wirtz
/
OpenAudio_ArduinoLibrary
mirror of https://github.com/chipaudette/OpenAudio_ArduinoLibrary.git
1
0
Fork 0
Code Issues Releases Wiki Activity

Add files from Tympan_Library

Browse Source
pull/5/head
Chip Audette 8 years ago
parent 5a43e34b15
commit 5466e69d17
29 changed files with 2255 additions and 257 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Show Stats Download Patch File Download Diff File
  1. 118
      AudioCalcEnvelope_F32.h
  2. 173
      AudioCalcGainWDRC_F32.h
  3. 274
      AudioConfigFIRFilterBank_F32.h
  4. 4
      AudioConvert_F32.h
  5. 278
      AudioEffectCompWDRC_F32.h
  6. 49
      AudioEffectEmpty_F32.h
  7. 24
      AudioEffectGain_F32.h
  8. 101
      AudioFilterFIR_F32.h
  9. 8
      AudioFilterIIR_F32.h
  10. 60
      AudioMixer_F32.cpp
  11. 63
      AudioMixer_F32.h
  12. 9
      AudioStream_F32.cpp
  13. 69
      AudioStream_F32.h
  14. 7
      OpenAudio_ArduinoLibrary.h
  15. 11
      control_tlv320aic3206.cpp
  16. 27
      control_tlv320aic3206.h
  17. 53
      input_i2s_f32.cpp
  18. 11
      input_i2s_f32.h
  19. 140
      output_i2s_f32.cpp
  20. 16
      output_i2s_f32.h
  21. 167
      synth_pinknoise_f32.cpp
  22. 99
      synth_pinknoise_f32.h
  23. 60
      synth_sine_f32.cpp
  24. 39
      synth_sine_f32.h
  25. 35
      synth_waveform_F32.h
  26. 125
      synth_whitenoise_f32.cpp
  27. 89
      synth_whitenoise_f32.h
  28. 384
      utility/rfft.c
  29. 19
      utility/textAndStringUtils.h

118
AudioCalcEnvelope_F32.h
Unescape View File

@ -0,0 +1,118 @@
/*
* AudioCalcEnvelope_F32
*
* Created: Chip Audette, Feb 2017
* Purpose: This module extracts the envelope of the audio signal.
* Derived From: Core envelope extraction algorithm is from "smooth_env"
* WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro
* As of Feb 2017, CHAPRO license is listed as "Creative Commons?"
*
* This processes a single stream fo audio data (ie, it is mono)
*
* MIT License. use at your own risk.
*/
#ifndef _AudioCalcEnvelope_F32_h
#define _AudioCalcEnvelope_F32_h
#include <arm_math.h> //ARM DSP extensions. for speed!
#include <AudioStream_F32.h>
class AudioCalcEnvelope_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:calc_envelope
public:
//default constructor
AudioCalcEnvelope_F32(void) : AudioStream_F32(1, inputQueueArray_f32),
sample_rate_Hz(AUDIO_SAMPLE_RATE) { setDefaultValues(); };
AudioCalcEnvelope_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32),
sample_rate_Hz(settings.sample_rate_Hz) { setDefaultValues(); };
//here's the method that does all the work
void update(void) {
//get the input audio data block
audio_block_f32_t *in_block = AudioStream_F32::receiveReadOnly_f32();
if (!in_block) return;
//check format
if (in_block->fs_Hz != sample_rate_Hz) {
Serial.println("AudioComputeEnvelope_F32: *** WARNING ***: Data sample rate does not match expected.");
Serial.println("AudioComputeEnvelope_F32: Changing sample rate.");
setSampleRate_Hz(in_block->fs_Hz);
}
//prepare an output data block
audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
if (!out_block) return;
// //////////////////////add your processing here!
smooth_env(in_block->data, out_block->data, in_block->length);
out_block->length = in_block->length; out_block->fs_Hz = in_block->fs_Hz;
//transmit the block and be done
AudioStream_F32::transmit(out_block);
AudioStream_F32::release(out_block);
AudioStream_F32::release(in_block);
}
//compute the smoothed signal envelope
//compute the envelope of the signal, not of the signal power)
void smooth_env(float x[], float y[], const int n) {
float xab, xpk;
int k;
// find envelope of x and return as y
//xpk = *ppk; // start with previous xpk
xpk = state_ppk;
for (k = 0; k < n; k++) {
xab = (x[k] >= 0.0f) ? x[k] : -x[k];
if (xab >= xpk) {
xpk = alfa * xpk + (1.f-alfa) * xab;
} else {
xpk = beta * xpk;
}
y[k] = xpk;
}
//*ppk = xpk; // save xpk for next time
state_ppk = xpk;
}
//convert time constants from seconds to unitless parameters, from CHAPRO, agc_prepare.c
void setAttackRelease_msec(const float atk_msec, const float rel_msec) {
given_attack_msec = atk_msec;
given_release_msec = rel_msec;
// convert ANSI attack & release times to filter time constants
float ansi_atk = 0.001f * atk_msec * sample_rate_Hz / 2.425f;
float ansi_rel = 0.001f * rel_msec * sample_rate_Hz / 1.782f;
alfa = (float) (ansi_atk / (1.0f + ansi_atk));
beta = (float) (ansi_rel / (10.f + ansi_rel));
}
void setDefaultValues(void) {
float32_t attack_msec = 5.0f;
float32_t release_msec = 50.0f;
setAttackRelease_msec(attack_msec, release_msec);
state_ppk = 0; //initialize
}
void setSampleRate_Hz(const float &fs_Hz) {
//change params that follow sample rate
sample_rate_Hz = fs_Hz;
}
void resetStates(void) { state_ppk = 1.0; }
float getCurrentLevel(void) { return state_ppk; }
private:
audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
float32_t sample_rate_Hz;
float32_t given_attack_msec, given_release_msec;
float32_t alfa, beta; //time constants, but in terms of samples, not seconds
float32_t state_ppk = 1.0f;
};
#endif

173
AudioCalcGainWDRC_F32.h
Unescape View File

@ -0,0 +1,173 @@
/*
* AudioCalcGainWDRC_F32
*
* Created: Chip Audette, Feb 2017
* Purpose: This module calculates the gain needed for wide dynamic range compression.
* Derived From: Core algorithm is from "WDRC_circuit"
* WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro
* As of Feb 2017, CHAPRO license is listed as "Creative Commons?"
*
* This processes a single stream fo audio data (ie, it is mono)
*
* MIT License. use at your own risk.
*/
#ifndef _AudioCalcGainWDRC_F32_h
#define _AudioCalcGainWDRC_F32_h
#include <arm_math.h> //ARM DSP extensions. for speed!
#include <AudioStream_F32.h>
typedef struct {
float attack; // attack time (ms), unused in this class
float release; // release time (ms), unused in this class
float fs; // sampling rate (Hz), set through other means in this class
float maxdB; // maximum signal (dB SPL)...I think this is the SPL corresponding to signal with rms of 1.0
float tkgain; // compression-start gain
float tk; // compression-start kneepoint
float cr; // compression ratio
float bolt; // broadband output limiting threshold
} CHA_WDRC;
class AudioCalcGainWDRC_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:calc_WDRCGain
public:
//default constructor
AudioCalcGainWDRC_F32(void) : AudioStream_F32(1, inputQueueArray_f32) { setDefaultValues(); };
//here's the method that does all the work
void update(void) {
//get the input audio data block
audio_block_f32_t *in_block = AudioStream_F32::receiveReadOnly_f32(); // must be the envelope!
if (!in_block) return;
//prepare an output data block
audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
if (!out_block) return;
// //////////////////////add your processing here!
calcGainFromEnvelope(in_block->data, out_block->data, in_block->length);
out_block->length = in_block->length; out_block->fs_Hz = in_block->fs_Hz;
//transmit the block and be done
AudioStream_F32::transmit(out_block);
AudioStream_F32::release(out_block);
AudioStream_F32::release(in_block);
}
void calcGainFromEnvelope(float *env, float *gain_out, const int n) {
//env = input, signal envelope (not the envelope of the power, but the envelope of the signal itslef)
//gain = output, the gain in natural units (not power, not dB)
//n = input, number of samples to process in each vector
//prepare intermediate data block
audio_block_f32_t *env_dB_block = AudioStream_F32::allocate_f32();
if (!env_dB_block) return;
//convert to dB
for (int k=0; k < n; k++) env_dB_block->data[k] = maxdB + db2(env[k]); //maxdb in the private section
// apply wide-dynamic range compression
WDRC_circuit_gain(env_dB_block->data, gain_out, n, tkgn, tk, cr, bolt);
AudioStream_F32::release(env_dB_block);
}
//original call to WDRC_circuit
//void WDRC_circuit(float *x, float *y, float *pdb, int n, float tkgn, float tk, float cr, float bolt)
//void WDRC_circuit(float *orig_signal, float *signal_out, float *env_dB, int n, float tkgn, float tk, float cr, float bolt)
//modified to output the gain instead of the fully processed signal
void WDRC_circuit_gain(float *env_dB, float *gain_out, const int n,
const float tkgn, const float tk, const float cr, const float bolt) {
float gdb, tkgo, pblt;
int k;
float *pdb = env_dB; //just rename it to keep the code below unchanged
float tk_tmp = tk;
if ((tk_tmp + tkgn) > bolt) {
tk_tmp = bolt - tkgn;
}
tkgo = tkgn + tk_tmp * (1.0f - 1.0f / cr);
pblt = cr * (bolt - tkgo);
const float cr_const = ((1.0f / cr) - 1.0f);
for (k = 0; k < n; k++) {
if ((pdb[k] < tk_tmp) && (cr >= 1.0f)) {
gdb = tkgn;
} else if (pdb[k] > pblt) {
gdb = bolt + ((pdb[k] - pblt) / 10.0f) - pdb[k];
} else {
gdb = cr_const * pdb[k] + tkgo;
}
gain_out[k] = undb2(gdb);
//y[k] = x[k] * undb2(gdb); //apply the gain
}
}
void setDefaultValues(void) {
CHA_WDRC gha = {1.0f, // attack time (ms), IGNORED HERE
50.0f, // release time (ms), IGNORED HERE
24000.0f, // fs, sampling rate (Hz), IGNORED HERE
119.0f, // maxdB, maximum signal (dB SPL)
0.0f, // tkgain, compression-start gain
105.0f, // tk, compression-start kneepoint
10.0f, // cr, compression ratio
105.0f // bolt, broadband output limiting threshold
};
//setParams(gha.maxdB, gha.tkgain, gha.cr, gha.tk, gha.bolt); //also sets calcEnvelope
setParams_from_CHA_WDRC(&gha);
}
void setParams_from_CHA_WDRC(CHA_WDRC *gha) {
setParams(gha->maxdB, gha->tkgain, gha->cr, gha->tk, gha->bolt); //also sets calcEnvelope
}
void setParams(float _maxdB, float _tkgain, float _cr, float _tk, float _bolt) {
maxdB = _maxdB;
tkgn = _tkgain;
tk = _tk;
cr = _cr;
bolt = _bolt;
}
static float undb2(const float &x) { return expf(0.11512925464970228420089957273422f*x); } //faster: exp(log(10.0f)*x/20); this is exact
static float db2(const float &x) { return 6.020599913279623f*log2f_approx(x); } //faster: 20*log2_approx(x)/log2(10); this is approximate
/* ----------------------------------------------------------------------
** Fast approximation to the log2() function. It uses a two step
** process. First, it decomposes the floating-point number into
** a fractional component F and an exponent E. The fraction component
** is used in a polynomial approximation and then the exponent added
** to the result. A 3rd order polynomial is used and the result
** when computing db20() is accurate to 7.984884e-003 dB.
** ------------------------------------------------------------------- */
//https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
static float log2f_approx(float X) {
//float *C = &log2f_approx_coeff[0];
float Y;
float F;
int E;
// This is the approximation to log2()
F = frexpf(fabsf(X), &E);
// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
Y = 1.23149591368684f; //C[0]
Y *= F;
Y += -4.11852516267426f; //C[1]
Y *= F;
Y += 6.02197014179219f; //C[2]
Y *= F;
Y += -3.13396450166353f; //C[3]
Y += E;
return(Y);
}
private:
audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
float maxdB, tkgn, tk, cr, bolt;
};
#endif

274
AudioConfigFIRFilterBank_F32.h
Unescape View File

@ -0,0 +1,274 @@
/*
* fir_filterbank.h
*
* Created: Chip Audette, Creare LLC, Feb 2017
* Primarly built upon CHAPRO "Generic Hearing Aid" from
* Boys Town National Research Hospital (BTNRH): https://github.com/BTNRH/chapro
*
* License: MIT License. Use at your own risk.
*
*/
#ifndef AudioConfigFIRFilterBank_F32_h
#define AudioConfigFIRFilterBank_F32_h
#include "utility/rfft.c"
#define fmove(x,y,n) memmove(x,y,(n)*sizeof(float))
#define fcopy(x,y,n) memcpy(x,y,(n)*sizeof(float))
#define fzero(x,n) memset(x,0,(n)*sizeof(float))
class AudioConfigFIRFilterBank_F32 {
//GUI: inputs:0, outputs:0 //this line used for automatic generation of GUI node
//GUI: shortName:config_FIRbank
public:
AudioConfigFIRFilterBank_F32(void) {
}
AudioConfigFIRFilterBank_F32(const int n_chan, const int n_fir, const float sample_rate_Hz, float *corner_freq, float *filter_coeff) {
createFilterCoeff(n_chan, n_fir, sample_rate_Hz, corner_freq, filter_coeff);
}
//createFilterCoeff:
// Purpose: create all of the FIR filter coefficients for the FIR filterbank
// Syntax: createFilterCoeff(n_chan, n_fir, sample_rate_Hz, corner_freq, filter_coeff)
// int n_chan (input): number of channels (number of filters) you desire. Must be 2 or greater
// int n_fir (input): length of each FIR filter (should probably be 8 or greater)
// float sample_rate_Hz (input): sample rate of your system (used to scale the corner_freq values)
// float *corner_freq (input): array of frequencies (Hz) seperating each band in your filter bank.
// should contain n_chan-1 values because it should exclude the bottom (0 Hz) and the top
// (Nyquist) as those values are already assumed by this routine. An valid example is below:
// int n_chan = 8; float cf[] = {317.1666, 502.9734, 797.6319, 1264.9, 2005.9, 3181.1, 5044.7};
// float *filter_coeff (output): array of FIR filter coefficients that are computed by this
// routine. You must have pre-allocated the array such as: float filter_coeff[N_CHAN][N_FIR];
//Optional Usage: if you want 8 default filters spaced logarithmically, use: float *corner_freq = NULL
void createFilterCoeff(const int n_chan, const int n_fir, const float sample_rate_Hz, float *corner_freq, float *filter_coeff) {
float *cf = corner_freq;
int flag__free_cf = 0;
if (cf == NULL) {
//compute corner frequencies that are logarithmically spaced
cf = (float *) calloc(n_chan, sizeof(float));
flag__free_cf = 1;
computeLogSpacedCornerFreqs(n_chan, sample_rate_Hz, cf);
}
const int window_type = 0; //0 = Hamming
fir_filterbank(filter_coeff, cf, n_chan, n_fir, window_type, sample_rate_Hz);
if (flag__free_cf) free(cf);
}
//compute frequencies that space zero to nyquist. Leave zero off, because it is assumed to exist in the later code.
//example of an *8* channel set of frequencies: cf = {317.1666, 502.9734, 797.6319, 1264.9, 2005.9, 3181.1, 5044.7}
void computeLogSpacedCornerFreqs(const int n_chan, const float sample_rate_Hz, float *cf) {
float cf_8_band[] = {317.1666, 502.9734, 797.6319, 1264.9, 2005.9, 3181.1, 5044.7};
float scale_fac = expf(logf(cf_8_band[6]/cf_8_band[0]) / ((float)(n_chan-2)));
//Serial.print("MakeFIRFilterBank: computeEvenlySpacedCornerFreqs: scale_fac = "); Serial.println(scale_fac);
cf[0] = cf_8_band[0];
//Serial.println("MakeFIRFilterBank: computeEvenlySpacedCornerFreqs: cf = ");Serial.print(cf[0]); Serial.print(", ");
for (int i=1; i < n_chan-1; i++) {
cf[i] = cf[i-1]*scale_fac;
//Serial.print(cf[i]); Serial.print(", ");
}
//Serial.println();
}
private:
int nextPowerOfTwo(int n) {
const int n_out_vals = 8;
int out_vals[n_out_vals] = {8, 16, 32, 64, 128, 256, 512, 1024};
if (n < out_vals[0]) return out_vals[0];
for (int i=1;i<n_out_vals; i++) {
if ((n > out_vals[i-1]) & (n <= out_vals[i])) {
return out_vals[i];
}
}
return n;
}
void fir_filterbank(float *bb, float *cf, const int nc, const int nw_orig, const int wt, const float sr)
{
double p, w, a = 0.16, sm = 0;
float *ww, *bk, *xx, *yy;
int j, k, kk, nt, nf, ns, *be;
int nw = nextPowerOfTwo(nw_orig);
Serial.print("fir_filterbank: nw_orig = "); Serial.print(nw_orig);
Serial.print(", nw = "); Serial.println(nw);
nt = nw * 2;
nf = nw + 1;
ns = nf * 2;
be = (int *) calloc(nc + 1, sizeof(int));
ww = (float *) calloc(nw, sizeof(float));
xx = (float *) calloc(ns, sizeof(float));
yy = (float *) calloc(ns, sizeof(float));
// window
for (j = 0; j < nw; j++) ww[j]=0.0f; //clear
for (j = 0; j < nw_orig; j++) {
p = M_PI * (2.0 * j - nw_orig) / nw_orig;
if (wt == 0) {
w = 0.54 + 0.46 * cos(p); // Hamming
} else {
w = (1 - a + cos(p) + a * cos(2 * p)) / 2; // Blackman
}
sm += w;
ww[j] = (float) w;
}
// frequency bands...add the DC-facing band and add the Nyquist-facing band
be[0] = 0;
for (k = 1; k < nc; k++) {
kk = round(nf * cf[k - 1] * (2 / sr));
be[k] = (kk > nf) ? nf : kk;
}
be[nc] = nf;
// channel tranfer functions
fzero(xx, ns);
xx[nw_orig / 2] = 1; //make a single-sample impulse centered on our eventual window
cha_fft_rc(xx, nt);
for (k = 0; k < nc; k++) {
fzero(yy, ns); //zero the temporary output
//int nbins = (be[k + 1] - be[k]) * 2; Serial.print("fir_filterbank: chan ");Serial.print(k); Serial.print(", nbins = ");Serial.println(nbins);
fcopy(yy + be[k] * 2, xx + be[k] * 2, (be[k + 1] - be[k]) * 2); //copy just our passband
cha_fft_cr(yy, nt); //IFFT back into the time domain
// apply window to iFFT of bandpass
for (j = 0; j < nw; j++) {
yy[j] *= ww[j];
}
bk = bb + k * nw_orig; //pointer to location in output array
fcopy(bk, yy, nw_orig); //copy the filter coefficients to the output array
//print out the coefficients
//for (int i=0; i<nw; i++) { Serial.print(yy[i]*1000.0f);Serial.print(" "); }; Serial.println();
}
free(be);
free(ww);
free(xx);
free(yy);
}
};
#endif
// static CHA_DSL dsl = {5, 50, 119, 0, 8,
// {317.1666,502.9734,797.6319,1264.9,2005.9,3181.1,5044.7}, //log spaced frequencies.
// {-13.5942,-16.5909,-3.7978,6.6176,11.3050,23.7183,35.8586,37.3885},
// {0.7,0.9,1,1.1,1.2,1.4,1.6,1.7},
// {32.2,26.5,26.7,26.7,29.8,33.6,34.3,32.7},
// {78.7667,88.2,90.7,92.8333,98.2,103.3,101.9,99.8}
// };
// //x is the input waveform
// //y is the processed waveform
// //n is the length of the waveform
// //fs is the sample rate...24000 Hz
// //dsl are the settings for each band
// t1 = amplify(x, y, n, fs, &dsl);
//amplify(float *x, float *y, int n, double fs, CHA_DSL *dsl)
//{
// int nc;
// static int nw = 256; // window size
// static int cs = 32; // chunk size
// static int wt = 0; // window type: 0=Hamming, 1=Blackman
// static void *cp[NPTR] = {0};
// static CHA_WDRC gha = {1, 50, 24000, 119, 0, 105, 10, 105};
//
// nc = dsl->nchannel; //8?
// cha_firfb_prepare(cp, dsl->cross_freq, nc, fs, nw, wt, cs);
// cha_agc_prepare(cp, dsl, &gha);
// sp_tic();
// WDRC(cp, x, y, n, nc);
// return (sp_toc());
//}
//FUNC(int)
//cha_firfb_prepare(CHA_PTR cp, double *cf, int nc, double fs,
// int nw, int wt, int cs)
//{
// float *bb;
// int ns, nt;
//
// if (cs <= 0) {
// return (1);
// }
// cha_prepare(cp);
// CHA_IVAR[_cs] = cs; //cs = 32
// CHA_DVAR[_fs] = fs; //fs = 24000
// // allocate window buffers
// CHA_IVAR[_nw] = nw; //nw = 256
// CHA_IVAR[_nc] = nc; //nc = 32
// nt = nw * 2; //nt = 256*2 = 512
// ns = nt + 2; //ns = 512+2 = 514
// cha_allocate(cp, ns, sizeof(float), _ffxx); //allocate for input
// cha_allocate(cp, ns, sizeof(float), _ffyy); //allocate for output
// cha_allocate(cp, nc * (nw + cs), sizeof(float), _ffzz); //allocate per channel
// // compute FIR-filterbank coefficients
// bb = calloc(nc * nw, sizeof(float)); //allocate for filter coeff (256 long, 8 channels)
// fir_filterbank(bb, cf, nc, nw, wt, fs); //make the fir filter bank
// // Fourier-transform FIR coefficients
// if (cs < nw) { // short chunk
// fir_transform_sc(cp, bb, nc, nw, cs);
// } else { // long chunk
// fir_transform_lc(cp, bb, nc, nw, cs);
// }
// free(bb);
//
// return (0);
//}
// fir_filterbank( float *bb, double *cf, int nc, int nw, int wt, double sr)
// filter coeff, corner freqs, 8, 256, 0, 24000)
//{
// double p, w, a = 0.16, sm = 0;
// float *ww, *bk, *xx, *yy;
// int j, k, kk, nt, nf, ns, *be;
//
// nt = nw * 2; //nt = 256*2 = 512
// nf = nw + 1; //nyquist frequency bin is 256+1 = 257
// ns = nf * 2; //when complex, number values to carry is nyquist * 2 = 514
// be = (int *) calloc(nc + 1, sizeof(int));
// ww = (float *) calloc(nw, sizeof(float)); //window is 256 long
// xx = (float *) calloc(ns, sizeof(float)); //input data is 514 points long
// yy = (float *) calloc(ns, sizeof(float)); //output data is 514 points long
// // window
// for (j = 0; j < nw; j++) { //nw = 256
// p = M_PI * (2.0 * j - nw) / nw; //phase for computing window, radians
// if (wt == 0) { //wt is zero
// w = 0.54 + 0.46 * cos(p); // Hamming
// } else {
// w = (1 - a + cos(p) + a * cos(2 * p)) / 2; // Blackman
// }
// sm += w; //sum the window value. Doesn't appear to be used anywhere
// ww[j] = (float) w; //save the windowing coefficient...there are 256 of them
// }
// // frequency bands
// be[0] = 0; //first channel is DC bin
// for (k = 1; k < nc; k++) { //loop over the rest of the 8 channels
// kk = round(nf * cf[k - 1] * (2 / sr)); //get bin of the channel (upper?) corner frequency...assumes factor of two zero-padding?
// be[k] = (kk > nf) ? nf : kk; //make sure we don't go above the nyquist bin (bin 257, assuming a 512 FFT)
// }
// be[nc] = nf; //the last one is the nyquist freuquency
// // channel tranfer functions
// fzero(xx, ns); //zero the xx vector
// xx[nw / 2] = 1; //create an impulse in the middle of the (non-overlapped part of the) time-domain...sample 129
// cha_fft_rc(xx, nt); //convert to frequency domain..512 points long
// for (k = 0; k < nc; k++) { //loop over each channel
// bk = bb + k * nw; //bin index for this channel
// fzero(yy, ns); //zero out the output bins
// fcopy(yy + be[k] * 2, xx + be[k] * 2, (be[k + 1] - be[k]) * 2); //copy just the desired frequeny bins in our passband
// cha_fft_cr(yy, nt); //convert back to time domain
// // apply window to iFFT of bandpass
// for (j = 0; j < nw; j++) {
// yy[j] *= ww[j];
// }
// fcopy(bk, yy, nw); //copy output into the output filter...just the 256 points
// }
// free(be);
// free(ww);
// free(xx);
// free(yy);
//}

4
AudioConvert_F32.h
Unescape View File

@ -24,7 +24,7 @@ class AudioConvert_I16toF32 : public AudioStream_F32 //receive Int and transmits
}
//convert to float
convertAudio_I16toF32(int_block, float_block, AUDIO_BLOCK_SAMPLES);
convertAudio_I16toF32(int_block, float_block, float_block->length);
//transmit the audio and return it to the system
AudioStream_F32::transmit(float_block,0);
@ -65,7 +65,7 @@ class AudioConvert_F32toI16 : public AudioStream_F32 //receive Float and transmi
}
//convert back to int16
convertAudio_F32toI16(float_block, int_block, AUDIO_BLOCK_SAMPLES);
convertAudio_F32toI16(float_block, int_block, float_block->length);
//return audio to the system
AudioStream::transmit(int_block);

278
AudioEffectCompWDRC_F32.h
Unescape View File

@ -0,0 +1,278 @@
/*
* AudioEffectCompWDR_F32: Wide Dynamic Rnage Compressor
*
* Created: Chip Audette (OpenAudio) Feb 2017
* Derived From: WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro
* As of Feb 2017, CHAPRO license is listed as "Creative Commons?"
*
* MIT License. Use at your own risk.
*
*/
#ifndef _AudioEffectCompWDRC_F32
#define _AudioEffectCompWDRC_F32
#include <Arduino.h>
#include <AudioStream_F32.h>
#include <arm_math.h>
#include <AudioCalcEnvelope_F32.h>
#include "AudioCalcGainWDRC_F32.h" //has definition of CHA_WDRC
#include "utility/textAndStringUtils.h"
// from CHAPRO cha_ff.h
#define DSL_MXCH 32
//class CHA_DSL {
typedef struct {
//public:
//CHA_DSL(void) {};
//static const int DSL_MXCH = 32; // maximum number of channels
float attack; // attack time (ms)
float release; // release time (ms)
float maxdB; // maximum signal (dB SPL)
int ear; // 0=left, 1=right
int nchannel; // number of channels
float cross_freq[DSL_MXCH]; // cross frequencies (Hz)
float tkgain[DSL_MXCH]; // compression-start gain
float cr[DSL_MXCH]; // compression ratio
float tk[DSL_MXCH]; // compression-start kneepoint
float bolt[DSL_MXCH]; // broadband output limiting threshold
} CHA_DSL;
/* int parseStringIntoDSL(String &text_buffer) {
int position = 0;
float foo_val;
const bool print_debug = false;
if (print_debug) Serial.println("parseTextAsDSL: values from file:");
position = parseNextNumberFromString(text_buffer, position, foo_val);
attack = foo_val;
if (print_debug) { Serial.print(" attack: "); Serial.println(attack); }
position = parseNextNumberFromString(text_buffer, position, foo_val);
release = foo_val;
if (print_debug) { Serial.print(" release: "); Serial.println(release); }
position = parseNextNumberFromString(text_buffer, position, foo_val);
maxdB = foo_val;
if (print_debug) { Serial.print(" maxdB: "); Serial.println(maxdB); }
position = parseNextNumberFromString(text_buffer, position, foo_val);
ear = int(foo_val + 0.5); //round
if (print_debug) { Serial.print(" ear: "); Serial.println(ear); }
position = parseNextNumberFromString(text_buffer, position, foo_val);
nchannel = int(foo_val + 0.5); //round
if (print_debug) { Serial.print(" nchannel: "); Serial.println(nchannel); }
//check to see if the number of channels is acceptable.
if ((nchannel < 0) || (nchannel > DSL_MXCH)) {
if (print_debug) Serial.print(" : channel number is too big (or negative). stopping.");
return -1;
}
//read the cross-over frequencies. There should be nchan-1 of them (0 and Nyquist are assumed)
if (print_debug) Serial.print(" cross_freq: ");
for (int i=0; i < (nchannel-1); i++) {
position = parseNextNumberFromString(text_buffer, position, foo_val);
cross_freq[i] = foo_val;
if (print_debug) { Serial.print(cross_freq[i]); Serial.print(", ");}
}
if (print_debug) Serial.println();
//read the tkgain values. There should be nchan of them
if (print_debug) Serial.print(" tkgain: ");
for (int i=0; i < nchannel; i++) {
position = parseNextNumberFromString(text_buffer, position, foo_val);
tkgain[i] = foo_val;
if (print_debug) { Serial.print(tkgain[i]); Serial.print(", ");}
}
if (print_debug) Serial.println();
//read the cr values. There should be nchan of them
if (print_debug) Serial.print(" cr: ");
for (int i=0; i < nchannel; i++) {
position = parseNextNumberFromString(text_buffer, position, foo_val);
cr[i] = foo_val;
if (print_debug) { Serial.print(cr[i]); Serial.print(", ");}
}
if (print_debug) Serial.println();
//read the tk values. There should be nchan of them
if (print_debug) Serial.print(" tk: ");
for (int i=0; i < nchannel; i++) {
position = parseNextNumberFromString(text_buffer, position, foo_val);
tk[i] = foo_val;
if (print_debug) { Serial.print(tk[i]); Serial.print(", ");}
}
if (print_debug) Serial.println();
//read the bolt values. There should be nchan of them
if (print_debug) Serial.print(" bolt: ");
for (int i=0; i < nchannel; i++) {
position = parseNextNumberFromString(text_buffer, position, foo_val);
bolt[i] = foo_val;
if (print_debug) { Serial.print(bolt[i]); Serial.print(", ");}
}
if (print_debug) Serial.println();
return 0;
}
void printToStream(Stream *s) {
s->print("CHA_DSL: attack (ms) = "); s->println(attack);
s->print(" : release (ms) = "); s->println(release);
s->print(" : maxdB (dB SPL) = "); s->println(maxdB);
s->print(" : ear (0 = left, 1 = right) "); s->println(ear);
s->print(" : nchannel = "); s->println(nchannel);
s->print(" : cross_freq (Hz) = ");
for (int i=0; i<nchannel-1;i++) { s->print(cross_freq[i]); s->print(", ");}; s->println();
s->print(" : tkgain = ");
for (int i=0; i<nchannel;i++) { s->print(tkgain[i]); s->print(", ");}; s->println();
s->print(" : cr = ");
for (int i=0; i<nchannel;i++) { s->print(cr[i]); s->print(", ");}; s->println();
s->print(" : tk = ");
for (int i=0; i<nchannel;i++) { s->print(tk[i]); s->print(", ");}; s->println();
s->print(" : bolt = ");
for (int i=0; i<nchannel;i++) { s->print(bolt[i]); s->print(", ");}; s->println();
}
} ; */
typedef struct {
float alfa; // attack constant (not time)
float beta; // release constant (not time
float fs; // sampling rate (Hz)
float maxdB; // maximum signal (dB SPL)
float tkgain; // compression-start gain
float tk; // compression-start kneepoint
float cr; // compression ratio
float bolt; // broadband output limiting threshold
} CHA_DVAR_t;
class AudioEffectCompWDRC_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName: CompWDRC
public:
AudioEffectCompWDRC_F32(void): AudioStream_F32(1,inputQueueArray) { //need to modify this for user to set sample rate
setSampleRate_Hz(AUDIO_SAMPLE_RATE);
setDefaultValues();
}
AudioEffectCompWDRC_F32(AudioSettings_F32 settings): AudioStream_F32(1,inputQueueArray) { //need to modify this for user to set sample rate
setSampleRate_Hz(settings.sample_rate_Hz);
setDefaultValues();
}
//here is the method called automatically by the audio library
void update(void) {
//receive the input audio data
audio_block_f32_t *block = AudioStream_F32::receiveReadOnly_f32();
if (!block) return;
//allocate memory for the output of our algorithm
audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
if (!out_block) return;
//do the algorithm
cha_agc_channel(block->data, out_block->data, block->length);
// transmit the block and release memory
AudioStream_F32::transmit(out_block); // send the FIR output
AudioStream_F32::release(out_block);
AudioStream_F32::release(block);
}
//here is the function that does all the work
void cha_agc_channel(float *input, float *output, int cs) {
//compress(input, output, cs, &prev_env,
// CHA_DVAR.alfa, CHA_DVAR.beta, CHA_DVAR.tkgain, CHA_DVAR.tk, CHA_DVAR.cr, CHA_DVAR.bolt, CHA_DVAR.maxdB);
compress(input, output, cs);
}
//void compress(float *x, float *y, int n, float *prev_env,
// float &alfa, float &beta, float &tkgn, float &tk, float &cr, float &bolt, float &mxdB)
void compress(float *x, float *y, int n)
//x, input, audio waveform data
//y, output, audio waveform data after compression
//n, input, number of samples in this audio block
{
// find smoothed envelope
audio_block_f32_t *envelope_block = AudioStream_F32::allocate_f32();
if (!envelope_block) return;
calcEnvelope.smooth_env(x, envelope_block->data, n);
//float *xpk = envelope_block->data; //get pointer to the array of (empty) data values
//calculate gain
audio_block_f32_t *gain_block = AudioStream_F32::allocate_f32();
if (!gain_block) return;
calcGain.calcGainFromEnvelope(envelope_block->data, gain_block->data, n);
//apply gain
arm_mult_f32(x, gain_block->data, y, n);
// release memory
AudioStream_F32::release(envelope_block);
AudioStream_F32::release(gain_block);
}
void setDefaultValues(void) {
//set default values...taken from CHAPRO, GHA_Demo.c from "amplify()"...ignores given sample rate
//assumes that the sample rate has already been set!!!!
CHA_WDRC gha = {1.0f, // attack time (ms)
50.0f, // release time (ms)
24000.0f, // fs, sampling rate (Hz), THIS IS IGNORED!
119.0f, // maxdB, maximum signal (dB SPL)
0.0f, // tkgain, compression-start gain
105.0f, // tk, compression-start kneepoint
10.0f, // cr, compression ratio
105.0f // bolt, broadband output limiting threshold
};
setParams_from_CHA_WDRC(&gha);
}
//set all of the parameters for the compressor using the CHA_WDRC structure
//assumes that the sample rate has already been set!!!
void setParams_from_CHA_WDRC(CHA_WDRC *gha) {
//configure the envelope calculator...assumes that the sample rate has already been set!
calcEnvelope.setAttackRelease_msec(gha->attack,gha->release); //these are in milliseconds
//configure the compressor
calcGain.setParams_from_CHA_WDRC(gha);
}
//set all of the user parameters for the compressor
//assumes that the sample rate has already been set!!!
void setParams(float attack_ms, float release_ms, float maxdB, float tkgain, float comp_ratio, float tk, float bolt) {
//configure the envelope calculator...assumes that the sample rate has already been set!
calcEnvelope.setAttackRelease_msec(attack_ms,release_ms);
//configure the WDRC gains
calcGain.setParams(maxdB, tkgain, comp_ratio, tk, bolt);
}
void setSampleRate_Hz(const float _fs_Hz) {
//pass this data on to its components that care
given_sample_rate_Hz = _fs_Hz;
calcEnvelope.setSampleRate_Hz(_fs_Hz);
}
float getCurrentLevel_dB(void) { return AudioCalcGainWDRC_F32::db2(calcEnvelope.getCurrentLevel()); } //this is 20*log10(abs(signal)) after the envelope smoothing
AudioCalcEnvelope_F32 calcEnvelope;
AudioCalcGainWDRC_F32 calcGain;
private:
audio_block_f32_t *inputQueueArray[1];
float given_sample_rate_Hz;
};
#endif

49
AudioEffectEmpty_F32.h
Unescape View File

@ -0,0 +1,49 @@
/*
* AudioEffectEmpty_F32
*
* Created: Chip Audette, Feb 2017
* Purpose: This module does nothing. It is an empty algorithm that can one
* can build from to make their own algorithm
*
* This processes a single stream fo audio data (ie, it is mono)
*
* MIT License. use at your own risk.
*/
#ifndef _AudioEffectEmpty_F32_h
#define _AudioEffectEmpty_F32_h
#include <arm_math.h> //ARM DSP extensions. for speed!
#include <AudioStream_F32.h>
class AudioEffectEmpty_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:empty
public:
//constructor
AudioEffectEmpty_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {};
//here's the method that does all the work
void update(void) {
//Serial.println("AudioEffectEmpty_F32: updating."); //for debugging.
audio_block_f32_t *block;
block = AudioStream_F32::receiveWritable_f32();
if (!block) return;
//add your processing here!
//transmit the block and be done
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
}
private:
audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
};
#endif

24
AudioEffectGain_F32.h
Unescape View File

@ -24,18 +24,18 @@ class AudioEffectGain_F32 : public AudioStream_F32
//here's the method that does all the work
void update(void) {
//Serial.println("AudioEffectGain_F32: updating."); //for debugging.
audio_block_f32_t *block;
block = AudioStream_F32::receiveWritable_f32();
if (!block) return;
//apply the gain
//for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) block->data[i] = gain * (block->data[i]); //non DSP way to do it
arm_scale_f32(block->data, gain, block->data, block->length); //use ARM DSP for speed!
//transmit the block and be done
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
//Serial.println("AudioEffectGain_F32: updating."); //for debugging.
audio_block_f32_t *block;
block = AudioStream_F32::receiveWritable_f32();
if (!block) return;
//apply the gain
//for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) block->data[i] = gain * (block->data[i]); //non DSP way to do it
arm_scale_f32(block->data, gain, block->data, block->length); //use ARM DSP for speed!
//transmit the block and be done
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
}
//methods to set parameters of this module

101
AudioFilterFIR_F32.h
Unescape View File

@ -22,44 +22,50 @@
class AudioFilterFIR_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
public:
AudioFilterFIR_F32(void): AudioStream_F32(1,inputQueueArray), coeff_p(FIR_F32_PASSTHRU) {
}
void begin(const float32_t *cp, int n_coeffs) {
coeff_p = cp;
// Initialize FIR instance (ARM DSP Math Library)
if (coeff_p && (coeff_p != FIR_F32_PASSTHRU) && n_coeffs <= FIR_MAX_COEFFS) {
arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p, &StateF32[0], AUDIO_BLOCK_SAMPLES);
//if (arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p, &StateF32[0], AUDIO_BLOCK_SAMPLES) != ARM_MATH_SUCCESS) {
// n_coeffs must be an even number, 4 or larger
//coeff_p = NULL;
//}
}
}
void end(void) {
coeff_p = NULL;
}
virtual void update(void);
void setBlockDC(void) {
//helper function that sets this up for a first-order HP filter at 20Hz
}
private:
audio_block_f32_t *inputQueueArray[1];
// pointer to current coefficients or NULL or FIR_PASSTHRU
const float32_t *coeff_p;
// ARM DSP Math library filter instance
arm_fir_instance_f32 fir_inst;
float32_t StateF32[AUDIO_BLOCK_SAMPLES + FIR_MAX_COEFFS];
public:
AudioFilterFIR_F32(void): AudioStream_F32(1,inputQueueArray),
coeff_p(FIR_F32_PASSTHRU), n_coeffs(1), configured_block_size(0) { }
//initialize the FIR filter by giving it the filter coefficients
void begin(const float32_t *cp, const int _n_coeffs) { begin(cp, _n_coeffs, AUDIO_BLOCK_SAMPLES); } //assume that the block size is the maximum
void begin(const float32_t *cp, const int _n_coeffs, const int block_size) { //or, you can provide it with the block size
coeff_p = cp;
n_coeffs = _n_coeffs;
// Initialize FIR instance (ARM DSP Math Library)
if (coeff_p && (coeff_p != FIR_F32_PASSTHRU) && n_coeffs <= FIR_MAX_COEFFS) {
arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p, &StateF32[0], block_size);
configured_block_size = block_size;
Serial.print("AudioFilterFIR_F32: FIR is initialized. N_FIR = "); Serial.print(n_coeffs);
Serial.print(", Block Size = "); Serial.println(block_size);
//} else {
// Serial.print("AudioFilterFIR_F32: *** ERROR ***: Cound not initialize. N_FIR = "); Serial.print(n_coeffs);
// Serial.print(", Block Size = "); Serial.println(block_size);
// coeff_p = NULL;
}
}
void end(void) { coeff_p = NULL; }
virtual void update(void);
//void setBlockDC(void) {} //helper function that sets this up for a first-order HP filter at 20Hz
private:
audio_block_f32_t *inputQueueArray[1];
// pointer to current coefficients or NULL or FIR_PASSTHRU
const float32_t *coeff_p;
int n_coeffs;
int configured_block_size;
// ARM DSP Math library filter instance
arm_fir_instance_f32 fir_inst;
float32_t StateF32[AUDIO_BLOCK_SAMPLES + FIR_MAX_COEFFS];
};
void AudioFilterFIR_F32::update(void)
{
audio_block_f32_t *block, *b_new;
audio_block_f32_t *block, *block_new;
block = AudioStream_F32::receiveReadOnly_f32();
if (!block) return;
@ -75,17 +81,30 @@ void AudioFilterFIR_F32::update(void)
// Just passthrough
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
//Serial.println("AudioFilterFIR_F32: update(): PASSTHRU.");
return;
}
// get a block for the FIR output
b_new = AudioStream_F32::allocate_f32();
if (b_new) {
arm_fir_f32(&fir_inst, (float32_t *)block->data, (float32_t *)b_new->data, block->length);
AudioStream_F32::transmit(b_new); // send the FIR output
AudioStream_F32::release(b_new);
}
AudioStream_F32::release(block);
// get a block for the FIR output
block_new = AudioStream_F32::allocate_f32();
if (block_new) {
//check to make sure our FIR instance has the right size
if (block->length != configured_block_size) {
//doesn't match. re-initialize
Serial.println("AudioFilterFIR_F32: block size doesn't match. Re-initializing FIR.");
begin(coeff_p, n_coeffs, block->length); //initialize with same coefficients, just a new block length
}
//apply the FIR
arm_fir_f32(&fir_inst, block->data, block_new->data, block->length);
block_new->length = block->length;
//transmit the data
AudioStream_F32::transmit(block_new); // send the FIR output
AudioStream_F32::release(block_new);
}
AudioStream_F32::release(block);
}
#endif

8
AudioFilterIIR_F32.h
Unescape View File

@ -1,8 +1,9 @@
/*
* AudioFilterFIR_F32
* AudioFilterIIR_F32
*
* Created: Chip Audette (OpenAudio) Feb 2017
* - Building from AudioFilterFIR from Teensy Audio Library (AudioFilterFIR credited to Pete (El Supremo))
*
* License: MIT License. Use at your own risk.
*
*/
@ -22,6 +23,7 @@
class AudioFilterIIR_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:IIR
public:
AudioFilterIIR_F32(void): AudioStream_F32(1,inputQueueArray), coeff_p(FIR_F32_PASSTHRU) {
}
@ -92,6 +94,8 @@ void AudioFilterIIR_F32::update(void)
// do IIR
arm_biquad_cascade_df1_f32(&iir_inst, block->data, block->data, block->length);
//transmit the data
AudioStream_F32::transmit(block); // send the IIR output
AudioStream_F32::release(block);
}

60
AudioMixer_F32.cpp
Unescape View File

@ -0,0 +1,60 @@
#include "AudioMixer_F32.h"
void AudioMixer4_F32::update(void) {
audio_block_f32_t *in, *out=NULL;
out = receiveWritable_f32(0);
if (!out) return;
arm_scale_f32(out->data, multiplier[0], out->data, out->length);
for (int channel=1; channel < 4; channel++) {
in = receiveReadOnly_f32(channel);
if (!in) {
continue;
}
audio_block_f32_t *tmp = allocate_f32();
arm_scale_f32(in->data, multiplier[channel], tmp->data, tmp->length);
arm_add_f32(out->data, tmp->data, out->data, tmp->length);
AudioStream_F32::release(tmp);
AudioStream_F32::release(in);
}
if (out) {
AudioStream_F32::transmit(out);
AudioStream_F32::release(out);
}
}
void AudioMixer8_F32::update(void) {
audio_block_f32_t *in, *out=NULL;
out = receiveWritable_f32(0); //try to get the first input channel
if (!out) return; //if it's not there, return immediately
arm_scale_f32(out->data, multiplier[0], out->data, out->length); //scale the first input channel
//load and process the rest of the channels
for (int channel=1; channel < 8; channel++) {
in = receiveReadOnly_f32(channel);
if (!in) {
continue;
}
audio_block_f32_t *tmp = allocate_f32();
arm_scale_f32(in->data, multiplier[channel], tmp->data, tmp->length);
arm_add_f32(out->data, tmp->data, out->data, tmp->length);
AudioStream_F32::release(tmp);
AudioStream_F32::release(in);
}
if (out) {
AudioStream_F32::transmit(out);
AudioStream_F32::release(out);
}
}

63
AudioMixer_F32.h
Unescape View File

@ -0,0 +1,63 @@
/*
* AudioMixer
*
* AudioMixer4
* Created: Patrick Radius, December 2016
* Purpose: Mix up to 4 audio channels with individual gain controls.
* Assumes floating-point data.
*
* This processes a single stream fo audio data (ie, it is mono)
*
* Extended to AudioMixer8
* By: Chip Audette, OpenAudio, Feb 2017
*
* MIT License. use at your own risk.
*/
#ifndef AUDIOMIXER_F32_H
#define AUDIOMIXER_F32_H
#include <arm_math.h>
#include <AudioStream_F32.h>
class AudioMixer4_F32 : public AudioStream_F32 {
//GUI: inputs:4, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:Mixer4
public:
AudioMixer4_F32() : AudioStream_F32(4, inputQueueArray) {
for (int i=0; i<4; i++) multiplier[i] = 1.0;
}
virtual void update(void);
void gain(unsigned int channel, float gain) {
if (channel >= 4 || channel < 0) return;
multiplier[channel] = gain;
}
private:
audio_block_f32_t *inputQueueArray[4];
float multiplier[4];
};
class AudioMixer8_F32 : public AudioStream_F32 {
//GUI: inputs:8, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:Mixer8
public:
AudioMixer8_F32() : AudioStream_F32(8, inputQueueArray) {
for (int i=0; i<8; i++) multiplier[i] = 1.0;
}
virtual void update(void);
void gain(unsigned int channel, float gain) {
if (channel >= 8 || channel < 0) return;
multiplier[channel] = gain;
}
private:
audio_block_f32_t *inputQueueArray[8];
float multiplier[8];
};
#endif

9
AudioStream_F32.cpp
Unescape View File

@ -30,6 +30,14 @@ void AudioStream_F32::initialize_f32_memory(audio_block_f32_t *data, unsigned in
__enable_irq();
} // end initialize_memory
void AudioStream_F32::initialize_f32_memory(audio_block_f32_t *data, unsigned int num, const AudioSettings_F32 &settings)
{
initialize_f32_memory(data,num);
for (unsigned int i=0; i < num; i++) {
data[i].fs_Hz = settings.sample_rate_Hz;
data[i].length = settings.audio_block_samples;
}
} // end initialize_memory
// Allocate 1 audio data block. If successful
// the caller is the only owner of this new block
@ -160,3 +168,4 @@ void AudioConnection_F32::connect(void) {
dst.active = true;
__enable_irq();
}

69
AudioStream_F32.h
Unescape View File

@ -10,31 +10,63 @@
* MIT License. use at your own risk.
*/
#ifndef _OpenAudio_ArduinoLibrary
#define _OpenAudio_ArduinoLibrary
#ifndef _AudioStream_F32_h
#define _AudioStream_F32_h
#include <arm_math.h> //ARM DSP extensions. for speed!
#include <Audio.h> //Teensy Audio Library
class AudioStream_F32;
class AudioConnection_F32;
class AudioSettings_F32;
class AudioSettings_F32 {
public:
AudioSettings_F32(float fs_Hz, int block_size) :
sample_rate_Hz(fs_Hz), audio_block_samples(block_size) {}
const float sample_rate_Hz;
const int audio_block_samples;
float cpu_load_percent(const int n) { //n is the number of cycles
#define CYCLE_COUNTER_APPROX_PERCENT(n) (((n) + (F_CPU / 32 / AUDIO_SAMPLE_RATE * AUDIO_BLOCK_SAMPLES / 100)) / (F_CPU / 16 / AUDIO_SAMPLE_RATE * AUDIO_BLOCK_SAMPLES / 100))
float foo1 = ((float)(F_CPU / 32))/sample_rate_Hz;
foo1 *= ((float)audio_block_samples);
foo1 /= 100.f;
foo1 += (float)n;
float foo2 = (float)(F_CPU / 16)/sample_rate_Hz;
foo2 *= ((float)audio_block_samples);
foo2 /= 100.f;
return foo1 / foo2;
//return (((n) + (F_CPU / 32 / sample_rate_Hz * audio_block_samples / 100)) / (F_CPU / 16 / sample_rate_Hz * audio_block_samples / 100));
}
float processorUsage(void) { return cpu_load_percent(AudioStream::cpu_cycles_total); };
float processorUsageMax(void) { return cpu_load_percent(AudioStream::cpu_cycles_total_max); }
void processorUsageMaxReset(void) { AudioStream::cpu_cycles_total_max = AudioStream::cpu_cycles_total; }
};
//create a new structure to hold audio as floating point values.
//modeled on the existing teensy audio block struct, which uses Int16
//https://github.com/PaulStoffregen/cores/blob/268848cdb0121f26b7ef6b82b4fb54abbe465427/teensy3/AudioStream.h
typedef struct audio_block_f32_struct {
unsigned char ref_count;
unsigned char memory_pool_index;
unsigned char reserved1;
unsigned char reserved2;
#if AUDIO_BLOCK_SAMPLES < 128
float32_t data[128]; //limit array size to be no smaller than 128. unstable otherwise?
#else
float32_t data[AUDIO_BLOCK_SAMPLES]; // AUDIO_BLOCK_SAMPLES is 128, from AudioStream.h
#endif
int length = AUDIO_BLOCK_SAMPLES; // AUDIO_BLOCK_SAMPLES is 128, from AudioStream.h
float fs_Hz = AUDIO_SAMPLE_RATE; // AUDIO_SAMPLE_RATE is 44117.64706 from AudioStream.h
} audio_block_f32_t;
class audio_block_f32_t {
public:
audio_block_f32_t(void) {};
audio_block_f32_t(const AudioSettings_F32 &settings) {
fs_Hz = settings.sample_rate_Hz;
length = settings.audio_block_samples;
};
unsigned char ref_count;
unsigned char memory_pool_index;
unsigned char reserved1;
unsigned char reserved2;
float32_t data[AUDIO_BLOCK_SAMPLES]; // AUDIO_BLOCK_SAMPLES is 128, from AudioStream.h
const int full_length = AUDIO_BLOCK_SAMPLES;
int length = AUDIO_BLOCK_SAMPLES; // AUDIO_BLOCK_SAMPLES is 128, from AudioStream.h
float fs_Hz = AUDIO_SAMPLE_RATE; // AUDIO_SAMPLE_RATE is 44117.64706 from AudioStream.h
};
class AudioConnection_F32
{
@ -64,6 +96,11 @@ class AudioConnection_F32
AudioStream_F32::initialize_f32_memory(data_f32, num); \
})
#define AudioMemory_F32_wSettings(num,settings) ({ \
static audio_block_f32_t data_f32[num]; \
AudioStream_F32::initialize_f32_memory(data_f32, num, settings); \
})
#define AudioMemoryUsage_F32() (AudioStream_F32::f32_memory_used)
#define AudioMemoryUsageMax_F32() (AudioStream_F32::f32_memory_used_max)
@ -80,6 +117,7 @@ class AudioStream_F32 : public AudioStream {
}
};
static void initialize_f32_memory(audio_block_f32_t *data, unsigned int num);
static void initialize_f32_memory(audio_block_f32_t *data, unsigned int num, const AudioSettings_F32 &settings);
//virtual void update(audio_block_f32_t *) = 0;
static uint8_t f32_memory_used;
static uint8_t f32_memory_used_max;