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analyze_fft256.cpp
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analyze_fft256.cpp
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/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, [email protected]
*
* 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 "analyze_fft256.h"
#include "sqrt_integer.h"
#include "utility/dspinst.h"
// 140312 - PAH - slightly faster copy
__attribute__((unused))
static void copy_to_fft_buffer(void *destination, const void *source)
{
const uint16_t *src = (const uint16_t *)source;
uint32_t *dst = (uint32_t *)destination;
for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
*dst++ = *src++; // real sample plus a zero for imaginary
}
}
__attribute__((unused))
static void apply_window_to_fft_buffer(void *buffer, const void *window)
{
int16_t *buf = (int16_t *)buffer;
const int16_t *win = (int16_t *)window;;
for (int i=0; i < 256; i++) {
int32_t val = *buf * *win++;
//*buf = signed_saturate_rshift(val, 16, 15);
*buf = val >> 15;
buf += 2;
}
}
void AudioAnalyzeFFT256::update(void)
{
audio_block_t *block;
block = receiveReadOnly();
if (!block) return;
#if AUDIO_BLOCK_SAMPLES == 128 && defined (__ARM_ARCH_7EM__)
if (!prevblock) {
prevblock = block;
return;
}
copy_to_fft_buffer(buffer, prevblock->data);
copy_to_fft_buffer(buffer+256, block->data);
//window = AudioWindowBlackmanNuttall256;
//window = NULL;
if (window) apply_window_to_fft_buffer(buffer, window);
arm_cfft_radix4_q15(&fft_inst, buffer);
// G. Heinzel's paper says we're supposed to average the magnitude
// squared, then do the square root at the end.
if (count == 0) {
for (int i=0; i < 128; i++) {
uint32_t tmp = *((uint32_t *)buffer + i);
uint32_t magsq = multiply_16tx16t_add_16bx16b(tmp, tmp);
sum[i] = magsq / naverage;
}
} else {
for (int i=0; i < 128; i++) {
uint32_t tmp = *((uint32_t *)buffer + i);
uint32_t magsq = multiply_16tx16t_add_16bx16b(tmp, tmp);
sum[i] += magsq / naverage;
}
}
if (++count == naverage) {
count = 0;
for (int i=0; i < 128; i++) {
output[i] = sqrt_uint32_approx(sum[i]);
}
outputflag = true;
}
release(prevblock);
prevblock = block;
#elif AUDIO_BLOCK_SAMPLES == 64
if (prevblocks[2] == NULL) {
prevblocks[2] = prevblocks[1];
prevblocks[1] = prevblocks[0];
prevblocks[0] = block;
return;
}
if (count == 0) {
count = 1;
copy_to_fft_buffer(buffer, prevblocks[2]->data);
copy_to_fft_buffer(buffer+128, prevblocks[1]->data);
copy_to_fft_buffer(buffer+256, prevblocks[1]->data);
copy_to_fft_buffer(buffer+384, block->data);
if (window) apply_window_to_fft_buffer(buffer, window);
arm_cfft_radix4_q15(&fft_inst, buffer);
} else {
count = 2;
const uint32_t *p = (uint32_t *)buffer;
for (int i=0; i < 128; i++) {
uint32_t tmp = *p++;
int16_t v1 = tmp & 0xFFFF;
int16_t v2 = tmp >> 16;
output[i] = sqrt_uint32_approx(v1 * v1 + v2 * v2);
}
}
release(prevblocks[2]);
prevblocks[2] = prevblocks[1];
prevblocks[1] = prevblocks[0];
prevblocks[0] = block;
#endif
}