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port AudioAnalyzeFFT256 to F32
It's about twice as slow as the I16 version but produces cleaner output
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‎OpenAudio_ArduinoLibrary.h‎

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#include "AudioMixer_F32.h"
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#include "AudioMultiply_F32.h"
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#include "AudioSettings_F32.h"
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#include "analyze_fft256_f32.h"
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#include "input_i2s_f32.h"
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#include "output_i2s_f32.h"
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#include "play_queue_f32.h"

‎analyze_fft256_f32.cpp‎

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/* 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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‎analyze_fft256_f32.h‎

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/* 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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#ifndef analyze_fft256_f32_h_
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#define analyze_fft256_f32_h_
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#include "Arduino.h"
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#include "AudioStream_F32.h"
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#include "arm_math.h"
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// windows.c
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extern "C" {
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extern const int16_t AudioWindowHanning256[];
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extern const int16_t AudioWindowBartlett256[];
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extern const int16_t AudioWindowBlackman256[];
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extern const int16_t AudioWindowFlattop256[];
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extern const int16_t AudioWindowBlackmanHarris256[];
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extern const int16_t AudioWindowNuttall256[];
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extern const int16_t AudioWindowBlackmanNuttall256[];
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extern const int16_t AudioWindowWelch256[];
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extern const int16_t AudioWindowHamming256[];
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extern const int16_t AudioWindowCosine256[];
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extern const int16_t AudioWindowTukey256[];
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}
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class AudioAnalyzeFFT256_F32 : public AudioStream_F32
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{
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public:
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AudioAnalyzeFFT256_F32() : AudioStream_F32(1, inputQueueArray),
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window(AudioWindowHanning256), count(0), outputflag(false) {
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arm_cfft_radix4_init_f32(&fft_inst, 256, 0, 1);
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#if AUDIO_BLOCK_SAMPLES == 128
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prevblock = NULL;
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naverage = 1;
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#elif AUDIO_BLOCK_SAMPLES == 64
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prevblocks[0] = NULL;
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prevblocks[1] = NULL;
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prevblocks[2] = NULL;
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#endif
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}
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bool available() {
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if (outputflag == true) {
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outputflag = false;
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return true;
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}
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return false;
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}
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float read(unsigned int binNumber) {
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if (binNumber > 127) return 0.0;
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return output[binNumber];
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}
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float read(unsigned int binFirst, unsigned int binLast) {
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if (binFirst > binLast) {
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unsigned int tmp = binLast;
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binLast = binFirst;
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binFirst = tmp;
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}
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if (binFirst > 127) return 0.0;
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if (binLast > 127) binLast = 127;
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uint32_t sum = 0;
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do {
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sum += output[binFirst++];
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} while (binFirst <= binLast);
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return sum;
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}
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void averageTogether(uint8_t n) {
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#if AUDIO_BLOCK_SAMPLES == 128
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if (n == 0) n = 1;
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naverage = n;
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#endif
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}
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void windowFunction(const int16_t *w) {
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window = w;
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}
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virtual void update(void);
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float32_t output[128] __attribute__ ((aligned (4)));
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private:
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const int16_t *window;
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#if AUDIO_BLOCK_SAMPLES == 128
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audio_block_f32_t *prevblock;
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#elif AUDIO_BLOCK_SAMPLES == 64
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audio_block_f32_t *prevblocks[3];
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#endif
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float32_t buffer[512] __attribute__ ((aligned (4)));
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#if AUDIO_BLOCK_SAMPLES == 128
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float32_t sum[128];
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uint8_t naverage;
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#endif
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uint8_t count;
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volatile bool outputflag;
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audio_block_f32_t *inputQueueArray[1];
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arm_cfft_radix4_instance_f32 fft_inst;
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};
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#endif

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