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124 lines
3.9 KiB
124 lines
3.9 KiB
/* Audio Library for Teensy 3.X
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* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
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*
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* Development of this audio library was funded by PJRC.COM, LLC by sales of
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* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
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* open source software by purchasing Teensy or other PJRC products.
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice, development funding notice, and this permission
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* notice shall be included in all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include <Arduino.h>
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#include "analyze_fft256_f32.h"
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#include "utility/dspinst.h"
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static void copy_to_fft_buffer(void *destination, const void *source)
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{
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const float32_t *src = (const float32_t *)source;
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float32_t *dst = (float32_t *)destination;
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for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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*dst++ = *src++; // real sample plus a zero for imaginary
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*dst++ = 0;
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}
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}
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static void apply_window_to_fft_buffer(void *buffer, const void *window)
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{
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float32_t *buf = (float32_t *)buffer;
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const int16_t *win = (int16_t *)window;;
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for (int i=0; i < 256; i++) {
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*buf *= *win++;
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*buf /= 32768;
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buf += 2;
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}
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}
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void AudioAnalyzeFFT256_F32::update(void)
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{
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audio_block_f32_t *block;
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block = receiveReadOnly_f32();
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if (!block) return;
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#if AUDIO_BLOCK_SAMPLES == 128
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if (!prevblock) {
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prevblock = block;
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return;
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}
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copy_to_fft_buffer(buffer, prevblock->data);
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copy_to_fft_buffer(buffer+256, block->data);
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//window = AudioWindowBlackmanNuttall256;
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//window = NULL;
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if (window) apply_window_to_fft_buffer(buffer, window);
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arm_cfft_radix4_f32(&fft_inst, buffer);
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// G. Heinzel's paper says we're supposed to average the magnitude
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// squared, then do the square root at the end.
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if (count == 0) {
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for (int i=0; i < 128; i++) {
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sum[i] = (buffer[i * 2] * buffer[i * 2] + buffer[i * 2 + 1] * buffer[i * 2 + 1]);
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}
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} else {
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for (int i=0; i < 128; i++) {
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sum[i] += (buffer[i * 2] * buffer[i * 2] + buffer[i * 2 + 1] * buffer[i * 2 + 1]);
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}
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}
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if (++count == naverage) {
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count = 0;
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for (int i=0; i < 128; i++) {
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output[i] = sqrtf(sum[i] / naverage) / 64; // I don't know why 64, but a full scale sine wave is 64.
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}
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outputflag = true;
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}
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release(prevblock);
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prevblock = block;
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#elif AUDIO_BLOCK_SAMPLES == 64
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if (prevblocks[2] == NULL) {
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prevblocks[2] = prevblocks[1];
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prevblocks[1] = prevblocks[0];
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prevblocks[0] = block;
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return;
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}
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if (count == 0) {
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count = 1;
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copy_to_fft_buffer(buffer, prevblocks[2]->data);
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copy_to_fft_buffer(buffer+128, prevblocks[1]->data);
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copy_to_fft_buffer(buffer+256, prevblocks[1]->data);
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copy_to_fft_buffer(buffer+384, block->data);
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if (window) apply_window_to_fft_buffer(buffer, window);
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arm_cfft_radix4_q15(&fft_inst, buffer);
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} else {
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count = 2;
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const uint32_t *p = (uint32_t *)buffer;
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for (int i=0; i < 128; i++) {
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uint32_t tmp = *p++;
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int16_t v1 = tmp & 0xFFFF;
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int16_t v2 = tmp >> 16;
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output[i] = sqrt_uint32_approx(v1 * v1 + v2 * v2);
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}
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}
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release(prevblocks[2]);
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prevblocks[2] = prevblocks[1];
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prevblocks[1] = prevblocks[0];
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prevblocks[0] = block;
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#endif
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}
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