//                + l_delayline[(l_circ_idx + delay_length/2) % delay_length]
               )/2;
 
       l_delay_rate_index += delay_rate_incr;
@@ -223,8 +246,8 @@ void AudioEffectFlange::update(void)
         r_circ_idx = 0;
       }
       r_delayline[r_circ_idx] = *bp;
-      idx = arm_sin_q15( (q15_t)((r_delay_rate_index >> 16)&0x7fff));
-       idx = (idx * delay_depth) >> 15;
+      frac = arm_sin_q15( (q15_t)((r_delay_rate_index >> 16)&0x7fff));
+      idx = (frac * delay_depth) >> 15;
 
       idx = r_circ_idx - (delay_offset_idx + idx);
       if(idx < 0) {
@@ -247,6 +270,7 @@ void AudioEffectFlange::update(void)
       frac = (r_delay_rate_index >> 1) &0x7fff;
       frac = (( (int)(r_delayline[idx1] - r_delayline[idx])*frac) >> 15);
 
+//frac = 0;
       *bp++ = (r_delayline[r_circ_idx]
                 + r_delayline[idx] + frac
                )/2;

effect_flange.h

@@ -29,7 +29,7 @@
 //                A u d i o E f f e c t F l a n g e
 // Written by Pete (El Supremo) Jan 2014
 
-#define DELAY_PASSTHRU -1
+#define FLANGE_DELAY_PASSTHRU 0
 
 class AudioEffectFlange : 
 public AudioStream
@@ -46,16 +46,16 @@ public:
   
 private:
   audio_block_t *inputQueueArray[2];
-  static short *l_delayline;
-  static short *r_delayline;
-  static int delay_length;
-  static short l_circ_idx;
-  static short r_circ_idx;
-  static int delay_depth;
-  static int delay_offset_idx;
-  static int   delay_rate_incr;
-  static unsigned int l_delay_rate_index;
-  static unsigned int r_delay_rate_index;
+  short *l_delayline;
+  short *r_delayline;
+  int delay_length;
+  short l_circ_idx;
+  short r_circ_idx;
+  int delay_depth;
+  int delay_offset_idx;
+  int   delay_rate_incr;
+  unsigned int l_delay_rate_index;
+  unsigned int r_delay_rate_index;
 };
 
 #endif

examples/effect_chorus/effect_chorus.ino

@@ -174,7 +174,7 @@ AudioControlSGTL5000 audioShield;
 
 
 // number of "voices" in the chorus which INCLUDES the original voice
-int n_chorus = 3;
+int n_chorus = 2;
 
 // <<<<<<<<<<<<<<>>>>>>>>>>>>>>>>
 void setup() {

examples/effect_flange/effect_flange.ino

@@ -150,7 +150,7 @@ many blocks you provided with AudioMemory().
 #include <Bounce.h>
 
 // Number of samples in ONE channel
-#define FLANGE_DELAY_LENGTH (16*AUDIO_BLOCK_SAMPLES)
+#define FLANGE_DELAY_LENGTH (6*AUDIO_BLOCK_SAMPLES)
 // Allocate the delay line for left and right channels
 // The delayline will hold left and right samples so it
 // should be declared to be twice as long as the desired
@@ -221,9 +221,14 @@ double s_freq = .0625;
 */
 
 //12 - good with Eric Clapton Unplugged
+/*
 int s_idx = 3*FLANGE_DELAY_LENGTH/4;
 int s_depth = FLANGE_DELAY_LENGTH/8;
 double s_freq = .0625;
+*/
+int s_idx = FLANGE_DELAY_LENGTH/4;
+int s_depth = FLANGE_DELAY_LENGTH/4;
+double s_freq = .5;
 
 void setup() {