forest-all-around/@sampler/lib/Audio_SdFat/effect_delay.cpp

139 lines
3.9 KiB
C++

/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* 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, development funding 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_delay.h"
void AudioEffectDelay::update(void)
{
audio_block_t *output;
uint32_t head, tail, count, channel, index, prev, offset;
const int16_t *src, *end;
int16_t *dst;
// grab incoming data and put it into the queue
head = headindex;
tail = tailindex;
if (++head >= DELAY_QUEUE_SIZE) head = 0;
if (head == tail) {
if (queue[tail] != NULL) release(queue[tail]);
if (++tail >= DELAY_QUEUE_SIZE) tail = 0;
}
queue[head] = receiveReadOnly();
headindex = head;
// testing only.... don't allow null pointers into the queue
// instead, fill the empty times with blocks of zeros
//if (queue[head] == NULL) {
// queue[head] = allocate();
// if (queue[head]) {
// dst = queue[head]->data;
// end = dst + AUDIO_BLOCK_SAMPLES;
// do {
// *dst++ = 0;
// } while (dst < end);
// } else {
// digitalWriteFast(2, HIGH);
// delayMicroseconds(5);
// digitalWriteFast(2, LOW);
// }
//}
// discard unneeded blocks from the queue
if (head >= tail) {
count = head - tail;
} else {
count = DELAY_QUEUE_SIZE + head - tail;
}
if (count > maxblocks) {
count -= maxblocks;
do {
if (queue[tail] != NULL) {
release(queue[tail]);
queue[tail] = NULL;
}
if (++tail >= DELAY_QUEUE_SIZE) tail = 0;
} while (--count > 0);
}
tailindex = tail;
// transmit the delayed outputs using queue data
for (channel = 0; channel < 8; channel++) {
if (!(activemask & (1<<channel))) continue;
index = position[channel] / AUDIO_BLOCK_SAMPLES;
offset = position[channel] % AUDIO_BLOCK_SAMPLES;
if (head >= index) {
index = head - index;
} else {
index = DELAY_QUEUE_SIZE + head - index;
}
if (offset == 0) {
// delay falls on the block boundary
if (queue[index]) {
transmit(queue[index], channel);
}
} else {
// delay requires grabbing data from 2 blocks
output = allocate();
if (!output) continue;
dst = output->data;
if (index > 0) {
prev = index - 1;
} else {
prev = DELAY_QUEUE_SIZE-1;
}
if (queue[prev]) {
end = queue[prev]->data + AUDIO_BLOCK_SAMPLES;
src = end - offset;
while (src < end) {
*dst++ = *src++; // TODO: optimize
}
} else {
end = dst + offset;
while (dst < end) {
*dst++ = 0;
}
}
end = output->data + AUDIO_BLOCK_SAMPLES;
if (queue[index]) {
src = queue[index]->data;
while (dst < end) {
*dst++ = *src++; // TODO: optimize
}
} else {
while (dst < end) {
*dst++ = 0;
}
}
transmit(output, channel);
release(output);
}
}
}