diff --git a/analyze_fft1024_F32.cpp b/analyze_fft1024_F32.cpp index 814be64..9be3237 100644 --- a/analyze_fft1024_F32.cpp +++ b/analyze_fft1024_F32.cpp @@ -1,5 +1,5 @@ /* analyze_fft1024_F32.cpp Converted from Teensy I16 Audio Library - * This version uses floating point F32 inputs + * This version uses float F32 inputs. See comments at analyze_fft1024_F32.h * * Conversion parts copyright (c) Bob Larkin 2021 * @@ -57,7 +57,6 @@ static void apply_window_to_fft_buffer(void *buffer, const void *window) void AudioAnalyzeFFT1024_F32::update(void) { audio_block_f32_t *block; - block = receiveReadOnly_f32(); if (!block) return; @@ -103,7 +102,7 @@ void AudioAnalyzeFFT1024_F32::update(void) copy_to_fft_buffer(fft_buffer+0x600, blocklist[6]->data); copy_to_fft_buffer(fft_buffer+0x700, blocklist[7]->data); - if (window) + if (pWin) apply_window_to_fft_buffer(fft_buffer, window); arm_cfft_radix4_f32(&fft_inst, fft_buffer); diff --git a/analyze_fft1024_F32.h b/analyze_fft1024_F32.h index 83217a4..c2db46f 100644 --- a/analyze_fft1024_F32.h +++ b/analyze_fft1024_F32.h @@ -26,9 +26,42 @@ * THE SOFTWARE. */ - /* Moved directly I16 to F32. Bob Larkin 16 Feb 2021 - * Only Hann window for now. - */ +/* Moved directly I16 to F32. Bob Larkin 16 Feb 2021 + * Does real input FFT of 1024 points. Output is not audio, and is magnitude + * only. Multiple output formats of RMS (same as I16 version, and default), + * Power or dBFS (full scale). Output can be bin by bin or a pointer to + * the output array is available. Several window functions are provided by + * in-class design, or a custom window can be provided from the INO. + * + * Functions (See comments below and #defines above: + * bool available() + * float read(unsigned int binNumber) + * float read(unsigned int binFirst, unsigned int binLast) + * int windowFunction(int wNum) + * int windowFunction(int wNum, float _kdb) // Kaiser only + * float* getData(void) + * float* getWindow(void) + * void putWindow(float *pwin) + * void setOutputType(int _type) + * + * Timing, max is longest update() time: + * T3.6 Windowed, RMS out, 1016 uSec max + * T3.6 Windowed, Power Out, 975 uSec max + * T3.6 Windowed, dBFS out, 1591 uSec max + * No Window saves 60 uSec on T3.6 for any output. + * T4.0 Windowed, RMS Out, 149 uSec + * + * Scaling: + * Full scale for floating point DSP is a nebulous concept. Normally the + * full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine + * wave centered in frequency on a bin and of FS amplitude, the power + * at that center bin will grow by 1024^2/4 = 262144 without windowing. + * Windowing loss cuts this down. The RMS level can grow to sqrt(262144) + * or 512. The dBFS has been scaled to make this max value 0 dBFS by + * removing 54.2 dB. With floating point, the dynamic range is maintained + * no matter how it is scaled, but this factor needs to be considered + * when building the INO. + */ #ifndef analyze_fft1024_F32_h_ #define analyze_fft1024_F32_h_ @@ -36,24 +69,17 @@ #include "Arduino.h" #include "AudioStream_F32.h" #include "arm_math.h" +#include "mathDSP_F32.h" + #define FFT_RMS 0 #define FFT_POWER 1 #define FFT_DBFS 2 -/* // windows.c -extern "C" { -extern const int16_t AudioWindowHanning1024[]; -extern const int16_t AudioWindowBartlett1024[]; -extern const int16_t AudioWindowBlackman1024[]; -extern const int16_t AudioWindowFlattop1024[]; -extern const int16_t AudioWindowBlackmanHarris1024[]; -extern const int16_t AudioWindowNuttall1024[]; -extern const int16_t AudioWindowBlackmanNuttall1024[]; -extern const int16_t AudioWindowWelch1024[]; -extern const int16_t AudioWindowHamming1024[]; -extern const int16_t AudioWindowCosine1024[]; -extern const int16_t AudioWindowTukey1024[]; ) -*/ +#define NO_WINDOW 0 +#define AudioWindowNone 0 +#define AudioWindowHanning1024 1 +#define AudioWindowKaiser1024 2 +#define AudioWindowBlackmanHarris1024 3 class AudioAnalyzeFFT1024_F32 : public AudioStream_F32 { //GUI: inputs:1, outputs:0 //this line used for automatic generation of GUI node @@ -63,6 +89,7 @@ public: arm_cfft_radix4_init_f32(&fft_inst, 1024, 0, 1); useHanningWindow(); // Revisit this for more flexibility <<<<< } + bool available() { if (outputflag == true) { outputflag = false; @@ -70,10 +97,14 @@ public: } return false; } + float read(unsigned int binNumber) { if (binNumber>511 || binNumber<0) return 0.0; return output[binNumber]; } + + // Return sum of several bins. Normally use with power output. + // This produces the equivalent of bigger bins. float read(unsigned int binFirst, unsigned int binLast) { if (binFirst > binLast) { unsigned int tmp = binLast; @@ -89,34 +120,115 @@ public: return (float)sum * (1.0 / 16384.0); } - void useHanningWindow(void) { - for (int i=0; i < 1024; i++) { - // 2*PI/1023 = 0.006141921 - window[i] = 0.5*(1.0 - cosf(0.006141921f*(float)i)); - } + int windowFunction(int wNum) { + if(wNum == AudioWindowKaiser1024) + return -1; // Kaiser needs the kdb + windowFunction(wNum, 0.0f); + return 0; } -// void windowFunction(const float *w) { -// window = w; -// } + int windowFunction(int wNum, float _kdb) { + float kd; + pWin = window; + if(wNum == NO_WINDOW) + pWin = NULL; + else if (wNum == AudioWindowKaiser1024) { + if(_kdb<20.0f) + kd = 20.0f; + else + kd = _kdb; + useKaiserWindow(kd); + } + else if (wNum == AudioWindowBlackmanHarris1024) + useBHWindow(); + else + useHanningWindow(); // Default + return 0; + } + + // Fast pointer transfer. Be aware that the data will go away + // after the next 512 data points occur. + float* getData(void) { + return output; + } + + // You can use this to design windows + float* getWindow(void) { + return window; + } + // Bring custom window from the INO + void putWindow(float *pwin) { + float *p = window; + for(int i=0; i<1024; i++) + *p++ = *pwin++; + } + + // Output RMS (default) Power or dBFS void setOutputType(int _type) { outputType = _type; } virtual void update(void); - + +private: float output[512]; -private: - // void init(void); - float window[1024]; int doPrint = 0; + float window[1024]; + float *pWin = window; audio_block_f32_t *blocklist[8]; float fft_buffer[2048]; uint8_t state = 0; bool outputflag = false; audio_block_f32_t *inputQueueArray[1]; arm_cfft_radix4_instance_f32 fft_inst; - int outputType = FFT_RMS; //Same type as I16 version has -}; + int outputType = FFT_RMS; //Same type as I16 version init + // The Hann window is a good all-around window + void useHanningWindow(void) { + for (int i=0; i < 1024; i++) { + // 2*PI/1023 = 0.006141921 + window[i] = 0.5*(1.0 - cosf(0.006141921f*(float)i)); + } + } + + // Blackman-Harris produces a first sidelobe more than 90 dB down. + // The price is a bandwidth of about 2 bins. Very useful at times. + void useBHWindow(void) { + for (int i=0; i < 1024; i++) { + float kx = 0.006141921; // 2*PI/1023 + int ix = (float) i; + window[i] = 0.35875 - + 0.48829*cosf( kx*ix) + + 0.14128*cosf(2.0f*kx*ix) - + 0.01168*cosf(3.0f*kx*ix); + } + } + + /* The windowing function here is that of James Kaiser. This has a number + * of desirable features. The sidelobes drop off as the frequency away from a transition. + * Also, the tradeoff of sidelobe level versus cutoff rate is variable. + * Here we specify it in terms of kdb, the highest sidelobe, in dB, next to a sharp cutoff. For + * calculating the windowing vector, we need a parameter beta, found as follows: + */ + void useKaiserWindow(float kdb) { + float32_t beta, kbes, xn2; + mathDSP_F32 mathEqualizer; // For Bessel function + + if (kdb < 20.0f) + beta = 0.0; + else + beta = -2.17+0.17153*kdb-0.0002841*kdb*kdb; // Within a dB or so + + // Note: i0f is the fp zero'th order modified Bessel function (see mathDSP_F32.h) + kbes = 1.0f / mathEqualizer.i0f(beta); // An additional derived parameter used in loop + for (int n=0; n<512; n++) { + xn2 = 0.5f+(float32_t)n; + // 4/(1023^2)=0.00000382215877f + xn2 = 0.00000382215877f*xn2*xn2; + window[511 - n]=kbes*(mathEqualizer.i0f(beta*sqrtf(1.0-xn2))); + window[512 + n] = window[511 - n]; + } + } + +}; #endif