/* 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 "analyze_fft256.h" #include "sqrt_integer.h" #include "utility/dspinst.h" // 140312 - PAH - slightly faster copy 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 } } 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 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 }