forest-all-around/@sampler/Audio_SdFat.latest/effect_flange.cpp

221 lines
7 KiB
C++
Executable file

/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Pete (El Supremo)
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <Arduino.h>
#include "effect_flange.h"
#include "arm_math.h"
/******************************************************************/
// A u d i o E f f e c t F l a n g e
// Written by Pete (El Supremo) Jan 2014
// 140529 - change to handle mono stream and change modify() to voices()
// 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_delayline = NULL;
//short * AudioEffectFlange::r_delayline = NULL;
// User-supplied offset for the delayed sample
// but start with passthru
//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;
// 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;
if(0) {
Serial.print("AudioEffectFlange.begin(offset = ");
Serial.print(delay_offset);
Serial.print(", depth = ");
Serial.print(d_depth);
Serial.print(", rate = ");
Serial.print(delay_rate,3);
Serial.println(")");
Serial.print(" FLANGE_DELAY_LENGTH = ");
Serial.println(d_length);
}
delay_length = d_length/2;
l_delayline = delayline;
delay_depth = d_depth;
// initial index
l_delay_rate_index = 0;
l_circ_idx = 0;
delay_rate_incr =(delay_rate * 2147483648.0)/ AUDIO_SAMPLE_RATE_EXACT;
//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::voices(int delay_offset,int d_depth,float delay_rate)
{
boolean all_ok = true;
delay_depth = d_depth;
delay_rate_incr =(delay_rate * 2147483648.0)/ AUDIO_SAMPLE_RATE_EXACT;
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;
l_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;
// do passthru
if(delay_offset_idx == FLANGE_DELAY_PASSTHRU) {
// Just passthrough
block = receiveWritable(0);
if(block) {
bp = block->data;
// fill the delay line
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 the unmodified block
transmit(block,0);
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++) {
// 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;
// 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;
}
if(idx >= delay_length) {
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);
*bp++ = (l_delayline[l_circ_idx]+ l_delayline[idx] + frac)/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);
}
}