Add chorus and flange effects and an example sketch for each

Audio.cpp

@@ -3148,3 +3148,415 @@ void AudioFilterFIR::update(void)
   if(b_new)release(b_new);
 }
 
+
+/******************************************************************/
+//                A u d i o E f f e c t F l a n g e
+// Written by Pete (El Supremo) Jan 2014
+
+// circular addressing indices for left and right channels
+short AudioEffectFlange::l_circ_idx;
+short AudioEffectFlange::r_circ_idx;
+
+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_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
+// CHORUS_DELAY_LENGTH-1 and return false.
+// delay_rate is the rate (in Hz) of the sine wave modulation
+// delay_depth is the maximum variation around delay_offset
+// i.e. the total offset is delay_offset + delay_depth * sin(delay_rate)
+boolean AudioEffectFlange::begin(short *delayline,int d_length,int delay_offset,int d_depth,float delay_rate)
+{
+  boolean all_ok = true;
+
+Serial.print("AudioEffectFlange.begin(ofsset = ");
+Serial.print(delay_offset);
+Serial.print(", depth = ");
+Serial.print(d_depth);
+Serial.print(", rate = ");
+Serial.print(delay_rate,3);
+Serial.println(")");
+Serial.print("    CHORUS_DELAY_LENGTH = ");
+Serial.println(d_length);
+
+  delay_length = d_length/2;
+  l_delayline = delayline;
+  r_delayline = delayline + delay_length;
+  
+  delay_depth = d_depth;
+  // initial index
+  l_delay_rate_index = 0;
+  r_delay_rate_index = 0;
+  l_circ_idx = 0;
+  r_circ_idx = 0;
+  delay_rate_incr = 2*PI*delay_rate/44100.*2147483648.; 
+//Serial.println(delay_rate_incr,HEX);
+
+  delay_offset_idx = delay_offset;
+  // Allow the passthru code to go through
+  if(delay_offset_idx < -1) {
+    delay_offset_idx = 0;
+    all_ok = false;
+  }
+  if(delay_offset_idx >= delay_length) {
+    delay_offset_idx = delay_length - 1;
+    all_ok = false;
+  }  
+  return(all_ok);
+}
+
+
+boolean AudioEffectFlange::modify(int delay_offset,int d_depth,float delay_rate)
+{
+  boolean all_ok = true;
+  
+  delay_depth = d_depth;
+
+  delay_rate_incr = 2*PI*delay_rate/44100.*2147483648.;
+  
+  delay_offset_idx = delay_offset;
+  // Allow the passthru code to go through
+  if(delay_offset_idx < -1) {
+    delay_offset_idx = 0;
+    all_ok = false;
+  }
+  if(delay_offset_idx >= delay_length) {
+    delay_offset_idx = delay_length - 1;
+    all_ok = false;
+  }
+  l_delay_rate_index = 0;
+  r_delay_rate_index = 0;
+  l_circ_idx = 0;
+  r_circ_idx = 0;
+  return(all_ok);
+}
+
+void AudioEffectFlange::update(void)
+{
+  audio_block_t *block;
+  int idx;
+  short *bp;
+  short frac;
+  int idx1;
+
+  if(l_delayline == NULL)return;
+  if(r_delayline == NULL)return; 
+
+  // do passthru
+  if(delay_offset_idx == DELAY_PASSTHRU) {
+    // Just passthrough
+    block = receiveWritable(0);
+    if(block) {
+      bp = block->data;
+      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+        l_circ_idx++;
+        if(l_circ_idx >= delay_length) {
+          l_circ_idx = 0;
+        }
+        l_delayline[l_circ_idx] = *bp++;
+      }
+      transmit(block,0);
+      release(block);
+    }
+    block = receiveWritable(1);
+    if(block) {
+      bp = block->data;
+      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+        r_circ_idx++;
+        if(r_circ_idx >= delay_length) {
+          r_circ_idx = 0;
+        }
+        r_delayline[r_circ_idx] = *bp++;
+      }
+      transmit(block,1);
+      release(block);
+    }
+    return;
+  }
+
+  //          L E F T  C H A N N E L
+
+  block = receiveWritable(0);
+  if(block) {
+    bp = block->data;
+    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+      l_circ_idx++;
+      if(l_circ_idx >= delay_length) {
+        l_circ_idx = 0;
+      }
+      l_delayline[l_circ_idx] = *bp;
+      idx = arm_sin_q15( (q15_t)((l_delay_rate_index >> 16) & 0x7fff));
+      idx = (idx * delay_depth) >> 15;
+//Serial.println(idx);
+      idx = l_circ_idx - (delay_offset_idx + idx);
+      if(idx < 0) {
+        idx += delay_length;
+      }
+      if(idx >= delay_length) {
+        idx -= delay_length;
+      }
+
+      if(frac < 0)
+        idx1 = idx - 1;
+      else
+        idx1 = idx + 1;
+      if(idx1 < 0) {
+        idx1 += delay_length;
+      }
+      if(idx1 >= delay_length) {
+        idx1 -= delay_length;
+      }
+      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[(l_circ_idx + delay_length/2) % delay_length]                
+              )/2;
+
+      l_delay_rate_index += delay_rate_incr;
+      if(l_delay_rate_index & 0x80000000) {
+        l_delay_rate_index &= 0x7fffffff;
+      }
+    }
+    // send the effect output to the left channel
+    transmit(block,0);
+    release(block);
+  }
+
+  //          R I G H T  C H A N N E L
+
+  block = receiveWritable(1);
+  if(block) {
+    bp = block->data;
+    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+      r_circ_idx++;
+      if(r_circ_idx >= delay_length) {
+        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;
+
+      idx = r_circ_idx - (delay_offset_idx + idx);
+      if(idx < 0) {
+        idx += delay_length;
+      }
+      if(idx >= delay_length) {
+        idx -= delay_length;
+      }
+
+      if(frac < 0)
+        idx1 = idx - 1;
+      else
+        idx1 = idx + 1;
+      if(idx1 < 0) {
+        idx1 += delay_length;
+      }
+      if(idx1 >= delay_length) {
+        idx1 -= delay_length;
+      }
+      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;
+
+      r_delay_rate_index += delay_rate_incr;
+      if(r_delay_rate_index & 0x80000000) {
+        r_delay_rate_index &= 0x7fffffff;
+      }
+
+    }
+    // send the effect output to the right channel
+    transmit(block,1);
+    release(block);
+  }
+}
+
+
+
+/******************************************************************/
+
+//                A u d i o E f f e c t C h o r u s
+// Written by Pete (El Supremo) Jan 2014
+
+// 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 = ");
+Serial.print(d_length);
+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;
+
+  if(delayline == NULL) {
+    return(false);
+  }
+  if(d_length < 10) {
+    return(false);
+  }
+  if(n_chorus < 1) {
+    return(false);
+  }
+  
+  l_delayline = delayline;
+  r_delayline = delayline + d_length/2;
+  delay_length = d_length/2;
+  num_chorus = n_chorus;
+ 
+  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;
+void AudioEffectChorus::update(void)
+{
+  audio_block_t *block;
+  short *bp;
+  int sum;
+  int c_idx;
+
+  if(l_delayline == NULL)return;
+  if(r_delayline == NULL)return;  
+  
+  // do passthru
+  // It stores the unmodified data in the delay line so that
+  // it isn't as likely to click
+  if(num_chorus < 1) {
+    // Just passthrough
+    block = receiveWritable(0);
+    if(block) {
+      bp = block->data;
+      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+        l_circ_idx++;
+        if(l_circ_idx >= delay_length) {
+          l_circ_idx = 0;
+        }
+        l_delayline[l_circ_idx] = *bp++;
+      }
+      transmit(block,0);
+      release(block);
+    }
+    block = receiveWritable(1);
+    if(block) {
+      bp = block->data;
+      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+        r_circ_idx++;
+        if(r_circ_idx >= delay_length) {
+          r_circ_idx = 0;
+        }
+        r_delayline[r_circ_idx] = *bp++;
+      }
+      transmit(block,1);
+      release(block);
+    }
+    return;
+  }
+
+  //          L E F T  C H A N N E L
+
+  block = receiveWritable(0);
+  if(block) {
+    bp = block->data;
+    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+      l_circ_idx++;
+      if(l_circ_idx >= delay_length) {
+        l_circ_idx = 0;
+      }
+      l_delayline[l_circ_idx] = *bp;
+      sum = 0;
+      c_idx = l_circ_idx;
+      for(int k = 0; k < num_chorus; k++) {
+        sum += l_delayline[c_idx];
+        if(num_chorus > 1)c_idx -= delay_length/(num_chorus - 1) - 1;
+        if(c_idx < 0) {
+          c_idx += delay_length;
+        }
+      }
+      *bp++ = sum/num_chorus;
+    }
+
+    // send the effect output to the left channel
+    transmit(block,0);
+    release(block);
+  }
+
+  //          R I G H T  C H A N N E L
+
+  block = receiveWritable(1);
+  if(block) {
+    bp = block->data;
+    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+      r_circ_idx++;
+      if(r_circ_idx >= delay_length) {
+        r_circ_idx = 0;
+      }
+      r_delayline[r_circ_idx] = *bp;
+      sum = 0;
+      c_idx = r_circ_idx;
+      for(int k = 0; k < num_chorus; k++) {
+        sum += r_delayline[c_idx];
+        if(num_chorus > 1)c_idx -= delay_length/(num_chorus - 1) - 1;
+        if(c_idx < 0) {
+          c_idx += delay_length;
+        }
+      }
+      *bp++ = sum/num_chorus;
+    }
+
+    // send the effect output to the left channel
+    transmit(block,1);
+    release(block);
+  }
+}
+

Audio.h

@@ -576,6 +576,69 @@ private:
 
 
 
+/******************************************************************/
+//                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 0
+
+class AudioEffectFlange : 
+public AudioStream
+{
+public:
+  AudioEffectFlange(void): 
+  AudioStream(2,inputQueueArray) { 
+  }
+
+  boolean begin(short *delayline,int d_length,int delay_offset,int d_depth,float delay_rate);
+  boolean modify(int delay_offset,int d_depth,float delay_rate);
+  virtual void update(void);
+  void stop(void);
+  
+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;
+};
+
+
+/******************************************************************/
+
+//                A u d i o E f f e c t C h o r u s
+// Written by Pete (El Supremo) Jan 2014
+
+#define DELAY_PASSTHRU -1
+
+class AudioEffectChorus : 
+public AudioStream
+{
+public:
+  AudioEffectChorus(void): 
+  AudioStream(2,inputQueueArray) { 
+  }
+
+  boolean begin(short *delayline,int delay_length,int n_chorus);
+  virtual void update(void);
+  void stop(void);
+  void modify(int n_chorus);
+  
+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;
+};
 
 
 

examples/effect_chorus/effect_chorus.ino

@@ -0,0 +1,264 @@
+/*
+PROC/MEM 9/4
+Modify filter_test_f to try out a chorus effect. 
+TODO:
+
+
+140203
+  o
+I have mixed up the names "chorus" and flange". The sketches named
+chorus have, up to version 'm', actually implemented a flanger.
+From version 'o' onwards the sketches named chorus will implement
+a real chorus effect and flange will do a flanging effect.
+
+  
+  n only changed the effect parameters
+  m
+
+140201
+  l - found the problem at last using my_flange_cd_usd_c
+YES! Way back when I found I hadn't been using d_depth. Now I
+    have just discovered that I haven't been using delay_offset!!!!!
+ 
+140201
+  k
+  interpolation doesn't remove the ticking
+  
+140131
+  j
+>>> The lower the frequency, the less ticking.
+  - try interpolation
+140201
+  - got this restored to the way it was last night
+    and then reinstated the changes. I had a couple of
+    changes to the right channel that were incorrect or
+    weren't carried over from the left channel changes
+  
+  i
+  - don't know why but this version seems to have more "presence"
+    than previous versions.
+    The presence occurred when "sign = 1" was put in front of the left.
+    It essentially makes it a passthrough.
+    If both have "sign=1" then it is identical to passthrough
+    
+    Ticking is still not fixed
+    
+  h
+  - add sign reversal. It seems to make audio much clearer
+    BUT it hasn't got rid of the ticking noise
+  
+  g
+  - I wasn't even using delay_depth!!!!
+  
+140131
+  f
+  - added code to print the processor and memory usage every 5 seconds
+NOW the problem is to try to remove the ticking
+
+  e
+FOUND the problem with the right channel. It was also in the left channel
+but the placement of the delay line arrays made it more noticeable in the
+right channel. I was not calculating idx properly. In particular, the
+resuling index could be negative.
+
+I have shortened the delay line to only 2*AUDIO_BLOCK_SAMPLES
+
+  - removed redundancies in the update code. rewrite the block
+    instead of getting a new one
+Haven't solved the noise in the right channel yet.
+Tried duplicating right channel code to left but noise stays on the right
+
+140130
+  d
+The noise stays in the right channel even if it is calculated first
+Switching the L/R inputs doesn't switch the noise to the left channel  
+  
+  c
+>> Now add a sinusoidal modulation to the offset
+    There's an awful noise in both channels but it is much louder in
+    the right channel. NOPE - it is ONLY in the right channel
+    but the audio does sound like it is working (sort of) except that it
+    is rather tinny. Maybe it needs to have the interpolation added.
+
+  b
+  - this works with clip16_6s.wav.
+    The original of this audio file was from http://www.donreiman.com/Chorus/Chorus.htm
+    I reworked it with Goldwave to make it a stereo WAV file
+    But with Rick Wakewan's Jane Seymour it seems to act more like
+    a high-pass filter.
+    
+  a
+  - removed FIR stuff and changed the name to AudioEffectChorus
+    it's basically a blank template and compiles.
+
+From: http://www.cs.cf.ac.uk/Dave/CM0268/PDF/10_CM0268_Audio_FX.pdf
+about Comb filter effects
+Effect      Delay range (ms)    Modulation
+Resonator      0 - 20            None
+Flanger        0 - 15            Sinusoidal (approx 1Hz)
+Chorus        25 - 50            None
+Echo            >50              None
+
+FMI:
+The audio board uses the following pins.
+ 6 - MEMCS
+ 7 - MOSI
+ 9 - BCLK
+10 - SDCS
+11 - MCLK
+12 - MISO
+13 - RX
+14 - SCLK
+15 - VOL
+18 - SDA
+19 - SCL
+22 - TX
+23 - LRCLK
+
+
+AudioProcessorUsage()
+AudioProcessorUsageMax()
+AudioProcessorUsageMaxReset()
+AudioMemoryUsage()
+AudioMemoryUsageMax()
+AudioMemoryUsageMaxReset()
+
+The CPU usage is an integer from 0 to 100, and the memory is from 0 to however
+many blocks you provided with AudioMemory().
+
+*/
+
+#include <arm_math.h>
+#include <Audio.h>
+#include <Wire.h>
+//#include <WM8731.h>
+#include <SD.h>
+#include <SPI.h>
+#include <Bounce.h>
+
+
+
+// Number of samples in ONE channel
+#define CHORUS_DELAY_LENGTH (16*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
+// number of samples in one channel
+#define CHORUS_DELAYLINE (CHORUS_DELAY_LENGTH*2)
+// The delay line for left and right channels
+short delayline[CHORUS_DELAYLINE];
+
+// If this pin is grounded the FIR filter is turned which
+// makes just pass through the audio
+// Don't use any of the pins listed above
+#define PASSTHRU_PIN 1
+
+Bounce b_passthru = Bounce(PASSTHRU_PIN,15);
+
+//const int myInput = AUDIO_INPUT_MIC;
+const int myInput = AUDIO_INPUT_LINEIN;
+
+AudioInputI2S       audioInput;         // audio shield: mic or line-in
+AudioEffectChorus   myEffect;
+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, myEffect, 0);
+AudioConnection c2(audioInput, 1, myEffect, 1);
+// both channels from the FIR filter go to the audio output
+AudioConnection c3(myEffect, 0, audioOutput, 0);
+AudioConnection c4(myEffect, 1, audioOutput, 1);
+
+AudioControlSGTL5000 audioShield;
+
+
+// number of "voices" in the chorus which INCLUDES the original voice
+int n_chorus = 3;
+
+// <<<<<<<<<<<<<<>>>>>>>>>>>>>>>>
+void setup() {
+  
+  Serial.begin(9600);
+  while (!Serial) ;
+  delay(3000);
+
+  pinMode(PASSTHRU_PIN,INPUT_PULLUP);
+
+  // Maximum memory usage was reported as 4
+  // Proc = 9 (9),  Mem = 4 (4)
+  AudioMemory(4);
+
+  audioShield.enable();
+  audioShield.inputSelect(myInput);
+  audioShield.volume(50);
+  
+  // Warn that the passthru pin is grounded
+  if(!digitalRead(PASSTHRU_PIN)) {
+    Serial.print("PASSTHRU_PIN (");
+    Serial.print(PASSTHRU_PIN);
+    Serial.println(") is grounded");
+  }
+
+  // Initialize the effect
+  // address of delayline
+  // total number of samples (left AND right) in the delay line
+  // number of voices in the chorus INCLUDING the original voice
+  if(!myEffect.begin(delayline,CHORUS_DELAYLINE,n_chorus)) {
+    Serial.println("AudioEffectChorus - begin failed");
+    while(1);
+  }
+  
+  // I want output on the line out too
+  audioShield.unmuteLineout();
+//  audioShield.muteHeadphone();
+  
+  Serial.println("setup done");
+AudioProcessorUsageMaxReset();
+AudioMemoryUsageMaxReset();
+}
+
+
+// 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 / 10.23);
+  }
+if(1) {
+  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();
+  }
+}
+  // update the button
+  b_passthru.update();
+ 
+  // If the passthru button is pushed, switch the effect to passthru
+  if(b_passthru.fallingEdge()) {
+    myEffect.modify(0);
+  }
+  
+  // If passthru button is released, restore the previous chorus
+  if(b_passthru.risingEdge()) {
+    myEffect.modify(n_chorus);
+  }
+
+}
+
+
+

examples/effect_chorus/effects_info.h

@@ -0,0 +1,43 @@
+/*
+CHORUS and FLANGE effects
+  Both effects use a delay line to hold previous samples. This allows
+  the current sample to be combined in some way with a sample that
+  occurred in the past. An obvious effect this would allow would be
+  an echo where the current sample is combined with a sample from,
+  say, 250 milliseconds ago. The chorus and flange effects do this
+  as well but they combine samples from only about 50ms or less ago.
+  
+CHORUS EFFECT
+  This combines one or more samples up to about 50ms ago. In this
+  library, the additional samples are evenly spread through the
+  supplied delay line.
+  E.G. If the number of voices is specified as 2 then the effect
+  combines the current sample and the oldest sample (the last one in
+  the delay line). If the number of voices is 3 then the effect
+  combines the most recent sample, the oldest sample and the sample
+  in the middle of the delay line.
+  For two voices the effect can be represented as:
+  result = sample(0) + sample(dt)
+  where sample(0) represents the current sample and sample(dt) is
+  the sample in the delay line from dt milliseconds ago.
+
+FLANGE EFFECT
+  This combines only one sample from the delay line but the position
+  of that sample varies sinusoidally.
+  In this case the effect can be represented as:
+  result = sample(0) + sample(dt + depth*sin(2*PI*Fe))
+  The value of the sine function is always a number from -1 to +1
+  and so the result of depth*(sinFe) is always a number from
+  -depth to +depth. Thus, the delayed sample will be selected from
+  the range (dt-depth) to (dt+depth). This selection will vary
+  at whatever rate is specified as the frequency of the effect Fe.
+  I have found that rates of .25 seconds or less are best, otherwise
+  the effect is very "watery" and in extreme cases the sound is
+  even off-key!
+
+  When trying out these effects with recorded music as input, it is
+  best to use those where there is a solo voice which is clearly
+  "in front" of the accompaninemnt. Tracks which already contain
+  flange or chorus effects don't work well.
+  
+*/

examples/effect_flange/effect_flange.ino

@@ -0,0 +1,314 @@
+/*
+Change the chorus code to produce a flange effect
+
+PROC/MEM 25/4
+140204
+  d
+  - fixed the problem with user-supplied delay line
+
+140203
+  c
+  
+140203
+  b
+  - when switching to/from passthru, keep the delay line filled
+    BUT to be effective must also fix up begin and add another
+    function to allow changing to/from passthru without 
+    reinitialing everything.
+    
+I have mixed up the names "chorus" and flange". The sketches named
+chorus have, up to version 'm', actually implemented a flanger.
+From version 'o' onwards the sketches named chorus will implement
+a real chorus effect and flange will do a flanging effect.
+
+140202
+  a
+Modify filter_test_f to try out a chorus effect
+
+m + n only changed the effect parameters
+    - 
+140201
+  l - found the problem at last using my_flange_cd_usd_c
+YES! Way back when I found I hadn't been using d_depth. Now I
+    have just discovered that I haven't been using delay_offset!!!!!
+ 
+140201
+  k
+  interpolation doesn't remove the ticking
+  
+140131
+  j
+>>> The lower the frequency, the less ticking.
+  - try interpolation
+140201
+  - got this restored to the way it was last night
+    and then reinstated the changes. I had a couple of
+    changes to the right channel that were incorrect or
+    weren't carried over from the left channel changes
+  
+  i
+  - don't know why but this version seems to have more "presence"
+    than previous versions.
+    The presence occurred when "sign = 1" was put in front of the left.
+    It essentially makes it a passthrough.
+    If both have "sign=1" then it is identical to passthrough
+    
+    Ticking is still not fixed
+    
+  h
+  - add sign reversal. It seems to make audio much clearer
+    BUT it hasn't got rid of the ticking noise
+  
+  g
+  - I wasn't even using delay_depth!!!!
+  
+140131
+  f
+  - added code to print the processor and memory usage every 5 seconds
+NOW the problem is to try to remove the ticking
+
+  e
+FOUND the problem with the right channel. It was also in the left channel
+but the placement of the delay line arrays made it more noticeable in the
+right channel. I was not calculating idx properly. In particular, the
+resuling index could be negative.
+
+I have shortened the delay line to only 2*AUDIO_BLOCK_SAMPLES
+
+  - removed redundancies in the update code. rewrite the block
+    instead of getting a new one
+Haven't solved the noise in the right channel yet.
+Tried duplicating right channel code to left but noise stays on the right
+
+140130
+  d
+The noise stays in the right channel even if it is calculated first
+Switching the L/R inputs doesn't switch the noise to the left channel  
+  
+  c
+>> Now add a sinusoidal modulation to the offset
+    There's an awful noise in both channels but it is much louder in
+    the right channel. NOPE - it is ONLY in the right channel
+    but the audio does sound like it is working (sort of) except that it
+    is rather tinny. Maybe it needs to have the interpolation added.
+
+  b
+  - this works with clip16_6s.wav.
+    The original of this audio file was from http://www.donreiman.com/Chorus/Chorus.htm
+    I reworked it with Goldwave to make it a stereo WAV file
+    But with Rick Wakewan's Jane Seymour it seems to act more like
+    a high-pass filter.
+    
+  a
+  - removed FIR stuff and changed the name to AudioEffectChorus
+    it's basically a blank template and compiles.
+
+From: http://www.cs.cf.ac.uk/Dave/CM0268/PDF/10_CM0268_Audio_FX.pdf
+about Comb filter effects
+Effect      Delay range (ms)    Modulation
+Resonator      0 - 20            None
+Flanger        0 - 15            Sinusoidal (approx 1Hz)
+Chorus        25 - 50            None
+Echo            >50              None
+
+FMI:
+The audio board uses the following pins.
+ 6 - MEMCS
+ 7 - MOSI
+ 9 - BCLK
+10 - SDCS
+11 - MCLK
+12 - MISO
+13 - RX
+14 - SCLK
+15 - VOL
+18 - SDA
+19 - SCL
+22 - TX
+23 - LRCLK
+
+
+AudioProcessorUsage()
+AudioProcessorUsageMax()
+AudioProcessorUsageMaxReset()
+AudioMemoryUsage()
+AudioMemoryUsageMax()
+AudioMemoryUsageMaxReset()
+
+The CPU usage is an integer from 0 to 100, and the memory is from 0 to however
+many blocks you provided with AudioMemory().
+
+
+*/
+
+#include <arm_math.h>
+#include <Audio.h>
+#include <Wire.h>
+//#include <WM8731.h>
+#include <SD.h>
+#include <SPI.h>
+#include <Bounce.h>
+
+// Number of samples in ONE channel
+#define FLANGE_DELAY_LENGTH (16*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
+// number of samples in one channel
+#define FLANGE_DELAYLINE (FLANGE_DELAY_LENGTH*2)
+// The delay line for left and right channels
+short delayline[FLANGE_DELAYLINE];
+
+// If this pin is grounded the FIR filter is turned which
+// makes just pass through the audio
+// Don't use any of the pins listed above
+#define PASSTHRU_PIN 1
+
+Bounce b_passthru = Bounce(PASSTHRU_PIN,15);
+
+//const int myInput = AUDIO_INPUT_MIC;
+const int myInput = AUDIO_INPUT_LINEIN;
+
+AudioInputI2S       audioInput;         // audio shield: mic or line-in
+AudioEffectFlange   myEffect;
+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, myEffect, 0);
+AudioConnection c2(audioInput, 1, myEffect, 1);
+// both channels from the FIR filter go to the audio output
+AudioConnection c3(myEffect, 0, audioOutput, 0);
+AudioConnection c4(myEffect, 1, audioOutput, 1);
+
+AudioControlSGTL5000 audioShield;
+
+/* 
+int s_idx = FLANGE_DELAY_LENGTH/2;
+int s_depth = FLANGE_DELAY_LENGTH/16;
+double s_freq = 1;
+
+// <<<<<<<<<<<<<<>>>>>>>>>>>>>>>>
+// 12
+int s_idx = FLANGE_DELAY_LENGTH/2;
+int s_depth = FLANGE_DELAY_LENGTH/8;
+double s_freq = .125;
+// with .125 the ticking is about 1Hz with music
+// but with the noise sample it is a bit slower than that
+// <<<<<<<<<<<<<<>>>>>>>>>>>>>>>>
+*/
+/*
+// <<<<<<<<<<<<<<>>>>>>>>>>>>>>>>
+// 12
+int s_idx = FLANGE_DELAY_LENGTH/2;
+int s_depth = FLANGE_DELAY_LENGTH/12;
+double s_freq = .125;
+// with .125 the ticking is about 1Hz with music
+// but with the noise sample it is a bit slower than that
+// <<<<<<<<<<<<<<>>>>>>>>>>>>>>>>
+*/
+/*
+//12
+int s_idx = 15*FLANGE_DELAY_LENGTH/16;
+int s_depth = 15*FLANGE_DELAY_LENGTH/16;
+double s_freq = 0;
+*/
+/*
+//12
+int s_idx = 2*FLANGE_DELAY_LENGTH/4;
+int s_depth = FLANGE_DELAY_LENGTH/8;
+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;
+
+void setup() {
+  
+  Serial.begin(9600);
+  while (!Serial) ;
+  delay(3000);
+
+  pinMode(PASSTHRU_PIN,INPUT_PULLUP);
+
+  // It doesn't work properly with any less than 8
+  // but that was an earlier version. Processor and
+  // memory usage are now (ver j)
+  // Proc = 24 (24),  Mem = 4 (4)
+  AudioMemory(8);
+
+  audioShield.enable();
+  audioShield.inputSelect(myInput);
+  audioShield.volume(50);
+  
+  // Warn that the passthru pin is grounded
+  if(!digitalRead(PASSTHRU_PIN)) {
+    Serial.print("PASSTHRU_PIN (");
+    Serial.print(PASSTHRU_PIN);
+    Serial.println(") is grounded");
+  }
+
+  // Set up the flange effect
+  // address of delayline
+  // total number of samples (left AND right) in the delay line
+  // Index (in samples) into the delay line for the added voice
+  // Depth of the flange effect
+  // frequency of the flange effect
+  myEffect.begin(delayline,FLANGE_DELAYLINE,s_idx,s_depth,s_freq);
+
+  // I want output on the line out too
+  audioShield.unmuteLineout();
+  
+  Serial.println("setup done");
+AudioProcessorUsageMaxReset();
+AudioMemoryUsageMaxReset();
+}
+
+
+// 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 / 10.23);
+  }
+if(1) {
+  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();
+  }
+}
+  // update the button
+  b_passthru.update();
+ 
+  // If the passthru button is pushed, save the current
+  // filter index and then switch the filter to passthru
+  if(b_passthru.fallingEdge()) {
+    myEffect.modify(DELAY_PASSTHRU,0,0);
+  }
+  
+  // If passthru button is released, restore the effect
+  if(b_passthru.risingEdge()) {
+    myEffect.modify(s_idx,s_depth,s_freq);
+  }
+
+}
+
+
+
+

examples/effect_flange/effects_info.h

@@ -0,0 +1,44 @@
+/*
+CHORUS and FLANGE effects
+  Both effects use a delay line to hold previous samples. This allows
+  the current sample to be combined in some way with a sample that
+  occurred in the past. An obvious effect this would allow would be
+  an echo where the current sample is combined with a sample from,
+  say, 250 milliseconds ago. The chorus and flange effects do this
+  as well but they combine samples from only about 50ms (or less) ago.
+  
+CHORUS EFFECT
+  This combines one or more samples up to about 50ms ago. In this
+  library, the additional samples are evenly spread through the
+  supplied delay line.
+  E.G. If the number of voices is specified as 2 then the effect
+  combines the current sample and the oldest sample (the last one in
+  the delay line). If the number of voices is 3 then the effect
+  combines the most recent sample, the oldest sample and the sample
+  in the middle of the delay line.
+  For two voices the effect can be represented as:
+  result = sample(0) + sample(dt)
+  where sample(0) represents the current sample and sample(dt) is
+  the sample in the delay line from dt milliseconds ago.
+
+
+FLANGE EFFECT
+  This combines only one sample from the delay line but the position
+  of that sample varies sinusoidally.
+  In this case the effect can be represented as:
+  result = sample(0) + sample(dt + depth*sin(2*PI*Fe))
+  The value of the sine function is always a number from -1 to +1
+  and so the result of depth*(sinFe) is always a number from
+  -depth to +depth. Thus, the delayed sample will be selected from
+  the range (dt-depth) to (dt+depth). This selection will vary
+  at whatever rate is specified as the frequency of the effect Fe.
+  I have found that rates of .25 seconds or less are best, otherwise
+  the effect is very "watery" and in extreme cases the sound is
+  even off-key!
+
+  When trying out these effects with recorded music as input, it is
+  best to use those where there is a solo voice which is clearly
+  "in front" of the accompaninemnt. Tracks which already contain
+  flange or chorus effects don't work well.
+  
+*/