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

Libarary: big code update from my Tympan_Library

Browse Source
pull/5/head
Chip Audette 7 years ago
parent d46ab56ed7
commit 81492cc5ac
42 changed files with 2097 additions and 886 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. 2
      AudioCalcEnvelope_F32.h
  2. 101
      AudioCalcGainWDRC_F32.h
  3. 82
      AudioConfigFIRFilterBank_F32.cpp
  4. 78
      AudioConfigFIRFilterBank_F32.h
  5. 101
      AudioControlTester.cpp
  6. 538
      AudioControlTester.h
  7. 2
      AudioConvert_F32.h
  8. 187
      AudioEffectCompWDRC_F32.h
  9. 17
      AudioEffectCompressor_F32.h
  10. 1
      AudioEffectEmpty_F32.h
  11. 5
      AudioEffectGain_F32.h
  12. 41
      AudioFilterBiquad_F32.cpp
  13. 204
      AudioFilterBiquad_F32.h
  14. 55
      AudioFilterFIR_F32.cpp
  15. 53
      AudioFilterFIR_F32.h
  16. 30
      AudioMixer4_F32.cpp
  17. 38
      AudioMixer4_F32.h
  18. 14
      AudioMixer_F32.h
  19. 2
      AudioMultiply_F32.h
  20. 29
      AudioSettings_F32.cpp
  21. 18
      AudioSettings_F32.h
  22. 18
      AudioStream_F32.cpp
  23. 61
      AudioStream_F32.h
  24. 181
      BTNRH_WDRC_Types.h
  25. 19
      OpenAudio_ArduinoLibrary.h
  26. 146
      USB_Audio_F32.h
  27. 96
      control_tlv320aic3206.cpp
  28. 6
      control_tlv320aic3206.h
  29. 4
      examples/BasicCompressor_Float/BasicCompressor_Float.ino
  30. 2
      examples/BasicGain_Float/BasicGain_Float.ino
  31. 6
      examples/MixStereoToMono_Float/MixStereoToMono_Float.ino
  32. 4
      examples/MyAudioEffect_Float/MyAudioEffect_Float.ino
  33. 8
      examples/OscillatorWithPitchmod_Float/OscillatorWithPitchmod_Float.ino
  34. 2
      output_i2s_f32.cpp
  35. 2
      play_queue_f32.h
  36. 2
      record_queue_f32.h
  37. 11
      synth_pinknoise_f32.h
  38. 2
      synth_sine_f32.cpp
  39. 5
      synth_whitenoise_f32.h
  40. 393
      utility/BTNRH_rfft.cpp
  41. 19
      utility/BTNRH_rfft.h
  42. 384
      utility/rfft.c

2
AudioCalcEnvelope_F32.h
Unescape View File

@ -47,7 +47,7 @@ class AudioCalcEnvelope_F32 : public AudioStream_F32
audio_block_f32_t *out_block = AudioStream_F32::allocate_f32(); audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
if (!out_block) return; if (!out_block) return;
// //////////////////////add your processing here! // /////////// put the actual processing here
smooth_env(in_block->data, out_block->data, in_block->length); 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; out_block->length = in_block->length; out_block->fs_Hz = in_block->fs_Hz;

101
AudioCalcGainWDRC_F32.h
Unescape View File

@ -17,17 +17,8 @@
#include <arm_math.h> //ARM DSP extensions. for speed! #include <arm_math.h> //ARM DSP extensions. for speed!
#include <AudioStream_F32.h> #include <AudioStream_F32.h>
#include "BTNRH_WDRC_Types.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 class AudioCalcGainWDRC_F32 : public AudioStream_F32
@ -35,8 +26,9 @@ class AudioCalcGainWDRC_F32 : public AudioStream_F32
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node //GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:calc_WDRCGain //GUI: shortName:calc_WDRCGain
public: public:
//default constructor //constructors
AudioCalcGainWDRC_F32(void) : AudioStream_F32(1, inputQueueArray_f32) { setDefaultValues(); }; AudioCalcGainWDRC_F32(void) : AudioStream_F32(1, inputQueueArray_f32) { setDefaultValues(); };
AudioCalcGainWDRC_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) { setDefaultValues(); };
//here's the method that does all the work //here's the method that does all the work
void update(void) { void update(void) {
@ -49,7 +41,7 @@ class AudioCalcGainWDRC_F32 : public AudioStream_F32
audio_block_f32_t *out_block = AudioStream_F32::allocate_f32(); audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
if (!out_block) return; if (!out_block) return;
// //////////////////////add your processing here! // ////////////////////// do the processing here!
calcGainFromEnvelope(in_block->data, out_block->data, in_block->length); calcGainFromEnvelope(in_block->data, out_block->data, in_block->length);
out_block->length = in_block->length; out_block->fs_Hz = in_block->fs_Hz; out_block->length = in_block->length; out_block->fs_Hz = in_block->fs_Hz;
@ -69,7 +61,7 @@ class AudioCalcGainWDRC_F32 : public AudioStream_F32
audio_block_f32_t *env_dB_block = AudioStream_F32::allocate_f32(); audio_block_f32_t *env_dB_block = AudioStream_F32::allocate_f32();
if (!env_dB_block) return; if (!env_dB_block) return;
//convert to dB //convert to dB and calibrate (via maxdB)
for (int k=0; k < n; k++) env_dB_block->data[k] = maxdB + db2(env[k]); //maxdb in the private section 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 // apply wide-dynamic range compression
@ -80,48 +72,60 @@ class AudioCalcGainWDRC_F32 : public AudioStream_F32
//original call to WDRC_circuit //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 *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) //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 //modified to output just the gain instead of the fully processed signal
void WDRC_circuit_gain(float *env_dB, float *gain_out, const int n, 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) { const float tkgn, const float tk, const float cr, const float bolt)
//tkgn = gain (dB?) at start of compression (ie, gain for linear behavior?)
//tk = compression start kneepoint (pre-compression, dB SPL?)
//cr = compression ratio
//bolt = broadband output limiting threshold (post-compression, dB SPL?)
{
//tkgain = 30; tk = 50; bolt = 100; cr = 3;
float gdb, tkgo, pblt; float gdb, tkgo, pblt;
int k; int k;
float *pdb = env_dB; //just rename it to keep the code below unchanged float *pdb = env_dB; //just rename it to keep the code below unchanged (input SPL dB)
float tk_tmp = tk; float tk_tmp = tk; //temporary, threshold for start of compression (input SPL dB)
if ((tk_tmp + tkgn) > bolt) { if ((tk_tmp + tkgn) > bolt) { //after gain, would the compression threshold be above the output-limitting threshold ("bolt")
tk_tmp = bolt - tkgn; tk_tmp = bolt - tkgn; //if so, lower the compression threshold to be the pre-gain value resulting in "bolt"
} }
tkgo = tkgn + tk_tmp * (1.0f - 1.0f / cr);
pblt = cr * (bolt - tkgo); tkgo = tkgn + tk_tmp * (1.0f - 1.0f / cr); //intermediate calc
const float cr_const = ((1.0f / cr) - 1.0f); pblt = cr * (bolt - tkgo); //calc input level (dB) where we need to start limiting, no just compression
for (k = 0; k < n; k++) { const float cr_const = ((1.0f / cr) - 1.0f); //pre-calc a constant that we'll need later
if ((pdb[k] < tk_tmp) && (cr >= 1.0f)) {
gdb = tkgn; for (k = 0; k < n; k++) { //loop over each sample
} else if (pdb[k] > pblt) { if ((pdb[k] < tk_tmp) && (cr >= 1.0f)) { //if below threshold and we're compressing
gdb = bolt + ((pdb[k] - pblt) / 10.0f) - pdb[k]; gdb = tkgn; //we're in the linear region. Apply linear gain.
} else if (pdb[k] > pblt) { //we're beyond the compression region into the limitting region
gdb = bolt + ((pdb[k] - pblt) / 10.0f) - pdb[k]; //10:1 limiting!
} else { } else {
gdb = cr_const * pdb[k] + tkgo; gdb = cr_const * pdb[k] + tkgo;
} }
gain_out[k] = undb2(gdb); gain_out[k] = undb2(gdb);
//y[k] = x[k] * undb2(gdb); //apply the gain //y[k] = x[k] * undb2(gdb); //apply the gain
}
}
last_gain = gain_out[n-1]; //hold this value, in case the user asks for it later (not needed for the algorithm)
} }
void setDefaultValues(void) { void setDefaultValues(void) { //set as limiter
CHA_WDRC gha = {1.0f, // attack time (ms), IGNORED HERE BTNRH_WDRC::CHA_WDRC gha = {
50.0f, // release time (ms), IGNORED HERE 5.0f, // attack time (ms)
24000.0f, // fs, sampling rate (Hz), IGNORED HERE 50.0f, // release time (ms)
119.0f, // maxdB, maximum signal (dB SPL) 24000.0f, // fs, sampling rate (Hz), THIS IS IGNORED!
0.0f, // tkgain, compression-start gain 115.0f, // maxdB, maximum signal (dB SPL)...assumed SPL for full-scale input signal
105.0f, // tk, compression-start kneepoint 0.0f, // tkgain, compression-start gain (dB)
10.0f, // cr, compression ratio 55.0f, // tk, compression-start kneepoint (dB SPL)
105.0f // bolt, broadband output limiting threshold 1.0f, // cr, compression ratio (set to 1.0 to defeat)
100.0f // bolt, broadband output limiting threshold (ie, the limiter. SPL. 10:1 comp ratio)
}; };
//setParams(gha.maxdB, gha.tkgain, gha.cr, gha.tk, gha.bolt); //also sets calcEnvelope //setParams(gha.maxdB, gha.tkgain, gha.cr, gha.tk, gha.bolt); //also sets calcEnvelope
setParams_from_CHA_WDRC(&gha); setParams_from_CHA_WDRC(&gha);
} }
void setParams_from_CHA_WDRC(CHA_WDRC *gha) { void setParams_from_CHA_WDRC(BTNRH_WDRC::CHA_WDRC *gha) {
setParams(gha->maxdB, gha->tkgain, gha->cr, gha->tk, gha->bolt); //also sets calcEnvelope 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) { void setParams(float _maxdB, float _tkgain, float _cr, float _tk, float _bolt) {
@ -132,6 +136,26 @@ class AudioCalcGainWDRC_F32 : public AudioStream_F32
bolt = _bolt; bolt = _bolt;
} }
void setKneeLimiter_dBSPL(float _bolt) { bolt = _bolt; }
void setKneeLimiter_dBFS(float _bolt_dBFS) { //convert to dB SPL
float bolt_dBSPL = maxdB + _bolt_dBFS;
setKneeLimiter_dBSPL(bolt_dBSPL);
}
void setGain_dB(float _gain_dB) { tkgn = _gain_dB; } //gain at start of compression
void setKneeCompressor_dBSPL(float _tk) { tk = _tk; }
void setKneeCompressor_dBFS(float _tk_dBFS) { // convert to dB SPL
float tk_dBSPL = maxdB + _tk_dBFS;
setKneeCompressor_dBSPL(tk_dBSPL);
}
void setCompRatio(float _cr) { cr = _cr; };
void setMaxdB(float _maxdB) { maxdB = _maxdB; }
float getGain_dB(void) { return tkgn; } //returns the linear gain of the system
float getCurrentGain(void) { return last_gain; }
float getCurrentGain_dB(void) { return db2(getCurrentGain()); }
//dB functions. Feed it the envelope amplitude (not squared) and it computes 20*log10(x) or it does 10.^(x/20)
static float undb2(const float &x) { return expf(0.11512925464970228420089957273422f*x); } //faster: exp(log(10.0f)*x/20); this is exact 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 static float db2(const float &x) { return 6.020599913279623f*log2f_approx(x); } //faster: 20*log2_approx(x)/log2(10); this is approximate
@ -168,6 +192,7 @@ class AudioCalcGainWDRC_F32 : public AudioStream_F32
private: private:
audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
float maxdB, tkgn, tk, cr, bolt; float maxdB, tkgn, tk, cr, bolt;
float last_gain = 1.0; //what was the last gain value computed for the signal
}; };
#endif #endif

82
AudioConfigFIRFilterBank_F32.cpp
Unescape View File

@ -0,0 +1,82 @@
/*
* fir_filterbank.cpp
*
* 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.
*
*/
#include "AudioConfigFIRFilterBank_F32.h"
#include "utility/BTNRH_rfft.h"
void AudioConfigFIRFilterBank_F32::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("AudioConfigFIRFilterBank: fir_filterbank: nw_orig = "); Serial.print(nw_orig);
//Serial.print(", nw = "); Serial.println(nw);
nt = nw * 2; //we're going to do an fft that's twice as long (zero padded)
nf = nw + 1; //number of bins to nyquist in the zero-padded FFT. Also nf = nt/2+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 if (wt==1) {
w = (1 - a + cos(p) + a * cos(2 * p)) / 2; // Blackman
} else {
//win = (1 - cos(2*pi*[1:N]/(N+1)))/2; //WEA's matlab call, indexing starts from 1, not zero
w = (1.0 - cosf(2.0*M_PI*((float)(j))/((float)(nw_orig-1))))/2.0;
}
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)); //original
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
BTNRH_FFT::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
BTNRH_FFT::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);
}

78
AudioConfigFIRFilterBank_F32.h
Unescape View File

@ -12,7 +12,8 @@
#ifndef AudioConfigFIRFilterBank_F32_h #ifndef AudioConfigFIRFilterBank_F32_h
#define AudioConfigFIRFilterBank_F32_h #define AudioConfigFIRFilterBank_F32_h
#include "utility/rfft.c" //include <Tympan_Library.h>
#include <OpenAudio_ArduinoLibrary.h>
#define fmove(x,y,n) memmove(x,y,(n)*sizeof(float)) #define fmove(x,y,n) memmove(x,y,(n)*sizeof(float))
#define fcopy(x,y,n) memcpy(x,y,(n)*sizeof(float)) #define fcopy(x,y,n) memcpy(x,y,(n)*sizeof(float))
@ -22,9 +23,10 @@ class AudioConfigFIRFilterBank_F32 {
//GUI: inputs:0, outputs:0 //this line used for automatic generation of GUI node //GUI: inputs:0, outputs:0 //this line used for automatic generation of GUI node
//GUI: shortName:config_FIRbank //GUI: shortName:config_FIRbank
public: public:
AudioConfigFIRFilterBank_F32(void) { AudioConfigFIRFilterBank_F32(void) {}
} AudioConfigFIRFilterBank_F32(const AudioSettings_F32 &settings) {}
AudioConfigFIRFilterBank_F32(const int n_chan, const int n_fir, const float sample_rate_Hz, float *corner_freq, float *filter_coeff) { 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(n_chan, n_fir, sample_rate_Hz, corner_freq, filter_coeff);
} }
@ -51,7 +53,7 @@ class AudioConfigFIRFilterBank_F32 {
flag__free_cf = 1; flag__free_cf = 1;
computeLogSpacedCornerFreqs(n_chan, sample_rate_Hz, cf); computeLogSpacedCornerFreqs(n_chan, sample_rate_Hz, cf);
} }
const int window_type = 0; //0 = Hamming const int window_type = 0; //0 = Hamming, 1=Blackmann, 2 = Hanning
fir_filterbank(filter_coeff, cf, n_chan, n_fir, window_type, sample_rate_Hz); fir_filterbank(filter_coeff, cf, n_chan, n_fir, window_type, sample_rate_Hz);
if (flag__free_cf) free(cf); if (flag__free_cf) free(cf);
} }
@ -84,71 +86,7 @@ class AudioConfigFIRFilterBank_F32 {
return n; return n;
} }
void fir_filterbank(float *bb, float *cf, const int nc, const int nw_orig, const int wt, const float sr) 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 #endif

101
AudioControlTester.cpp
Unescape View File

@ -0,0 +1,101 @@
#include "AudioControlTester.h"
void AudioTestSignalGenerator_F32::update(void) {
//receive the input audio data
audio_block_f32_t *in_block = AudioStream_F32::receiveReadOnly_f32();
if (!in_block) return;
//if we're not testing, just transmit the block
if (!is_testing) {
AudioStream_F32::transmit(in_block); // send the FIR output
AudioStream_F32::release(in_block);
return;
}
//if we are testing, we're going to substitute a new signal,
//so we can release the block for the original signal
AudioStream_F32::release(in_block);
//allocate memory for block that we'll send
audio_block_f32_t *out_block = allocate_f32();
if (!out_block) return;
//now generate the new siginal
sine_gen.begin(); record_queue.begin(); //activate
sine_gen.update(); sine_gen.end();
gain_alg.update();
record_queue.update(); record_queue.end();
audio_block_f32_t *queue_block = record_queue.getAudioBlock();
//copy to out_block (why can't I just send the queue_block? I tried. It doesn't seem to work. try it again later!
for (int i=0; i < queue_block->length; i++) out_block->data[i] = queue_block->data[i];
record_queue.freeAudioBlock();
//send the data
AudioStream_F32::transmit(out_block); // send the FIR output
AudioStream_F32::release(out_block);
}
void AudioTestSignalMeasurement_F32::update(void) {
//if we're not testing, just return
if (!is_testing) {
return;
}
//receive the input audio data...the baseline and the test
audio_block_f32_t *in_block_baseline = AudioStream_F32::receiveReadOnly_f32(0);
if (!in_block_baseline) return;
audio_block_f32_t *in_block_test = AudioStream_F32::receiveReadOnly_f32(1);
if (!in_block_test) {
AudioStream_F32::release(in_block_baseline);
return;
}
//compute the rms of both signals
float baseline_rms = computeRMS(in_block_baseline->data, in_block_baseline->length);
float test_rms = computeRMS(in_block_test->data, in_block_test->length);
//Release memory
AudioStream_F32::release(in_block_baseline);
AudioStream_F32::release(in_block_test);
//notify controller
if (testController != NULL) testController->transferRMSValues(baseline_rms, test_rms);
}
void AudioTestSignalMeasurementMulti_F32::update(void) {
//if we're not testing, just return
if (!is_testing) {
return;
}
//receive the input audio data...the baseline and the test
audio_block_f32_t *in_block_baseline = AudioStream_F32::receiveReadOnly_f32(0);
if (in_block_baseline==NULL) return;
float baseline_rms = computeRMS(in_block_baseline->data, in_block_baseline->length);
AudioStream_F32::release(in_block_baseline);
//loop over each of the test data connections
float test_rms[num_test_values];
int n_with_data = 0;
for (int Ichan=0; Ichan < num_test_values; Ichan++) {
audio_block_f32_t *in_block_test = AudioStream_F32::receiveReadOnly_f32(1+Ichan);
if (in_block_test==NULL) {
//no data
test_rms[Ichan]=0.0f;
} else {
//process data
n_with_data = Ichan+1;
test_rms[Ichan]=computeRMS(in_block_test->data, in_block_test->length);
AudioStream_F32::release(in_block_test);
}
}
//notify controller
if (testController != NULL) testController->transferRMSValues(baseline_rms, test_rms, n_with_data);
}

538
AudioControlTester.h
Unescape View File

@ -0,0 +1,538 @@
#ifndef _AudioControlTester_h
#define _AudioControlTester_h
//include <Tympan_Library.h>
#include <OpenAudio_ArduinoLibrary.h>
#define max_steps 64
#define max_num_chan 16 //max number of test signal inputs to the AudioTestSignalMeasurementMulti_F32
//prototypes
class AudioTestSignalGenerator_F32;
class AudioTestSignalMeasurementInterface_F32;
class AudioTestSignalMeasurement_F32;
class AudioTestSignalMeasurementMulti_F32;
class AudioControlSignalTesterInterface_F32;
class AudioControlSignalTester_F32;
class AudioControlTestAmpSweep_F32;
class AudioControlTestFreqSweep_F32;
// class definitions
class AudioTestSignalGenerator_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName: testSignGen
public:
AudioTestSignalGenerator_F32(void): AudioStream_F32(1,inputQueueArray) {
setSampleRate_Hz(AUDIO_SAMPLE_RATE);
setDefaultValues();
makeConnections();
}
AudioTestSignalGenerator_F32(const AudioSettings_F32 &settings): AudioStream_F32(1,inputQueueArray) {
setAudioSettings(settings);
setDefaultValues();
makeConnections();
}
~AudioTestSignalGenerator_F32(void) {
if (patchCord1 != NULL) delete patchCord1;
}
void setAudioSettings(const AudioSettings_F32 &settings) {
setSampleRate_Hz(settings.sample_rate_Hz);
}
void setSampleRate_Hz(const float _fs_Hz) {
//pass this data on to its components that care
sine_gen.setSampleRate_Hz(_fs_Hz);
}
void makeConnections(void) {
patchCord1 = new AudioConnection_F32(sine_gen, 0, gain_alg, 0);
patchCord2 = new AudioConnection_F32(gain_alg, 0, record_queue, 0);
}
virtual void update(void);
void begin(void) {
is_testing = true;
//if (Serial) Serial.println("AudioTestSignalGenerator_F32: begin(): ...");
}
void end(void) { is_testing = false; }
AudioSynthWaveformSine_F32 sine_gen;
AudioEffectGain_F32 gain_alg;
AudioRecordQueue_F32 record_queue;
AudioConnection_F32 *patchCord1;
AudioConnection_F32 *patchCord2;
void amplitude(float val) {
sine_gen.amplitude(1.0);
gain_alg.setGain(val);
}
void frequency(float val) {
sine_gen.frequency(val);
}
virtual void setSignalAmplitude_dBFS(float val_dBFS) {
amplitude(sqrtf(2.0)*sqrtf(powf(10.0f,0.1*val_dBFS)));
};
virtual void setSignalFrequency_Hz(float val_Hz) {
frequency(val_Hz);
}
private:
bool is_testing = false;
audio_block_f32_t *inputQueueArray[1];
void setDefaultValues(void) {
sine_gen.end(); //disable it for now
record_queue.end(); //disable it for now;
is_testing = false;
frequency(1000.f);
amplitude(0.0f);
}
};
// //////////////////////////////////////////////////////////////////////////
class AudioTestSignalMeasurementInterface_F32 {
public:
AudioTestSignalMeasurementInterface_F32 (void) {};
void setAudioSettings(const AudioSettings_F32 &settings) {
setSampleRate_Hz(settings.sample_rate_Hz);
}
void setSampleRate_Hz(const float _fs_Hz) {
//pass this data on to its components that care. None care right now.
}
virtual void update(void);
virtual float computeRMS(float data[], int n) {
float rms_value;
arm_rms_f32 (data, n, &rms_value);
return rms_value;
}
virtual void begin(AudioControlSignalTester_F32 *p_controller) {
//if (Serial) Serial.println("AudioTestSignalMeasurement_F32: begin(): ...");
testController = p_controller;
is_testing = true;
}
virtual void end(void) {
//if (Serial) Serial.println("AudioTestSignalMeasurement_F32: end(): ...");
testController = NULL;
is_testing = false;
}
protected:
bool is_testing = false;
//audio_block_f32_t *inputQueueArray[2];
AudioControlSignalTester_F32 *testController = NULL;
virtual void setDefaultValues(void) {
is_testing = false;
}
};
class AudioTestSignalMeasurement_F32 : public AudioStream_F32, public AudioTestSignalMeasurementInterface_F32
{
//GUI: inputs:2, outputs:0 //this line used for automatic generation of GUI node
//GUI: shortName: testSigMeas
public:
AudioTestSignalMeasurement_F32(void): AudioStream_F32(2,inputQueueArray) {
setSampleRate_Hz(AUDIO_SAMPLE_RATE);
setDefaultValues();
}
AudioTestSignalMeasurement_F32(const AudioSettings_F32 &settings): AudioStream_F32(2,inputQueueArray) {
setAudioSettings(settings);
setDefaultValues();
}
void update(void);
private:
audio_block_f32_t *inputQueueArray[2];
};
class AudioTestSignalMeasurementMulti_F32 : public AudioStream_F32, public AudioTestSignalMeasurementInterface_F32
{
//GUI: inputs:10, outputs:0 //this line used for automatic generation of GUI node
//GUI: shortName: testSigMeas
public:
AudioTestSignalMeasurementMulti_F32(void): AudioStream_F32(max_num_chan+1,inputQueueArray) {
setSampleRate_Hz(AUDIO_SAMPLE_RATE);
setDefaultValues();
}
AudioTestSignalMeasurementMulti_F32(const AudioSettings_F32 &settings): AudioStream_F32(max_num_chan+1,inputQueueArray) {
setAudioSettings(settings);
setDefaultValues();
}
void update(void);
private:
//int num_input_connections = max_num_chan+1;
int num_test_values = max_num_chan;
audio_block_f32_t *inputQueueArray[max_num_chan+1];
};
// ///////////////////////////////////////////////////////////////////////////////////
class AudioControlSignalTesterInterface_F32 {
public:
AudioControlSignalTesterInterface_F32(void) {};
//virtual void setAudioBlockSamples(void) = 0;
//virtual void setSampleRate_hz(void) = 0;
virtual void begin(void) = 0;
virtual void end(void) = 0;
virtual void setStepPattern(float, float, float) = 0;
virtual void transferRMSValues(float, float) = 0;
virtual void transferRMSValues(float, float *, int) = 0;
virtual bool available(void) = 0;
};
class AudioControlSignalTester_F32 : public AudioControlSignalTesterInterface_F32
{
//GUI: inputs:0, outputs:0 //this line used for automatic generation of GUI node
//GUI: shortName: sigTest(Abstract)
public:
AudioControlSignalTester_F32(AudioSettings_F32 &settings, AudioTestSignalGenerator_F32 &_sig_gen, AudioTestSignalMeasurementInterface_F32 &_sig_meas)
: AudioControlSignalTesterInterface_F32(), sig_gen(_sig_gen), sig_meas(_sig_meas) {
setAudioBlockSamples(settings.audio_block_samples);
setSampleRate_Hz(settings.sample_rate_Hz);
resetState();
}
virtual void begin(void) {
Serial.println("AudioControlSignalTester_F32: begin(): ...");
recomputeTargetCountsPerStep(); //not needed, just to print some debugging messages
//activate the instrumentation
sig_gen.begin();
sig_meas.begin(this);
//start the test
resetState();
gotoNextStep();
}
//use this to cancel the test
virtual void end(void) {
finishTest();
}
void setAudioSettings(AudioSettings_F32 audio_settings) {
setAudioBlockSamples(audio_settings.audio_block_samples);
setSampleRate_Hz(audio_settings.sample_rate_Hz);
}
void setAudioBlockSamples(int block_samples) {
audio_block_samples = block_samples;
recomputeTargetCountsPerStep();
}
void setSampleRate_Hz(float fs_Hz) {
sample_rate_Hz = fs_Hz;
recomputeTargetCountsPerStep();
}
//define how long (seconds) to spend at each step of the test
void setTargetDurPerStep_sec(float sec) {
if (sec > 0.001) {
target_dur_per_step_sec = sec;
recomputeTargetCountsPerStep();
} else {
Serial.print(F("AudioControlSignalTester_F32: setTargetDurPerStep_sec: given duration too short: "));
Serial.print(target_dur_per_step_sec);
Serial.print(F(". Ignoring..."));
return;
}
}
virtual void setStepPattern(float _start_val, float _end_val, float _step_val) {
start_val = _start_val; end_val = _end_val; step_val = _step_val;
recomputeTargetNumberOfSteps();
}
virtual void transferRMSValues(float baseline_rms, float test_rms) {
transferRMSValues(baseline_rms, &test_rms, 1);
}
virtual void transferRMSValues(float baseline_rms, float *test_rms, int num_chan) {
if (counter_ignore > 0) {
//ignore this reading
counter_ignore--;
return;
}
given_num_chan = num_chan;
if (given_num_chan > max_num_chan) {
Serial.println(F("AudioControlSignalTester_F32: transferRMSValues: *** ERROR ***"));
Serial.print(F(" : num_chan (")); Serial.print(num_chan); Serial.print(")");
Serial.print(F(" is bigger max_num_chan (")); Serial.println(max_num_chan);
Serial.println(F(" : Skipping..."));
return;
}
//add this number
sum_sig_pow_baseline[counter_step] += (baseline_rms*baseline_rms);
for (int Ichan=0; Ichan < num_chan; Ichan++) {
sum_sig_pow_test[counter_step][Ichan] += (test_rms[Ichan]*test_rms[Ichan]);
}
freq_at_each_step_Hz[counter_step] = signal_frequency_Hz;
counter_sum[counter_step]++;
//have all the channels checked in?
if (counter_sum[counter_step] >= target_counts_per_step) {
gotoNextStep();
}
}
virtual void setSignalFrequency_Hz(float freq_Hz) {
signal_frequency_Hz = freq_Hz;
sig_gen.setSignalFrequency_Hz(signal_frequency_Hz);
}
virtual void setSignalAmplitude_dBFS(float amp_dBFS) {
signal_amplitude_dBFS = amp_dBFS;
sig_gen.setSignalAmplitude_dBFS(amp_dBFS);
}
virtual void printTableOfResults(Stream *s) {
float ave1_dBFS, ave2_dBFS, ave3_dBFS, gain_dB, total_pow, total_wav, foo_pow;
s->println(" : Freq (Hz), Input (dBFS), Per-Chan Output (dBFS), Total Gain (inc) (dB), Total Gain (coh) (dB)");
//s->print(" : given_num_chan = ");s->println(given_num_chan);
for (int i=0; i < target_n_steps; i++) {
ave1_dBFS = 10.f*log10f(sum_sig_pow_baseline[i]/counter_sum[i]);
s->print(" "); s->print(freq_at_each_step_Hz[i],0);
s->print(", "); s->print(ave1_dBFS,1);
total_pow = 0.0f;
total_wav = 0.0f;
for (int Ichan=0; Ichan < given_num_chan; Ichan++) {
if (Ichan==0) {
s->print(", ");
} else {
s->print(", ");
}
foo_pow = sum_sig_pow_test[i][Ichan]/counter_sum[i];
ave2_dBFS = 10.f*log10f(foo_pow);
s->print(ave2_dBFS,1);
total_pow += foo_pow; //sum as if it's noise being recombined incoherently
total_wav += sqrtf(foo_pow); //sum as it it's a in-phase tone being combined coherently
}
ave2_dBFS = 10.f*log10f(total_pow);
gain_dB = ave2_dBFS - ave1_dBFS;
s->print(", "); s->print(gain_dB,2);
ave3_dBFS = 20.f*log10f(total_wav);
gain_dB = ave3_dBFS - ave1_dBFS;
s->print(", "); s->println(gain_dB,2);
}
}
bool isDataAvailable = false;
bool available(void) { return isDataAvailable; }
protected:
AudioTestSignalGenerator_F32 &sig_gen;
AudioTestSignalMeasurementInterface_F32 &sig_meas;
float signal_frequency_Hz = 1000.f;
float signal_amplitude_dBFS = -50.0f;
int counter_ignore = 0;
//bool is_testing = 0;
int audio_block_samples = AUDIO_BLOCK_SAMPLES;
float sample_rate_Hz = AUDIO_SAMPLE_RATE_EXACT;
float target_dur_per_step_sec = 0.2;
int target_counts_per_step = 1;
//const int max_steps = 64;
float start_val = 0, end_val = 1.f, step_val = 1.f;
int target_n_steps = 1;
int given_num_chan = max_num_chan;
float sum_sig_pow_baseline[max_steps];
float sum_sig_pow_test[max_steps][max_num_chan];
float freq_at_each_step_Hz[max_steps];
int counter_sum[max_steps], counter_step=-1;
int recomputeTargetCountsPerStep(void) {
target_counts_per_step = max(1,(int)((target_dur_per_step_sec * sample_rate_Hz / ((float)audio_block_samples))+0.5)); //round
// if (Serial) {
// Serial.println("AudioControlSignalTester_F32: recomputeTargetCountsPerStep: ");
// Serial.print(" : target_dur_per_step_sec = "); Serial.println(target_dur_per_step_sec);
// Serial.print(" : sample_rate_Hz = "); Serial.println(sample_rate_Hz);
// Serial.print(" : audio_block_samples = "); Serial.println(audio_block_samples);
// Serial.print(" : target_counts_per_step = "); Serial.println(target_counts_per_step);
// }
return target_counts_per_step;
}
virtual int recomputeTargetNumberOfSteps(void) {
return target_n_steps = (int)((end_val - start_val)/step_val + 0.5)+1; //round
}
virtual void resetState(void) {
isDataAvailable = false;
for (int i=0; i<max_steps; i++) {
sum_sig_pow_baseline[i]=0.0f;
for (int Ichan=0; Ichan<max_num_chan; Ichan++) sum_sig_pow_test[i][Ichan]=0.0f;
counter_sum[i] = 0;
}
counter_step = -1;
}
virtual void gotoNextStep(void) {
counter_step++;
//Serial.print("AudioControlSignalTester_F32: gotoNextStep: starting step ");
//Serial.println(counter_step);
if (counter_step >= target_n_steps) {
finishTest();
return;
} else {
counter_ignore = 10; //ignore first 10 packets
counter_sum[counter_step]=0;
sum_sig_pow_baseline[counter_step]=0.0f;
for (int Ichan=0; Ichan < max_num_chan; Ichan++) sum_sig_pow_test[counter_step][Ichan]=0.0f;
updateSignalGenerator();
freq_at_each_step_Hz[counter_step]=0.0f;
}
}
virtual void updateSignalGenerator(void)
{
//if (Serial) Serial.println("AudioControlSignalTester_F32: updateSignalGenerator(): did the child version get called?");
} //override this is a child class!
virtual void finishTest(void) {
//Serial.println("AudioControlSignalTester_F32: finishTest()...");
//disable the test instrumentation
sig_gen.end();
sig_meas.end();
//let listeners know that data is available
isDataAvailable = true;
}
};
// //////////////////////////////////////////////////////////////////////////
class AudioControlTestAmpSweep_F32 : public AudioControlSignalTester_F32
{
//GUI: inputs:0, outputs:0 //this line used for automatic generation of GUI node
//GUI: shortName: ampSweepTester
public:
AudioControlTestAmpSweep_F32(AudioSettings_F32 &settings, AudioTestSignalGenerator_F32 &_sig_gen, AudioTestSignalMeasurementInterface_F32 &_sig_meas)
: AudioControlSignalTester_F32(settings, _sig_gen,_sig_meas)
{
float start_amp_dB = -100.0f, end_amp_dB = 0.0f, step_amp_dB = 2.5f;
setStepPattern(start_amp_dB, end_amp_dB, step_amp_dB);
setTargetDurPerStep_sec(0.25);
resetState();
}
void begin(void) {
//activate the instrumentation
sig_gen.begin();
sig_meas.begin(this);
//start the test
resetState();
gotoNextStep();
}
//use this to cancel the test
//void end(void) {
// finishTest();
//}
void printTableOfResults(Stream *s) {
s->println("AudioControlTestAmpSweep_F32: Start Table of Results...");
AudioControlSignalTester_F32::printTableOfResults(s);
s->println("AudioControlTestAmpSweep_F32: End Table of Results...");
}
protected:
virtual void updateSignalGenerator(void) {
float new_amp_dB = start_val + ((float)counter_step)*step_val; //start_val and step_val are in parent class
Serial.print("AudioControlTestAmpSweep_F32: updateSignalGenerator(): setting amplitude to (dBFS) ");
Serial.println(new_amp_dB);
setSignalAmplitude_dBFS(new_amp_dB);
}
void finishTest(void) {
//disable the test instrumentation
setSignalAmplitude_dBFS(-1000.0f); //some very quiet value
//do all of the common actions
AudioControlSignalTester_F32::finishTest();
//print results
printTableOfResults(&Serial);
}
//void resetState(void) {
// AudioControlSignalTester_F32::resetState();
//}
};
// //////////////////////////////////////////////////////////////////////////
class AudioControlTestFreqSweep_F32 : public AudioControlSignalTester_F32
{
//GUI: inputs:0, outputs:0 //this line used for automatic generation of GUI node
//GUI: shortName: freqSweepTester
public:
AudioControlTestFreqSweep_F32(AudioSettings_F32 &settings, AudioTestSignalGenerator_F32 &_sig_gen, AudioTestSignalMeasurementInterface_F32 &_sig_meas)
: AudioControlSignalTester_F32(settings, _sig_gen,_sig_meas)
{
float start_freq_Hz = 125.f, end_freq_Hz = 16000.f, step_freq_octave = sqrtf(2.0);
setStepPattern(start_freq_Hz, end_freq_Hz, step_freq_octave);
setTargetDurPerStep_sec(0.25);
resetState();
}
void begin(void) {
//activate the instrumentation
sig_gen.begin();
sig_meas.begin(this);
//start the test
resetState();
recomputeTargetNumberOfSteps();
gotoNextStep();
}
//use this to cancel the test
//void end(void) {
// finishTest();
//}
void printTableOfResults(Stream *s) {
s->println("AudioControlTestFreqSweep_F32: Start Table of Results...");
AudioControlSignalTester_F32::printTableOfResults(s);
s->println("AudioControlTestFreqSweep_F32: End Table of Results...");
}
protected:
float signal_frequency_Hz = 1000.f;
float signal_amplitude_dBFS = -50.0f;
virtual int recomputeTargetNumberOfSteps(void) {
return target_n_steps = (int)(log10f(end_val/start_val)/log10f(step_val)+0.5) + 1; //round
}
virtual void updateSignalGenerator(void) {
//logarithmically step the frequency
float new_freq_Hz = start_val * powf(step_val,counter_step);
Serial.print("AudioControlTestFreqSweep_F32: updateSignalGenerator(): setting freq to ");
Serial.println(new_freq_Hz);
setSignalFrequency_Hz(new_freq_Hz);
}
void finishTest(void) {
//disable the test instrumentation
setSignalAmplitude_dBFS(-1000.0f); //some very quiet value
setSignalFrequency_Hz(1000.f);
//do all of the common actions
AudioControlSignalTester_F32::finishTest();
//print results
printTableOfResults(&Serial);
}
//void resetState(void) {
// AudioControlSignalTester_F32::resetState();
//}
};
#endif

2
AudioConvert_F32.h
Unescape View File

@ -9,6 +9,8 @@ class AudioConvert_I16toF32 : public AudioStream_F32 //receive Int and transmits
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node //GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
public: public:
AudioConvert_I16toF32(void) : AudioStream_F32(1, inputQueueArray_f32) { }; AudioConvert_I16toF32(void) : AudioStream_F32(1, inputQueueArray_f32) { };
AudioConvert_I16toF32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) { };
void update(void) { void update(void) {
//get the Int16 block //get the Int16 block
audio_block_t *int_block; audio_block_t *int_block;

187
AudioEffectCompWDRC_F32.h
Unescape View File

@ -12,150 +12,21 @@
#ifndef _AudioEffectCompWDRC_F32 #ifndef _AudioEffectCompWDRC_F32
#define _AudioEffectCompWDRC_F32 #define _AudioEffectCompWDRC_F32
class AudioCalcGainWDRC_F32; //forward declared. Actually defined in later header file, but I need this here to avoid circularity
#include <Arduino.h> #include <Arduino.h>
#include <AudioStream_F32.h> #include <AudioStream_F32.h>
#include <arm_math.h> #include <arm_math.h>
#include <AudioCalcEnvelope_F32.h> #include <AudioCalcEnvelope_F32.h>
#include "AudioCalcGainWDRC_F32.h" //has definition of CHA_WDRC #include <AudioCalcGainWDRC_F32.h> //has definition of CHA_WDRC
#include "utility/textAndStringUtils.h" #include "BTNRH_WDRC_Types.h"
//#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 class AudioEffectCompWDRC_F32 : public AudioStream_F32
{ {
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node //GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName: CompWDRC //GUI: shortName: CompressWDRC
public: public:
AudioEffectCompWDRC_F32(void): AudioStream_F32(1,inputQueueArray) { //need to modify this for user to set sample rate AudioEffectCompWDRC_F32(void): AudioStream_F32(1,inputQueueArray) { //need to modify this for user to set sample rate
setSampleRate_Hz(AUDIO_SAMPLE_RATE); setSampleRate_Hz(AUDIO_SAMPLE_RATE);
@ -178,7 +49,7 @@ class AudioEffectCompWDRC_F32 : public AudioStream_F32
if (!out_block) return; if (!out_block) return;
//do the algorithm //do the algorithm
cha_agc_channel(block->data, out_block->data, block->length); compress(block->data, out_block->data, block->length);
// transmit the block and release memory // transmit the block and release memory
AudioStream_F32::transmit(out_block); // send the FIR output AudioStream_F32::transmit(out_block); // send the FIR output
@ -188,11 +59,11 @@ class AudioEffectCompWDRC_F32 : public AudioStream_F32
//here is the function that does all the work //here is the function that does all the work
void cha_agc_channel(float *input, float *output, int cs) { //void cha_agc_channel(float *input, float *output, int cs) {
//compress(input, output, cs, &prev_env, // //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); // // 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); // compress(input, output, cs);
} //}
//void compress(float *x, float *y, int n, float *prev_env, //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) // float &alfa, float &beta, float &tkgn, float &tk, float &cr, float &bolt, float &mxdB)
@ -222,23 +93,23 @@ class AudioEffectCompWDRC_F32 : public AudioStream_F32
void setDefaultValues(void) { void setDefaultValues(void) {
//set default values...taken from CHAPRO, GHA_Demo.c from "amplify()"...ignores given sample rate //set default values...configure as limitter
//assumes that the sample rate has already been set!!!! BTNRH_WDRC::CHA_WDRC gha = {
CHA_WDRC gha = {1.0f, // attack time (ms) 5.0f, // attack time (ms)
50.0f, // release time (ms) 50.0f, // release time (ms)
24000.0f, // fs, sampling rate (Hz), THIS IS IGNORED! 24000.0f, // fs, sampling rate (Hz), THIS IS IGNORED!
119.0f, // maxdB, maximum signal (dB SPL) 115.0f, // maxdB, maximum signal (dB SPL)...assumed SPL for full-scale input signal
0.0f, // tkgain, compression-start gain 0.0f, // tkgain, compression-start gain (dB)
105.0f, // tk, compression-start kneepoint 55.0f, // tk, compression-start kneepoint (dB SPL)
10.0f, // cr, compression ratio 1.0f, // cr, compression ratio (set to 1.0 to defeat)
105.0f // bolt, broadband output limiting threshold 100.0f // bolt, broadband output limiting threshold (ie, the limiter. SPL. 10:1 comp ratio)
}; };
setParams_from_CHA_WDRC(&gha); setParams_from_CHA_WDRC(&gha);
} }
//set all of the parameters for the compressor using the CHA_WDRC structure //set all of the parameters for the compressor using the CHA_WDRC structure
//assumes that the sample rate has already been set!!! //assumes that the sample rate has already been set!!!
void setParams_from_CHA_WDRC(CHA_WDRC *gha) { void setParams_from_CHA_WDRC(BTNRH_WDRC::CHA_WDRC *gha) {
//configure the envelope calculator...assumes that the sample rate has already been set! //configure the envelope calculator...assumes that the sample rate has already been set!
calcEnvelope.setAttackRelease_msec(gha->attack,gha->release); //these are in milliseconds calcEnvelope.setAttackRelease_msec(gha->attack,gha->release); //these are in milliseconds
@ -263,8 +134,24 @@ class AudioEffectCompWDRC_F32 : public AudioStream_F32
calcEnvelope.setSampleRate_Hz(_fs_Hz); calcEnvelope.setSampleRate_Hz(_fs_Hz);
} }
void setAttackRelease_msec(const float atk_msec, const float rel_msec) {
calcEnvelope.setAttackRelease_msec(atk_msec, rel_msec);
}
void setKneeLimiter_dBSPL(float _bolt) { calcGain.setKneeLimiter_dBSPL(_bolt); }
void setKneeLimiter_dBFS(float _bolt_dBFS) { calcGain.setKneeLimiter_dBFS(_bolt_dBFS); }
void setGain_dB(float _gain_dB) { calcGain.setGain_dB(_gain_dB); } //gain at start of compression
void setKneeCompressor_dBSPL(float _tk) { calcGain.setKneeCompressor_dBSPL(_tk); }
void setKneeCompressor_dBFS(float _tk_dBFS) { calcGain.setKneeCompressor_dBFS(_tk_dBFS); }
void setCompRatio(float _cr) { calcGain.setCompRatio(_cr); }
void setMaxdB(float _maxdB) { calcGain.setMaxdB(_maxdB); };
float getCurrentLevel_dB(void) { return AudioCalcGainWDRC_F32::db2(calcEnvelope.getCurrentLevel()); } //this is 20*log10(abs(signal)) after the envelope smoothing float getCurrentLevel_dB(void) { return AudioCalcGainWDRC_F32::db2(calcEnvelope.getCurrentLevel()); } //this is 20*log10(abs(signal)) after the envelope smoothing
float getGain_dB(void) { return calcGain.getGain_dB(); } //returns the linear gain of the system
float getCurrentGain(void) { return calcGain.getCurrentGain(); }
float getCurrentGain_dB(void) { return calcGain.getCurrentGain_dB(); }
AudioCalcEnvelope_F32 calcEnvelope; AudioCalcEnvelope_F32 calcEnvelope;
AudioCalcGainWDRC_F32 calcGain; AudioCalcGainWDRC_F32 calcGain;

17
AudioEffectCompressor_F32.h
Unescape View File

@ -22,13 +22,20 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
public: public:
//constructor //constructor
AudioEffectCompressor_F32(void) : AudioStream_F32(1, inputQueueArray_f32) { AudioEffectCompressor_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {
setDefaultValues(AUDIO_SAMPLE_RATE); resetStates();
};
AudioEffectCompressor_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) {
setDefaultValues(settings.sample_rate_Hz); resetStates();
};
void setDefaultValues(const float sample_rate_Hz) {
setThresh_dBFS(-20.0f); //set the default value for the threshold for compression setThresh_dBFS(-20.0f); //set the default value for the threshold for compression
setCompressionRatio(5.0f); //set the default copression ratio setCompressionRatio(5.0f); //set the default copression ratio
setAttack_sec(0.005f, AUDIO_SAMPLE_RATE); //default to this value setAttack_sec(0.005f, sample_rate_Hz); //default to this value
setRelease_sec(0.200f, AUDIO_SAMPLE_RATE); //default to this value setRelease_sec(0.200f, sample_rate_Hz); //default to this value
setHPFilterCoeff(); enableHPFilter(true); //enable the HP filter to remove any DC offset from the audio setHPFilterCoeff(); enableHPFilter(true); //enable the HP filter to remove any DC offset from the audio
resetStates(); }
};
//here's the method that does all the work //here's the method that does all the work
void update(void) { void update(void) {
@ -133,7 +140,7 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
inst_targ_gain_dB_block->data, //this is the output inst_targ_gain_dB_block->data, //this is the output
above_thresh_dB_block->length); above_thresh_dB_block->length);
// compute the instantaneous gai...which is the difference between the target level and the original level // compute the instantaneous gain...which is the difference between the target level and the original level
arm_sub_f32(inst_targ_gain_dB_block->data, //CMSIS DSP for "subtract two vectors element-by-element" arm_sub_f32(inst_targ_gain_dB_block->data, //CMSIS DSP for "subtract two vectors element-by-element"
above_thresh_dB_block->data, //this is the vector to be subtracted above_thresh_dB_block->data, //this is the vector to be subtracted
inst_targ_gain_dB_block->data, //this is the output inst_targ_gain_dB_block->data, //this is the output

1
AudioEffectEmpty_F32.h
Unescape View File

@ -23,6 +23,7 @@ class AudioEffectEmpty_F32 : public AudioStream_F32
public: public:
//constructor //constructor
AudioEffectEmpty_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {}; AudioEffectEmpty_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {};
AudioEffectEmpty_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) {};
//here's the method that does all the work //here's the method that does all the work
void update(void) { void update(void) {

5
AudioEffectGain_F32.h
Unescape View File

@ -9,8 +9,8 @@
* MIT License. use at your own risk. * MIT License. use at your own risk.
*/ */
#ifndef _AudioEffectGain_h #ifndef _AudioEffectGain_F32_h
#define _AudioEffectGain_h #define _AudioEffectGain_F32_h
#include <arm_math.h> //ARM DSP extensions. for speed! #include <arm_math.h> //ARM DSP extensions. for speed!
#include <AudioStream_F32.h> #include <AudioStream_F32.h>
@ -21,6 +21,7 @@ class AudioEffectGain_F32 : public AudioStream_F32
public: public:
//constructor //constructor
AudioEffectGain_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {}; AudioEffectGain_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {};
AudioEffectGain_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) {};
//here's the method that does all the work //here's the method that does all the work
void update(void) { void update(void) {

41
AudioFilterBiquad_F32.cpp
Unescape View File

@ -0,0 +1,41 @@
/*
* AudioFilterBiquad_F32.cpp
*
* Chip Audette, OpenAudio, Apr 2017
*
* MIT License, Use at your own risk.
*
*/
#include "AudioFilterBiquad_F32.h"
void AudioFilterBiquad_F32::update(void)
{
audio_block_f32_t *block;
block = AudioStream_F32::receiveWritable_f32();
if (!block) return;
// If there's no coefficient table, give up.
if (coeff_p == NULL) {
AudioStream_F32::release(block);
return;
}
// do passthru
if (coeff_p == IIR_F32_PASSTHRU) {
// Just passthrough
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
return;
}
// 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);
}

204
AudioFilterBiquad_F32.h
Unescape View File

@ -0,0 +1,204 @@
/*
* AudioFilterBiquad_F32
*
* Created: Chip Audette (OpenAudio) Feb 2017
*
* License: MIT License. Use at your own risk.
*
*/
#ifndef _filter_iir_f32
#define _filter_iir_f32
#include "Arduino.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
// Indicates that the code should just pass through the audio
// without any filtering (as opposed to doing nothing at all)
#define IIR_F32_PASSTHRU ((const float32_t *) 1)
#define IIR_MAX_STAGES 1 //meaningless right now
class AudioFilterBiquad_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:IIR
public:
AudioFilterBiquad_F32(void): AudioStream_F32(1,inputQueueArray), coeff_p(IIR_F32_PASSTHRU) {
setSampleRate_Hz(AUDIO_SAMPLE_RATE_EXACT);
}
AudioFilterBiquad_F32(const AudioSettings_F32 &settings):
AudioStream_F32(1,inputQueueArray), coeff_p(IIR_F32_PASSTHRU) {
setSampleRate_Hz(settings.sample_rate_Hz);
}
void begin(const float32_t *cp, int n_stages = 1) {
coeff_p = cp;
// Initialize FIR instance (ARM DSP Math Library)
if (coeff_p && (coeff_p != IIR_F32_PASSTHRU) && n_stages <= IIR_MAX_STAGES) {
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html
arm_biquad_cascade_df1_init_f32(&iir_inst, n_stages, (float32_t *)coeff_p, &StateF32[0]);
}
}
void end(void) {
coeff_p = NULL;
}
void setSampleRate_Hz(float _fs_Hz) { sampleRate_Hz = _fs_Hz; }
void setBlockDC(void) {
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
//Use matlab to compute the coeff for HP at 40Hz: [b,a]=butter(2,40/(44100/2),'high'); %assumes fs_Hz = 44100
float32_t b[] = {8.173653471988667e-01, -1.634730694397733e+00, 8.173653471988667e-01}; //from Matlab
float32_t a[] = { 1.000000000000000e+00, -1.601092394183619e+00, 6.683689946118476e-01}; //from Matlab
setFilterCoeff_Matlab(b, a);
}
void setFilterCoeff_Matlab(float32_t b[], float32_t a[]) { //one stage of N=2 IIR
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
//Use matlab to compute the coeff, such as: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
coeff[0] = b[0]; coeff[1] = b[1]; coeff[2] = b[2]; //here are the matlab "b" coefficients
coeff[3] = -a[1]; coeff[4] = -a[2]; //the DSP needs the "a" terms to have opposite sign vs Matlab ;
begin(coeff);
}
//note: stage is currently ignored
void setCoefficients(int stage, float c[]) {
if (stage > 0) {
if (Serial) {
Serial.println(F("AudioFilterBiquad_F32: setCoefficients: *** ERROR ***"));
Serial.print(F(" : This module only accepts one stage."));
Serial.print(F(" : You are attempting to set stage "));Serial.print(stage);
Serial.print(F(" : Ignoring this filter."));
}
return;
}
coeff[0] = c[0];
coeff[1] = c[1];
coeff[2] = c[2];
coeff[3] = -c[3];
coeff[4] = -c[4];
begin(coeff);
}
// Compute common filter functions
// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
//void setLowpass(uint32_t stage, float frequency, float q = 0.7071) {
void setLowpass(uint32_t stage, float frequency, float q = 0.7071) {
//int coeff[5];
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
//double scale = 1073741824.0 / (1.0 + alpha);
double scale = 1.0 / (1.0+alpha); // which is equal to 1.0 / a0
/* b0 */ coeff[0] = ((1.0 - cosW0) / 2.0) * scale;
/* b1 */ coeff[1] = (1.0 - cosW0) * scale;
/* b2 */ coeff[2] = coeff[0];
/* a1 */ coeff[3] = (-2.0 * cosW0) * scale;
/* a2 */ coeff[4] = (1.0 - alpha) * scale;
setCoefficients(stage, coeff);
}
void setHighpass(uint32_t stage, float frequency, float q = 0.7071) {
//int coeff[5];
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
double scale = 1.0 / (1.0+alpha); // which is equal to 1.0 / a0
/* b0 */ coeff[0] = ((1.0 + cosW0) / 2.0) * scale;
/* b1 */ coeff[1] = -(1.0 + cosW0) * scale;
/* b2 */ coeff[2] = coeff[0];
/* a1 */ coeff[3] = (-2.0 * cosW0) * scale;
/* a2 */ coeff[4] = (1.0 - alpha) * scale;
setCoefficients(stage, coeff);
}
void setBandpass(uint32_t stage, float frequency, float q = 1.0) {
//int coeff[5];
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
double scale = 1.0 / (1.0+alpha); // which is equal to 1.0 / a0
/* b0 */ coeff[0] = alpha * scale;
/* b1 */ coeff[1] = 0;
/* b2 */ coeff[2] = (-alpha) * scale;
/* a1 */ coeff[3] = (-2.0 * cosW0) * scale;
/* a2 */ coeff[4] = (1.0 - alpha) * scale;
setCoefficients(stage, coeff);
}
void setNotch(uint32_t stage, float frequency, float q = 1.0) {
//int coeff[5];
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
double scale = 1.0 / (1.0+alpha); // which is equal to 1.0 / a0
/* b0 */ coeff[0] = scale;
/* b1 */ coeff[1] = (-2.0 * cosW0) * scale;
/* b2 */ coeff[2] = coeff[0];
/* a1 */ coeff[3] = (-2.0 * cosW0) * scale;
/* a2 */ coeff[4] = (1.0 - alpha) * scale;
setCoefficients(stage, coeff);
}
void setLowShelf(uint32_t stage, float frequency, float gain, float slope = 1.0f) {
//int coeff[5];
double a = pow(10.0, gain/40.0);
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
//double alpha = (sinW0 * sqrt((a+1/a)*(1/slope-1)+2) ) / 2.0;
double cosW0 = cos(w0);
//generate three helper-values (intermediate results):
double sinsq = sinW0 * sqrt( (pow(a,2.0)+1.0)*(1.0/slope-1.0)+2.0*a );
double aMinus = (a-1.0)*cosW0;
double aPlus = (a+1.0)*cosW0;
double scale = 1.0 / ( (a+1.0) + aMinus + sinsq);
/* b0 */ coeff[0] = a * ( (a+1.0) - aMinus + sinsq ) * scale;
/* b1 */ coeff[1] = 2.0*a * ( (a-1.0) - aPlus ) * scale;
/* b2 */ coeff[2] = a * ( (a+1.0) - aMinus - sinsq ) * scale;
/* a1 */ coeff[3] = -2.0* ( (a-1.0) + aPlus ) * scale;
/* a2 */ coeff[4] = ( (a+1.0) + aMinus - sinsq ) * scale;
setCoefficients(stage, coeff);
}
void setHighShelf(uint32_t stage, float frequency, float gain, float slope = 1.0f) {
//int coeff[5];
double a = pow(10.0, gain/40.0);
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
//double alpha = (sinW0 * sqrt((a+1/a)*(1/slope-1)+2) ) / 2.0;
double cosW0 = cos(w0);
//generate three helper-values (intermediate results):
double sinsq = sinW0 * sqrt( (pow(a,2.0)+1.0)*(1.0/slope-1.0)+2.0*a );
double aMinus = (a-1.0)*cosW0;
double aPlus = (a+1.0)*cosW0;
double scale = 1.0 / ( (a+1.0) - aMinus + sinsq);
/* b0 */ coeff[0] = a * ( (a+1.0) + aMinus + sinsq ) * scale;
/* b1 */ coeff[1] = -2.0*a * ( (a-1.0) + aPlus ) * scale;
/* b2 */ coeff[2] = a * ( (a+1.0) + aMinus - sinsq ) * scale;
/* a1 */ coeff[3] = 2.0* ( (a-1.0) - aPlus ) * scale;
/* a2 */ coeff[4] = ( (a+1.0) - aMinus - sinsq ) * scale;
setCoefficients(stage, coeff);
}
void update(void);
private:
audio_block_f32_t *inputQueueArray[1];
float32_t coeff[5 * 1] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
float sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT; //default. from AudioStream.h??
// pointer to current coefficients or NULL or FIR_PASSTHRU
const float32_t *coeff_p;
// ARM DSP Math library filter instance
arm_biquad_casd_df1_inst_f32 iir_inst;
float32_t StateF32[4*IIR_MAX_STAGES];
};
#endif

55
AudioFilterFIR_F32.cpp
Unescape View File

@ -0,0 +1,55 @@
/*
* AudioFilterFIR_F32.cpp
*
* Chip Audette, OpenAudio, Apr 2017
*
* MIT License, Use at your own risk.
*
*/
#include "AudioFilterFIR_F32.h"
void AudioFilterFIR_F32::update(void)
{
audio_block_f32_t *block, *block_new;
block = AudioStream_F32::receiveReadOnly_f32();
if (!block) return;
// If there's no coefficient table, give up.
if (coeff_p == NULL) {
AudioStream_F32::release(block);
return;
}
// do passthru
if (coeff_p == FIR_F32_PASSTHRU) {
// Just passthrough
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
//Serial.println("AudioFilterFIR_F32: update(): PASSTHRU.");
return;
}
// 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);
}

53
AudioFilterFIR_F32.h
Unescape View File

@ -16,7 +16,6 @@
// Indicates that the code should just pass through the audio // Indicates that the code should just pass through the audio
// without any filtering (as opposed to doing nothing at all) // without any filtering (as opposed to doing nothing at all)
#define FIR_F32_PASSTHRU ((const float32_t *) 1) #define FIR_F32_PASSTHRU ((const float32_t *) 1)
#define FIR_MAX_COEFFS 200 #define FIR_MAX_COEFFS 200
class AudioFilterFIR_F32 : public AudioStream_F32 class AudioFilterFIR_F32 : public AudioStream_F32
@ -25,6 +24,8 @@ class AudioFilterFIR_F32 : public AudioStream_F32
public: public:
AudioFilterFIR_F32(void): AudioStream_F32(1,inputQueueArray), AudioFilterFIR_F32(void): AudioStream_F32(1,inputQueueArray),
coeff_p(FIR_F32_PASSTHRU), n_coeffs(1), configured_block_size(0) { } coeff_p(FIR_F32_PASSTHRU), n_coeffs(1), configured_block_size(0) { }
AudioFilterFIR_F32(const AudioSettings_F32 &settings): 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 //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) { begin(cp, _n_coeffs, AUDIO_BLOCK_SAMPLES); } //assume that the block size is the maximum
@ -36,8 +37,8 @@ class AudioFilterFIR_F32 : public AudioStream_F32
if (coeff_p && (coeff_p != FIR_F32_PASSTHRU) && n_coeffs <= FIR_MAX_COEFFS) { 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); arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p, &StateF32[0], block_size);
configured_block_size = block_size; configured_block_size = block_size;
Serial.print("AudioFilterFIR_F32: FIR is initialized. N_FIR = "); Serial.print(n_coeffs); //Serial.print("AudioFilterFIR_F32: FIR is initialized. N_FIR = "); Serial.print(n_coeffs);
Serial.print(", Block Size = "); Serial.println(block_size); //Serial.print(", Block Size = "); Serial.println(block_size);
//} else { //} else {
// Serial.print("AudioFilterFIR_F32: *** ERROR ***: Cound not initialize. N_FIR = "); Serial.print(n_coeffs); // Serial.print("AudioFilterFIR_F32: *** ERROR ***: Cound not initialize. N_FIR = "); Serial.print(n_coeffs);
// Serial.print(", Block Size = "); Serial.println(block_size); // Serial.print(", Block Size = "); Serial.println(block_size);
@ -45,7 +46,7 @@ class AudioFilterFIR_F32 : public AudioStream_F32
} }
} }
void end(void) { coeff_p = NULL; } void end(void) { coeff_p = NULL; }
virtual void update(void); void update(void);
//void setBlockDC(void) {} //helper function that sets this up for a first-order HP filter at 20Hz //void setBlockDC(void) {} //helper function that sets this up for a first-order HP filter at 20Hz
@ -63,50 +64,6 @@ class AudioFilterFIR_F32 : public AudioStream_F32
}; };
void AudioFilterFIR_F32::update(void)
{
audio_block_f32_t *block, *block_new;
block = AudioStream_F32::receiveReadOnly_f32();
if (!block) return;
// If there's no coefficient table, give up.
if (coeff_p == NULL) {
AudioStream_F32::release(block);
return;
}
// do passthru
if (coeff_p == FIR_F32_PASSTHRU) {
// Just passthrough
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
//Serial.println("AudioFilterFIR_F32: update(): PASSTHRU.");
return;
}
// 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 #endif

30
AudioMixer4_F32.cpp
Unescape View File

@ -1,30 +0,0 @@
#include "AudioMixer4_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, AUDIO_BLOCK_SAMPLES);
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, AUDIO_BLOCK_SAMPLES);
arm_add_f32(out->data, tmp->data, out->data, AUDIO_BLOCK_SAMPLES);
release(tmp);
release(in);
}
if (out) {
transmit(out);
release(out);
}
}

38
AudioMixer4_F32.h
Unescape View File

@ -1,38 +0,0 @@
/*
* 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)
*
* MIT License. use at your own risk.
*/
#ifndef AUDIOMIXER4F32_H
#define AUDIOMIXER4F32_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
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];
};
#endif

14
AudioMixer_F32.h
Unescape View File

@ -24,9 +24,12 @@ class AudioMixer4_F32 : public AudioStream_F32 {
//GUI: inputs:4, outputs:1 //this line used for automatic generation of GUI node //GUI: inputs:4, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:Mixer4 //GUI: shortName:Mixer4
public: public:
AudioMixer4_F32() : AudioStream_F32(4, inputQueueArray) { AudioMixer4_F32() : AudioStream_F32(4, inputQueueArray) { setDefaultValues(); }
for (int i=0; i<4; i++) multiplier[i] = 1.0; AudioMixer4_F32(const AudioSettings_F32 &settings) : AudioStream_F32(4, inputQueueArray) { setDefaultValues(); }
}
void setDefaultValues(void) {
for (int i=0; i<4; i++) multiplier[i] = 1.0;
}
virtual void update(void); virtual void update(void);
@ -44,7 +47,10 @@ class AudioMixer8_F32 : public AudioStream_F32 {
//GUI: inputs:8, outputs:1 //this line used for automatic generation of GUI node //GUI: inputs:8, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:Mixer8 //GUI: shortName:Mixer8
public: public:
AudioMixer8_F32() : AudioStream_F32(8, inputQueueArray) { AudioMixer8_F32() : AudioStream_F32(8, inputQueueArray) { setDefaultValues();}
AudioMixer8_F32(const AudioSettings_F32 &settings) : AudioStream_F32(8, inputQueueArray) { setDefaultValues();}
void setDefaultValues(void) {
for (int i=0; i<8; i++) multiplier[i] = 1.0; for (int i=0; i<8; i++) multiplier[i] = 1.0;
} }

2
AudioMultiply_F32.h
Unescape View File

@ -20,6 +20,8 @@ class AudioMultiply_F32 : public AudioStream_F32
//GUI: inputs:2, outputs:1 //this line used for automatic generation of GUI node //GUI: inputs:2, outputs:1 //this line used for automatic generation of GUI node
public: public:
AudioMultiply_F32(void) : AudioStream_F32(2, inputQueueArray_f32) {}; AudioMultiply_F32(void) : AudioStream_F32(2, inputQueueArray_f32) {};
AudioMultiply_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32) {};
void update(void); void update(void);
private: private:

29
AudioSettings_F32.cpp
Unescape View File

@ -0,0 +1,29 @@
#include "AudioSettings_F32.h"
#include "AudioStream_F32.h"
float AudioSettings_F32::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 AudioSettings_F32::processorUsage(void) {
return cpu_load_percent(AudioStream::cpu_cycles_total);
};
float AudioSettings_F32::processorUsageMax(void) {
return cpu_load_percent(AudioStream::cpu_cycles_total_max);
}
void AudioSettings_F32::processorUsageMaxReset(void) {
AudioStream::cpu_cycles_total_max = AudioStream::cpu_cycles_total;
}

18
AudioSettings_F32.h
Unescape View File

@ -0,0 +1,18 @@
#ifndef _AudioSettings_F32_
#define _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);
float processorUsage(void);
float processorUsageMax(void);
void processorUsageMaxReset(void);
};
#endif

18
AudioStream_F32.cpp
Unescape View File

@ -7,6 +7,24 @@ uint32_t AudioStream_F32::f32_memory_pool_available_mask[6];
uint8_t AudioStream_F32::f32_memory_used = 0; uint8_t AudioStream_F32::f32_memory_used = 0;
uint8_t AudioStream_F32::f32_memory_used_max = 0; uint8_t AudioStream_F32::f32_memory_used_max = 0;
audio_block_f32_t* allocate_f32_memory(const int num) {
static bool firstTime=true;
static audio_block_f32_t *data_f32;
if (firstTime == true) {
firstTime = false;
data_f32 = new audio_block_f32_t[num];
}
return data_f32;
}
void AudioMemory_F32(const int num) {
audio_block_f32_t *data_f32 = allocate_f32_memory(num);
if (data_f32 != NULL) AudioStream_F32::initialize_f32_memory(data_f32, num);
}
void AudioMemory_F32(const int num, const AudioSettings_F32 &settings) {
audio_block_f32_t *data_f32 = allocate_f32_memory(num);
if (data_f32 != NULL) AudioStream_F32::initialize_f32_memory(data_f32, num, settings);
}
// Set up the pool of audio data blocks // Set up the pool of audio data blocks
// placing them all onto the free list // placing them all onto the free list
void AudioStream_F32::initialize_f32_memory(audio_block_f32_t *data, unsigned int num) void AudioStream_F32::initialize_f32_memory(audio_block_f32_t *data, unsigned int num)

61
AudioStream_F32.h
Unescape View File

@ -15,37 +15,15 @@
#include <arm_math.h> //ARM DSP extensions. for speed! #include <arm_math.h> //ARM DSP extensions. for speed!
#include <Audio.h> //Teensy Audio Library #include <Audio.h> //Teensy Audio Library
#include "AudioSettings_F32.h"
// /////////////// class prototypes
class AudioStream_F32; class AudioStream_F32;
class AudioConnection_F32; class AudioConnection_F32;
class AudioSettings_F32;
class AudioSettings_F32 { // ///////////// class definitions
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. //create a new structure to hold audio as floating point values.
//modeled on the existing teensy audio block struct, which uses Int16 //modeled on the existing teensy audio block struct, which uses Int16
@ -91,20 +69,6 @@ class AudioConnection_F32
AudioConnection_F32 *next_dest; AudioConnection_F32 *next_dest;
}; };
#define AudioMemory_F32(num) ({ \
static audio_block_f32_t data_f32[num]; \
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)
#define AudioMemoryUsageMaxReset_F32() (AudioStream_F32::f32_memory_used_max = AudioStream_F32::f32_memory_used)
class AudioStream_F32 : public AudioStream { class AudioStream_F32 : public AudioStream {
public: public:
@ -121,12 +85,12 @@ class AudioStream_F32 : public AudioStream {
//virtual void update(audio_block_f32_t *) = 0; //virtual void update(audio_block_f32_t *) = 0;
static uint8_t f32_memory_used; static uint8_t f32_memory_used;
static uint8_t f32_memory_used_max; static uint8_t f32_memory_used_max;
static audio_block_f32_t * allocate_f32(void);
static void release(audio_block_f32_t * block);
protected: protected:
//bool active_f32; //bool active_f32;
unsigned char num_inputs_f32; unsigned char num_inputs_f32;
static audio_block_f32_t * allocate_f32(void);
static void release(audio_block_f32_t * block);
void transmit(audio_block_f32_t *block, unsigned char index = 0); void transmit(audio_block_f32_t *block, unsigned char index = 0);
audio_block_f32_t * receiveReadOnly_f32(unsigned int index = 0); audio_block_f32_t * receiveReadOnly_f32(unsigned int index = 0);
audio_block_f32_t * receiveWritable_f32(unsigned int index = 0); audio_block_f32_t * receiveWritable_f32(unsigned int index = 0);
@ -141,6 +105,21 @@ class AudioStream_F32 : public AudioStream {
static uint32_t f32_memory_pool_available_mask[6]; static uint32_t f32_memory_pool_available_mask[6];
}; };
/*
#define AudioMemory_F32(num) ({ \
static audio_block_f32_t data_f32[num]; \
AudioStream_F32::initialize_f32_memory(data_f32, num); \
})
*/
void AudioMemory_F32(const int num);
void AudioMemory_F32(const int num, const AudioSettings_F32 &settings);
#define AudioMemory_F32_wSettings(num,settings) (AudioMemory_F32(num,settings)) //for historical compatibility
#define AudioMemoryUsage_F32() (AudioStream_F32::f32_memory_used)
#define AudioMemoryUsageMax_F32() (AudioStream_F32::f32_memory_used_max)
#define AudioMemoryUsageMaxReset_F32() (AudioStream_F32::f32_memory_used_max = AudioStream_F32::f32_memory_used)
#endif #endif

181
BTNRH_WDRC_Types.h
Unescape View File

@ -0,0 +1,181 @@
#ifndef _BTNRH_WDRC_TYPES_H
#define _BTNRH_WDRC_TYPES_H
namespace BTNRH_WDRC {
// 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;
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 exp_cr[DSL_MXCH]; // compression ratio for low-SPL region (ie, the expander)
float exp_end_knee[DSL_MXCH]; // expansion-end kneepoint
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_DSL2;
/* 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;
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;
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 exp_cr; // compression ratio for low-SPL region (ie, the expander)
float exp_end_knee; // expansion-end kneepoint
float tkgain; // compression-start gain
float tk; // compression-start kneepoint
float cr; // compression ratio
float bolt; // broadband output limiting threshold
} CHA_WDRC2;
};
#endif

19
OpenAudio_ArduinoLibrary.h
Unescape View File

@ -5,20 +5,23 @@
#include "AudioCalcEnvelope_F32.h" #include "AudioCalcEnvelope_F32.h"
#include "AudioCalcGainWDRC_F32.h" #include "AudioCalcGainWDRC_F32.h"
#include "AudioConfigFIRFilterBank_F32.h" #include "AudioConfigFIRFilterBank_F32.h"
#include <AudioConvert_F32.h> #include "AudioControlTester.h"
#include "AudioConvert_F32.h"
#include "AudioEffectCompressor_F32.h"
#include "AudioEffectCompWDRC_F32.h" #include "AudioEffectCompWDRC_F32.h"
#include "AudioEffectEmpty_F32.h" #include "AudioEffectEmpty_F32.h"
#include <AudioEffectGain_F32.h> #include "AudioEffectGain_F32.h"
#include <AudioEffectCompressor_F32.h> #include "AudioFilterBiquad_F32.h"
#include <AudioFilterFIR_F32.h> #include <AudioFilterFIR_F32.h>
#include <AudioFilterIIR_F32.h> #include <AudioFilterIIR_F32.h>
#include <AudioMixer_F32.h> #include "AudioMixer_F32.h"
#include <AudioMultiply_F32.h> #include "AudioMultiply_F32.h"
#include "AudioSettings_F32.h"
#include "input_i2s_f32.h" #include "input_i2s_f32.h"
#include "output_i2s_f32.h"
#include "play_queue_f32.h" #include "play_queue_f32.h"
#include "record_queue_f32.h" #include "record_queue_f32.h"
#include "synth_pinknoise_f32.h" #include "synth_pinknoise_f32.h"
#include <synth_waveform_F32.h>
#include "synth_whitenoise_f32.h"
#include "synth_sine_f32.h" #include "synth_sine_f32.h"
#include "output_i2s_f32.h" #include "synth_waveform_F32.h"
#include "synth_whitenoise_f32.h"

146
USB_Audio_F32.h
Unescape View File

@ -0,0 +1,146 @@
/*
* USB_Audio_F32
*
* Created: Chip Audette (OpenAudio), Mar 2017
* Float32 wrapper for the Audio USB classes from the Teensy Audio Library
*
* License: MIT License. Use at your own risk.
*/
#ifndef usb_audio_f32_h_
#define usb_audio_f32_h_
//#include "Arduino.h"
#include <AudioStream_F32.h>
#include <Audio.h>
class AudioInputUSB_F32 : public AudioStream_F32
{
//GUI: inputs:0, outputs:2 //this line used for automatic generation of GUI node
//GUI: shortName:usbAudioIn //this line used for automatic generation of GUI node
public:
AudioInputUSB_F32() : AudioStream_F32(0, NULL) {
//i16_to_f32.disconnectFromUpdateAll(); //requires modification to AudioStream.h
//output_queue.disconnectFromUpdateAll(); //requires modification to AudioStream.h
makeConnections();
}
AudioInputUSB_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
//i16_to_f32.disconnectFromUpdateAll(); //requires modification to AudioStream.h
//output_queue.disconnectFromUpdateAll(); //requires modification to AudioStream.h
makeConnections();
}
void makeConnections(void) {
//make the audio connections
patchCord100_L = new AudioConnection(usb_in, 0, i16_to_f32_L, 0); //usb_in is an Int16 audio object. So, convert it!
patchCord100_R = new AudioConnection(usb_in, 1, i16_to_f32_R, 0); //usb_in is an Int16 audio object. So, convert it!
patchCord101_L = new AudioConnection_F32(i16_to_f32_L, 0, output_queue_L, 0);
patchCord101_R = new AudioConnection_F32(i16_to_f32_R, 0, output_queue_R, 0);
}
//define audio processing blocks.
AudioInputUSB usb_in; //from the original Teensy Audio Library, expects Int16 audio data
AudioConvert_I16toF32 i16_to_f32_L, i16_to_f32_R;
AudioRecordQueue_F32 output_queue_L,output_queue_R;
//define the audio connections
AudioConnection *patchCord100_L, *patchCord100_R;
AudioConnection_F32 *patchCord101_L, *patchCord101_R;
void update(void) {
//Serial.println("AudioSynthNoiseWhite_F32: update().");
output_queue_L.begin();
output_queue_R.begin();
//manually update audio blocks in the desired order
usb_in.update(); //the output should be routed directly via the AudioConnection
i16_to_f32_L.update(); // output is routed via the AudioConnection
i16_to_f32_R.update(); // output is routed via the AudioConnection
output_queue_L.update();
output_queue_R.update();
//handle the output for the left channel
audio_block_f32_t *block;
block = output_queue_L.getAudioBlock();
if (block == NULL) return;
AudioStream_F32::transmit(block,0);
output_queue_L.freeAudioBlock();
output_queue_L.end();
//handle the output for the left channel
block = output_queue_R.getAudioBlock();
if (block == NULL) return;
AudioStream_F32::transmit(block,1);
output_queue_R.freeAudioBlock();
output_queue_R.end();
}
private:
};
class AudioOutputUSB_F32 : public AudioStream_F32
{
//GUI: inputs:2, outputs:0 //this line used for automatic generation of GUI node
//GUI: shortName:usbAudioOut //this line used for automatic generation of GUI node
public:
AudioOutputUSB_F32() : AudioStream_F32(2, inputQueueArray_f32) {
makeConnections();
}
AudioOutputUSB_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32) {
makeConnections();
}
void makeConnections(void) {
//make the audio connections
patchCord100_L = new AudioConnection_F32(queue_L, 0, f32_to_i16_L, 0); //noise is an Int16 audio object. So, convert it!
patchCord100_R = new AudioConnection_F32(queue_R, 0, f32_to_i16_R, 0); //noise is an Int16 audio object. So, convert it!
patchCord101_L = new AudioConnection(f32_to_i16_L, 0, usb_out, 0); //Int16 audio connection
patchCord101_R = new AudioConnection(f32_to_i16_R, 0, usb_out, 1); //Int16 audio connection
}
//define audio processing blocks.
AudioPlayQueue_F32 queue_L,queue_R;
AudioConvert_F32toI16 f32_to_i16_L, f32_to_i16_R;
AudioOutputUSB usb_out; //from the original Teensy Audio Library, expects Int16 audio data
//define the audio connections
AudioConnection_F32 *patchCord100_L, *patchCord100_R;
AudioConnection *patchCord101_L, *patchCord101_R;
void update(void) {
//Serial.println("AudioSynthNoiseWhite_F32: update().");
//queue_L.begin();
//queue_R.begin();
//is there audio waiting for us for the left channel?
audio_block_f32_t *block;
block = receiveReadOnly_f32(0);
if (!block) return; //if no audio, return now.
//there is some audio, so execute the processing chain for the left channel
queue_L.playAudioBlock(block);
AudioStream_F32::release(block);
queue_L.update();
f32_to_i16_L.update();
//see if there is a right channel
block = receiveReadOnly_f32(1);
if (block) {
//there is a right channel. process it now
queue_R.playAudioBlock(block);
AudioStream_F32::release(block);
queue_R.update();
f32_to_i16_R.update();
}
//whether or not there was right-channel audio, update the usb_out
usb_out.update();
return;
}
private:
audio_block_f32_t *inputQueueArray_f32[2];
};
#endif

96
control_tlv320aic3206.cpp
Unescape View File

@ -142,14 +142,22 @@ bool AudioControlTLV320AIC3206::enable(void)
Wire.begin(); Wire.begin();
delay(5); delay(5);
aic_reset(); delay(100); //hard reset the AIC
//Serial.println("Hardware reset of AIC...");
#define RESET_PIN 21
pinMode(RESET_PIN,OUTPUT);
digitalWrite(RESET_PIN,HIGH);delay(50); //not reset
digitalWrite(RESET_PIN,LOW);delay(50); //reset
digitalWrite(RESET_PIN,HIGH);delay(50);//not reset
aic_reset(); delay(100); //soft reset
aic_init(); delay(100); aic_init(); delay(100);
aic_initADC(); delay(100); aic_initADC(); delay(100);
aic_initDAC(); delay(100); aic_initDAC(); delay(100);
aic_readPage(0, 27); // check a specific register - a register read test aic_readPage(0, 27); // check a specific register - a register read test
Serial.println("TLV320 enable done"); if (debugToSerial) Serial.println("TLV320 enable done");
return true; return true;
@ -174,7 +182,7 @@ bool AudioControlTLV320AIC3206::inputSelect(int n) {
// BIAS OFF // BIAS OFF
setMicBias(TYMPAN_MIC_BIAS_OFF); setMicBias(TYMPAN_MIC_BIAS_OFF);
Serial.println("Set Audio Input to Line In"); if (debugToSerial) Serial.println("Set Audio Input to Line In");
return true; return true;
} else if (n == TYMPAN_INPUT_JACK_AS_MIC) { } else if (n == TYMPAN_INPUT_JACK_AS_MIC) {
// mic-jack = IN3 // mic-jack = IN3
@ -185,7 +193,7 @@ bool AudioControlTLV320AIC3206::inputSelect(int n) {
// BIAS on, using default // BIAS on, using default
setMicBias(TYMPAN_DEFAULT_MIC_BIAS); setMicBias(TYMPAN_DEFAULT_MIC_BIAS);
Serial.println("Set Audio Input to JACK AS MIC, BIAS SET TO DEFAULT 2.5V"); if (debugToSerial) Serial.println("Set Audio Input to JACK AS MIC, BIAS SET TO DEFAULT 2.5V");
return true; return true;
} else if (n == TYMPAN_INPUT_JACK_AS_LINEIN) { } else if (n == TYMPAN_INPUT_JACK_AS_LINEIN) {
// 1 // 1
@ -197,7 +205,7 @@ bool AudioControlTLV320AIC3206::inputSelect(int n) {
// BIAS Off // BIAS Off
setMicBias(TYMPAN_MIC_BIAS_OFF); setMicBias(TYMPAN_MIC_BIAS_OFF);
Serial.println("Set Audio Input to JACK AS LINEIN, BIAS OFF"); if (debugToSerial) Serial.println("Set Audio Input to JACK AS LINEIN, BIAS OFF");
return true; return true;
} else if (n == TYMPAN_INPUT_ON_BOARD_MIC) { } else if (n == TYMPAN_INPUT_ON_BOARD_MIC) {
// on-board = IN2 // on-board = IN2
@ -207,11 +215,11 @@ bool AudioControlTLV320AIC3206::inputSelect(int n) {
aic_writeAddress(TYMPAN_MICPGA_RIGHT_NEGATIVE_REG, TYMPAN_MIC_ROUTING_NEGATIVE_CM_TO_CM1L & TYMPAN_MIC_ROUTING_RESISTANCE_DEFAULT); aic_writeAddress(TYMPAN_MICPGA_RIGHT_NEGATIVE_REG, TYMPAN_MIC_ROUTING_NEGATIVE_CM_TO_CM1L & TYMPAN_MIC_ROUTING_RESISTANCE_DEFAULT);
// BIAS Off // BIAS Off
setMicBias(TYMPAN_MIC_BIAS_OFF); setMicBias(TYMPAN_MIC_BIAS_OFF);
Serial.println("Set Audio Input to Tympan On-Board MIC, BIAS OFF"); if (debugToSerial) Serial.println("Set Audio Input to Tympan On-Board MIC, BIAS OFF");
return true; return true;
} }
Serial.print("ERROR: Unable to Select Input - Value not supported: "); Serial.print("controlTLV320AIC3206: ERROR: Unable to Select Input - Value not supported: ");
Serial.println(n); Serial.println(n);
return false; return false;
} }
@ -233,13 +241,13 @@ bool AudioControlTLV320AIC3206::setMicBias(int n) {
aic_writeAddress(TYMPAN_MIC_BIAS_REG, TYMPAN_MIC_BIAS_POWER_OFF); // power up mic bias aic_writeAddress(TYMPAN_MIC_BIAS_REG, TYMPAN_MIC_BIAS_POWER_OFF); // power up mic bias
return true; return true;
} }
Serial.print("ERROR: Unable to set MIC BIAS - Value not supported: "); Serial.print("controlTLV320AIC3206: ERROR: Unable to set MIC BIAS - Value not supported: ");
Serial.println(n); Serial.println(n);
return false; return false;
} }
void AudioControlTLV320AIC3206::aic_reset() { void AudioControlTLV320AIC3206::aic_reset() {
Serial.println("INFO: Reseting AIC"); if (debugToSerial) Serial.println("INFO: Reseting AIC");
aic_writePage(0x00, 0x01, 0x01); aic_writePage(0x00, 0x01, 0x01);
// aic_writeAddress(0x0001, 0x01); // aic_writeAddress(0x0001, 0x01);
@ -254,7 +262,7 @@ void AudioControlTLV320AIC3206::aic_reset() {
// aic_writeAddress(TYMPAN_RIGHT_MICPGA_NEGATIVE_REG, TYMPAN_MIC_ROUTING_NEGATIVE_CM_TO_CM1L & TYMPAN_MIC_ROUTING_RESISTANCE_DEFAULT); // aic_writeAddress(TYMPAN_RIGHT_MICPGA_NEGATIVE_REG, TYMPAN_MIC_ROUTING_NEGATIVE_CM_TO_CM1L & TYMPAN_MIC_ROUTING_RESISTANCE_DEFAULT);
void AudioControlTLV320AIC3206::aic_initADC() { void AudioControlTLV320AIC3206::aic_initADC() {
Serial.println("INFO: Initializing AIC ADC"); if (debugToSerial) Serial.println("INFO: Initializing AIC ADC");
aic_writeAddress(TYMPAN_ADC_PROCESSING_BLOCK_REG, PRB_R); // processing blocks - ADC aic_writeAddress(TYMPAN_ADC_PROCESSING_BLOCK_REG, PRB_R); // processing blocks - ADC
aic_writePage(1, 61, 0); // 0x3D // Select ADC PTM_R4 Power Tune? aic_writePage(1, 61, 0); // 0x3D // Select ADC PTM_R4 Power Tune?
aic_writePage(1, 71, 0b00110001); // 0x47 // Set MicPGA startup delay to 3.1ms aic_writePage(1, 71, 0b00110001); // 0x47 // Set MicPGA startup delay to 3.1ms
@ -276,21 +284,22 @@ void AudioControlTLV320AIC3206::aic_initADC() {
bool AudioControlTLV320AIC3206::setInputGain_dB(float volume) { bool AudioControlTLV320AIC3206::setInputGain_dB(float volume) {
if (volume < 0.0) { if (volume < 0.0) {
volume = 0.0; // 0.0 dB volume = 0.0; // 0.0 dB
Serial.println("WARNING: Attempting to set MIC volume outside range"); Serial.println("controlTLV320AIC3206: WARNING: Attempting to set MIC volume outside range");
} }
if (volume > 47.5) { if (volume > 47.5) {
volume = 47.5; // 47.5 dB volume = 47.5; // 47.5 dB
Serial.println("WARNING: Attempting to set MIC volume outside range"); Serial.println("controlTLV320AIC3206: WARNING: Attempting to set MIC volume outside range");
} }
Serial.print("INFO: Setting MIC volume to ");
Serial.print(volume, 1);
volume = volume * 2.0; // convert to value map (0.5 dB steps) volume = volume * 2.0; // convert to value map (0.5 dB steps)
int8_t volume_int = (int8_t) (round(volume)); // round int8_t volume_int = (int8_t) (round(volume)); // round
Serial.print(". Converted to volume map => "); if (debugToSerial) {
Serial.println(volume_int); Serial.print("INFO: Setting MIC volume to ");
Serial.print(volume, 1);
Serial.print(". Converted to volume map => ");
Serial.println(volume_int);
}
aic_writeAddress(TYMPAN_MICPGA_LEFT_VOLUME_REG, TYMPAN_MICPGA_VOLUME_ENABLE | volume_int); // enable Left MicPGA, set gain to 0 dB aic_writeAddress(TYMPAN_MICPGA_LEFT_VOLUME_REG, TYMPAN_MICPGA_VOLUME_ENABLE | volume_int); // enable Left MicPGA, set gain to 0 dB
aic_writeAddress(TYMPAN_MICPGA_RIGHT_VOLUME_REG, TYMPAN_MICPGA_VOLUME_ENABLE | volume_int); // enable Right MicPGA, set gain to 0 dB aic_writeAddress(TYMPAN_MICPGA_RIGHT_VOLUME_REG, TYMPAN_MICPGA_VOLUME_ENABLE | volume_int); // enable Right MicPGA, set gain to 0 dB
@ -318,21 +327,22 @@ bool AudioControlTLV320AIC3206::volume_dB(float volume) {
// Constrain to limits // Constrain to limits
if (volume > 24.0) { if (volume > 24.0) {
volume = 24.0; volume = 24.0;
Serial.println("WARNING: Attempting to set DAC Volume outside range"); Serial.println("controlTLV320AIC3206: WARNING: Attempting to set DAC Volume outside range");
} }
if (volume < -63.5) { if (volume < -63.5) {
volume = -63.5; volume = -63.5;
Serial.println("WARNING: Attempting to set DAC Volume outside range"); Serial.println("controlTLV320AIC3206: WARNING: Attempting to set DAC Volume outside range");
} }
Serial.print("INFO: Setting DAC volume to ");
Serial.print(volume, 1);
volume = volume * 2.0; // convert to value map (0.5 dB steps) volume = volume * 2.0; // convert to value map (0.5 dB steps)
int8_t volume_int = (int8_t) (round(volume)); // round int8_t volume_int = (int8_t) (round(volume)); // round
Serial.print(". Converted to volume map => "); if (debugToSerial) {
Serial.println(volume_int); Serial.print("INFO: Setting DAC volume to ");
Serial.print(volume, 1);
Serial.print(". Converted to volume map => ");
Serial.println(volume_int);
}
aic_writeAddress(TYMPAN_DAC_VOLUME_RIGHT_REG, volume_int); aic_writeAddress(TYMPAN_DAC_VOLUME_RIGHT_REG, volume_int);
aic_writeAddress(TYMPAN_DAC_VOLUME_LEFT_REG, volume_int); aic_writeAddress(TYMPAN_DAC_VOLUME_LEFT_REG, volume_int);
@ -340,7 +350,7 @@ bool AudioControlTLV320AIC3206::volume_dB(float volume) {
} }
void AudioControlTLV320AIC3206::aic_initDAC() { void AudioControlTLV320AIC3206::aic_initDAC() {
Serial.println("INFO: Initializing AIC DAC"); if (debugToSerial) Serial.println("INFO: Initializing AIC DAC");
// PLAYBACK SETUP // PLAYBACK SETUP
aic_writeAddress(TYMPAN_DAC_PROCESSING_BLOCK_REG, PRB_P); // processing blocks - DAC aic_writeAddress(TYMPAN_DAC_PROCESSING_BLOCK_REG, PRB_P); // processing blocks - DAC
@ -363,7 +373,7 @@ void AudioControlTLV320AIC3206::aic_initDAC() {
} }
void AudioControlTLV320AIC3206::aic_init() { void AudioControlTLV320AIC3206::aic_init() {
Serial.println("INFO: Initializing AIC"); if (debugToSerial) Serial.println("INFO: Initializing AIC");
// PLL // PLL
aic_writePage(0, 4, 3); // 0x04 low PLL clock range, MCLK is PLL input, PLL_OUT is CODEC_CLKIN aic_writePage(0, 4, 3); // 0x04 low PLL clock range, MCLK is PLL input, PLL_OUT is CODEC_CLKIN
@ -407,7 +417,7 @@ unsigned int AudioControlTLV320AIC3206::aic_readPage(uint8_t page, uint8_t reg)
Wire.write(reg); Wire.write(reg);
unsigned int result = Wire.endTransmission(); unsigned int result = Wire.endTransmission();
if (result != 0) { if (result != 0) {
Serial.print("ERROR: Read Page. Page: ");Serial.print(page); Serial.print("controlTLV320AIC3206: ERROR: Read Page. Page: ");Serial.print(page);
Serial.print(" Reg: ");Serial.print(reg); Serial.print(" Reg: ");Serial.print(reg);
Serial.print(". Received Error During Read Page: "); Serial.print(". Received Error During Read Page: ");
Serial.println(result); Serial.println(result);
@ -415,7 +425,7 @@ unsigned int AudioControlTLV320AIC3206::aic_readPage(uint8_t page, uint8_t reg)
return val; return val;
} }
if (Wire.requestFrom(AIC3206_I2C_ADDR, 1) < 1) { if (Wire.requestFrom(AIC3206_I2C_ADDR, 1) < 1) {
Serial.print("ERROR: Read Page. Page: ");Serial.print(page); Serial.print("controlTLV320AIC3206: ERROR: Read Page. Page: ");Serial.print(page);
Serial.print(" Reg: ");Serial.print(reg); Serial.print(" Reg: ");Serial.print(reg);
Serial.println(". Nothing to return"); Serial.println(". Nothing to return");
val = 400; val = 400;
@ -423,16 +433,18 @@ unsigned int AudioControlTLV320AIC3206::aic_readPage(uint8_t page, uint8_t reg)
} }
if (Wire.available() >= 1) { if (Wire.available() >= 1) {
uint16_t val = Wire.read(); uint16_t val = Wire.read();
Serial.print("INFO: Read Page. Page: ");Serial.print(page); if (debugToSerial) {
Serial.print(" Reg: ");Serial.print(reg); Serial.print("INFO: Read Page. Page: ");Serial.print(page);
Serial.print(". Received: "); Serial.print(" Reg: ");Serial.print(reg);
Serial.println(val, HEX); Serial.print(". Received: ");
Serial.println(val, HEX);
}
return val; return val;
} }
} else { } else {
Serial.print("INFO: Read Page. Page: ");Serial.print(page); Serial.print("controlTLV320AIC3206: INFO: Read Page. Page: ");Serial.print(page);
Serial.print(" Reg: ");Serial.print(reg); Serial.print(" Reg: ");Serial.print(reg);
Serial.println(". Failed to go to read page. Could not go there"); Serial.println(". Failed to go to read page. Could not go there.");
val = 500; val = 500;
return val; return val;
} }
@ -448,9 +460,11 @@ bool AudioControlTLV320AIC3206::aic_writeAddress(uint16_t address, uint8_t val)
} }
bool AudioControlTLV320AIC3206::aic_writePage(uint8_t page, uint8_t reg, uint8_t val) { bool AudioControlTLV320AIC3206::aic_writePage(uint8_t page, uint8_t reg, uint8_t val) {
Serial.print("INFO: Write Page. Page: ");Serial.print(page); if (debugToSerial) {
Serial.print(" Reg: ");Serial.print(reg); Serial.print("INFO: Write Page. Page: ");Serial.print(page);
Serial.print(" Val: ");Serial.println(val); Serial.print(" Reg: ");Serial.print(reg);
Serial.print(" Val: ");Serial.println(val);
}
if (aic_goToPage(page)) { if (aic_goToPage(page)) {
Wire.beginTransmission(AIC3206_I2C_ADDR); Wire.beginTransmission(AIC3206_I2C_ADDR);
Wire.write(reg);delay(10); Wire.write(reg);delay(10);
@ -458,7 +472,7 @@ bool AudioControlTLV320AIC3206::aic_writePage(uint8_t page, uint8_t reg, uint8_t
uint8_t result = Wire.endTransmission(); uint8_t result = Wire.endTransmission();
if (result == 0) return true; if (result == 0) return true;
else { else {
Serial.print("ERROR: Received Error During Write Page: "); Serial.print("controlTLV320AIC3206: Received Error During writePage(): Error = ");
Serial.println(result); Serial.println(result);
} }
} }
@ -467,11 +481,11 @@ bool AudioControlTLV320AIC3206::aic_writePage(uint8_t page, uint8_t reg, uint8_t
bool AudioControlTLV320AIC3206::aic_goToPage(byte page) { bool AudioControlTLV320AIC3206::aic_goToPage(byte page) {
Wire.beginTransmission(AIC3206_I2C_ADDR); Wire.beginTransmission(AIC3206_I2C_ADDR);
Wire.write(0x00); delay(10);// page register Wire.write(0x00); delay(10);// page register //was delay(10) from BPF
Wire.write(page); delay(10);// go to page Wire.write(page); delay(10);// go to page //was delay(10) from BPF
byte result = Wire.endTransmission(); byte result = Wire.endTransmission();
if (result != 0) { if (result != 0) {
Serial.print("ERROR: Received Error During GoTo Page: "); Serial.print("controlTLV320AIC3206: Received Error During goToPage(): Error = ");
Serial.println(result); Serial.println(result);
if (result == 2) { if (result == 2) {
// failed to transmit address // failed to transmit address

6
control_tlv320aic3206.h
Unescape View File

@ -16,7 +16,8 @@ class AudioControlTLV320AIC3206: public AudioControl
{ {
public: public:
//GUI: inputs:0, outputs:0 //this line used for automatic generation of GUI node //GUI: inputs:0, outputs:0 //this line used for automatic generation of GUI node
AudioControlTLV320AIC3206(void) {}; AudioControlTLV320AIC3206(void) { debugToSerial = false; };
AudioControlTLV320AIC3206(bool _debugToSerial) { debugToSerial = _debugToSerial; };
bool enable(void); bool enable(void);
bool disable(void); bool disable(void);
bool volume(float n); bool volume(float n);
@ -25,7 +26,7 @@ public:
bool inputSelect(int n); bool inputSelect(int n);
bool setInputGain_dB(float n); bool setInputGain_dB(float n);
bool setMicBias(int n); bool setMicBias(int n);
bool debugToSerial;
private: private:
void aic_reset(void); void aic_reset(void);
void aic_init(void); void aic_init(void);
@ -35,6 +36,7 @@ private:
bool aic_writePage(uint8_t page, uint8_t reg, uint8_t val); bool aic_writePage(uint8_t page, uint8_t reg, uint8_t val);
bool aic_writeAddress(uint16_t address, uint8_t val); bool aic_writeAddress(uint16_t address, uint8_t val);
bool aic_goToPage(uint8_t page); bool aic_goToPage(uint8_t page);
}; };
#define TYMPAN_OUTPUT_HEADPHONE_JACK_OUT 1 #define TYMPAN_OUTPUT_HEADPHONE_JACK_OUT 1

4
examples/BasicCompressor_Float/BasicCompressor_Float.ino
Unescape View File

@ -23,10 +23,10 @@
//create audio library objects for handling the audio //create audio library objects for handling the audio
AudioControlSGTL5000_Extended sgtl5000; //controller for the Teensy Audio Board AudioControlSGTL5000_Extended sgtl5000; //controller for the Teensy Audio Board
AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo. AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo.
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h
AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h
AudioEffectCompressor_F32 comp1, comp2; AudioEffectCompressor_F32 comp1, comp2;
AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
//Make all of the audio connections, with the option of USB audio in and out //Make all of the audio connections, with the option of USB audio in and out
//note that you ALWAYS have to have an I2S connection (either in or out) to have a clock //note that you ALWAYS have to have an I2S connection (either in or out) to have a clock

2
examples/BasicGain_Float/BasicGain_Float.ino
Unescape View File

@ -24,10 +24,10 @@
//create audio library objects for handling the audio //create audio library objects for handling the audio
AudioControlSGTL5000 sgtl5000_1; //controller for the Teensy Audio Board AudioControlSGTL5000 sgtl5000_1; //controller for the Teensy Audio Board
AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo. AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo.
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h
AudioEffectGain_F32 gain1, gain2; //Applies digital gain to audio data. Expected Float data. AudioEffectGain_F32 gain1, gain2; //Applies digital gain to audio data. Expected Float data.
AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
//Make all of the audio connections //Make all of the audio connections
AudioConnection patchCord1(i2s_in, 0, int2Float1, 0); //connect the Left input to the Left Int->Float converter AudioConnection patchCord1(i2s_in, 0, int2Float1, 0); //connect the Left input to the Left Int->Float converter

6
examples/MixStereoToMono_Float/MixStereoToMono_Float.ino
Unescape View File

@ -23,13 +23,13 @@
//create audio library objects for handling the audio //create audio library objects for handling the audio
AudioControlSGTL5000_Extended sgtl5000; //controller for the Teensy Audio Board AudioControlSGTL5000_Extended sgtl5000; //controller for the Teensy Audio Board
AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo. AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo.
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h
AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h
AudioMixer4_F32 mixer; //mix floating point data AudioMixer4_F32 mixer; //mix floating point data
AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
//Do you want to use the USB audio as your input, or do you want to use i2s as your input? //Do you want to use the USB audio as your input, or do you want to use i2s as your input?
#define DO_USB 1 //set to 1 to enable USB audio. Be sure to go under the "Tools" menu and do "USB Type" -> "Audio" #define DO_USB 0 //set to 1 to enable USB audio. Be sure to go under the "Tools" menu and do "USB Type" -> "Audio"
#if DO_USB #if DO_USB
AudioInputUSB usb_in; AudioInputUSB usb_in;
AudioConnection patchCord1(usb_in, 0, int2Float1, 0); AudioConnection patchCord1(usb_in, 0, int2Float1, 0);

4
examples/MyAudioEffect_Float/MyAudioEffect_Float.ino
Unescape View File

@ -24,10 +24,10 @@
//create audio library objects for handling the audio //create audio library objects for handling the audio
AudioControlSGTL5000 sgtl5000_1; //controller for the Teensy Audio Board AudioControlSGTL5000 sgtl5000_1; //controller for the Teensy Audio Board
AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo. AudioInputI2S i2s_in; //Digital audio *from* the Teensy Audio Board ADC. Sends Int16. Stereo.
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h AudioConvert_I16toF32 int2Float1, int2Float2; //Converts Int16 to Float. See class in AudioStream_F32.h
AudioEffectMine_F32 effect1, effect2; //This is your own algorithms
AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h AudioConvert_F32toI16 float2Int1, float2Int2; //Converts Float to Int16. See class in AudioStream_F32.h
AudioEffectMine_F32 effect1, effect2; //This is your own algorithms
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
//Make all of the audio connections //Make all of the audio connections
AudioConnection patchCord1(i2s_in, 0, int2Float1, 0); //connect the Left input to the Left Int->Float converter AudioConnection patchCord1(i2s_in, 0, int2Float1, 0); //connect the Left input to the Left Int->Float converter

8
examples/OscillatorWithPitchmod_Float/OscillatorWithPitchmod_Float.ino
Unescape View File

@ -19,15 +19,15 @@
#include <OpenAudio_ArduinoLibrary.h> //for AudioConvert_I16toF32, AudioConvert_F32toI16, and AudioEffectGain_F32 #include <OpenAudio_ArduinoLibrary.h> //for AudioConvert_I16toF32, AudioConvert_F32toI16, and AudioEffectGain_F32
#define DO_USB 1 //set to 1 to enable USB audio. Be sure to go under the "Tools" menu and do "USB Type" -> "Audio" #define DO_USB 0 //set to 1 to enable USB audio. Be sure to go under the "Tools" menu and do "USB Type" -> "Audio"
//create audio library objects for handling the audio //create audio library objects for handling the audio
AudioControlSGTL5000_Extended sgtl5000; //controller for the Teensy Audio Board AudioControlSGTL5000_Extended sgtl5000; //controller for the Teensy Audio Board
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
AudioConvert_F32toI16 float2Int; //Converts Float to Int16. See class in AudioStream_F32.h
AudioSynthWaveform_F32 osc1; // Audio-rate oscillator. AudioSynthWaveform_F32 osc1; // Audio-rate oscillator.
AudioSynthWaveform_F32 lfo1; // Low-frequency oscillator to modulate the main oscillator with. AudioSynthWaveform_F32 lfo1; // Low-frequency oscillator to modulate the main oscillator with.
AudioConvert_F32toI16 float2Int; //Converts Float to Int16. See class in AudioStream_F32.h
AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
AudioConnection_F32 patchCord1(osc1, 0, float2Int, 0); // connect the oscillator to the float 2 int converter AudioConnection_F32 patchCord1(osc1, 0, float2Int, 0); // connect the oscillator to the float 2 int converter
AudioConnection_F32 patchCord2(lfo1, 0, osc1, 0); // connect the output of the lfo to the mod input of the oscillator AudioConnection_F32 patchCord2(lfo1, 0, osc1, 0); // connect the output of the lfo to the mod input of the oscillator

2
output_i2s_f32.cpp
Unescape View File

@ -41,7 +41,7 @@ float setI2SFreq(const float freq_Hz) {
} __attribute__((__packed__)) tmclk; } __attribute__((__packed__)) tmclk;
const int numfreqs = 16; const int numfreqs = 16;
const int samplefreqs[numfreqs] = { 2000, 8000, 11025, 16000, 22050, 24000, 32000, 44100, 44117.64706 , 48000, 88200, 44117.64706 * 2, 96000, 176400, 44117.64706 * 4, 192000}; const int samplefreqs[numfreqs] = { 2000, 8000, 11025, 16000, 22050, 24000, 32000, 44100, (int)44117.64706 , 48000, 88200, (int)(44117.64706 * 2), 96000, 176400, (int)(44117.64706 * 4), 192000};
#if (F_PLL==16000000) #if (F_PLL==16000000)
const tmclk clkArr[numfreqs] = {{4, 125}, {16, 125}, {148, 839}, {32, 125}, {145, 411}, {48, 125}, {64, 125}, {151, 214}, {12, 17}, {96, 125}, {151, 107}, {24, 17}, {192, 125}, {127, 45}, {48, 17}, {255, 83} }; const tmclk clkArr[numfreqs] = {{4, 125}, {16, 125}, {148, 839}, {32, 125}, {145, 411}, {48, 125}, {64, 125}, {151, 214}, {12, 17}, {96, 125}, {151, 107}, {24, 17}, {192, 125}, {127, 45}, {48, 17}, {255, 83} };

2
play_queue_f32.h
Unescape View File

@ -19,6 +19,8 @@ class AudioPlayQueue_F32 : public AudioStream_F32
public: public:
AudioPlayQueue_F32(void) : AudioStream_F32(0, NULL), AudioPlayQueue_F32(void) : AudioStream_F32(0, NULL),
userblock(NULL), head(0), tail(0) { } userblock(NULL), head(0), tail(0) { }
AudioPlayQueue_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL),
userblock(NULL), head(0), tail(0) { }
//void play(int16_t data); //void play(int16_t data);
//void play(const int16_t *data, uint32_t len); //void play(const int16_t *data, uint32_t len);
//void play(float32_t data); //void play(float32_t data);

2
record_queue_f32.h
Unescape View File

@ -20,6 +20,8 @@ class AudioRecordQueue_F32 : public AudioStream_F32
public: public:
AudioRecordQueue_F32(void) : AudioStream_F32(1, inputQueueArray), AudioRecordQueue_F32(void) : AudioStream_F32(1, inputQueueArray),
userblock(NULL), head(0), tail(0), enabled(0) { } userblock(NULL), head(0), tail(0), enabled(0) { }
AudioRecordQueue_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray),
userblock(NULL), head(0), tail(0), enabled(0) { }
void begin(void) { void begin(void) {
clear(); clear();
enabled = 1; enabled = 1;

11
synth_pinknoise_f32.h
Unescape View File

@ -44,13 +44,20 @@ class AudioSynthNoisePink_F32 : public AudioStream_F32
//GUI: shortName:pinknoise //this line used for automatic generation of GUI node //GUI: shortName:pinknoise //this line used for automatic generation of GUI node
public: public:
AudioSynthNoisePink_F32() : AudioStream_F32(0, NULL) { AudioSynthNoisePink_F32() : AudioStream_F32(0, NULL) {
setDefaultValues();
enabled = 1;
}
AudioSynthNoisePink_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
setDefaultValues();
enabled = 1;
}
void setDefaultValues() {
plfsr = 0x5EED41F5 + instance_cnt++; plfsr = 0x5EED41F5 + instance_cnt++;
paccu = 0; paccu = 0;
pncnt = 0; pncnt = 0;
pinc = 0x0CCC; pinc = 0x0CCC;
pdec = 0x0CCC; pdec = 0x0CCC;
enabled = 1;
} }
void amplitude(float n) { void amplitude(float n) {
if (n < 0.0) n = 0.0; if (n < 0.0) n = 0.0;

2
synth_sine_f32.cpp
Unescape View File

@ -47,7 +47,7 @@ void AudioSynthWaveformSine_F32::update(void)
#endif #endif
ph += inc; ph += inc;
block->data[i] = block->data[i] / 32768.0f; // scale to float block->data[i] = block->data[i] / 32767.0f; // scale to float
} }
phase_accumulator = ph; phase_accumulator = ph;

5
synth_whitenoise_f32.h
Unescape View File

@ -44,7 +44,10 @@ class AudioSynthNoiseWhite_F32 : public AudioStream_F32
//GUI: inputs:0, outputs:1 //this line used for automatic generation of GUI node //GUI: inputs:0, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:whitenoise //this line used for automatic generation of GUI node //GUI: shortName:whitenoise //this line used for automatic generation of GUI node
public: public:
AudioSynthNoiseWhite_F32() : AudioStream_F32(0, NULL) { AudioSynthNoiseWhite_F32() : AudioStream_F32(0, NULL) { setDefaultValues(); }
AudioSynthNoiseWhite_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) { setDefaultValues(); }
void setDefaultValues(void) {
level = 0; level = 0;
seed = 1 + instance_count++; seed = 1 + instance_count++;
} }

393
utility/BTNRH_rfft.cpp
Unescape View File

@ -0,0 +1,393 @@
//ifndef _BTNRH_FFT_H
//define _BTNRH_FFT_H
#include "BTNRH_rfft.h"
#include <math.h>
//#include "chapro.h"
//#include "cha_ff.h"
/***********************************************************/
// FFT functions adapted from G. D. Bergland, "Subroutines FAST and FSST," (1979).
// In IEEE Acoustics, Speech, and Signal Processing Society.
// "Programs for Digital Signal Processing," IEEE Press, New York,
namespace BTNRH_FFT {
static __inline int
ilog2(int n)
{
int m;
for (m = 1; m < 32; m++)
if (n == (1 << m))
return (m);
return (-1);
}
static __inline int
bitrev(int ii, int m)
{
register int jj;
jj = ii & 1;
--m;
while (--m > 0) {
ii >>= 1;
jj <<= 1;
jj |= ii & 1;
}
return (jj);
}
static __inline void
rad2(int ii, float *x0, float *x1)
{
int k;
float t;
for (k = 0; k < ii; k++) {
t = x0[k] + x1[k];
x1[k] = x0[k] - x1[k];
x0[k] = t;
}
}
static __inline void
reorder1(int m, float *x)
{
int j, k, kl, n;
float t;
k = 4;
kl = 2;
n = 1 << m;
for (j = 4; j <= n; j += 2) {
if (k > j) {
t = x[j - 1];
x[j - 1] = x[k - 1];
x[k - 1] = t;
}
k -= 2;
if (k <= kl) {
k = 2 * j;
kl = j;
}
}
}
static __inline void
reorder2(int m, float *x)
{
int ji, ij, n;
float t;
n = 1 << m;
for (ij = 0; ij <= (n - 2); ij += 2) {
ji = bitrev(ij >> 1, m) << 1;
if (ij < ji) {
t = x[ij];
x[ij] = x[ji];
x[ji] = t;
t = x[ij + 1];
x[ij + 1] = x[ji + 1];
x[ji + 1] = t;
}
}
}
/***********************************************************/
// rcfft
static void
rcrad4(int ii, int nn,
float *x0, float *x1, float *x2, float *x3,
float *x4, float *x5, float *x6, float *x7)
{
double arg, tpiovn;
float c1, c2, c3, s1, s2, s3, pr, pi, r1, r5;
float t0, t1, t2, t3, t4, t5, t6, t7;
int i0, i4, j, j0, ji, jl, jr, jlast, k, k0, kl, m, n, ni;
n = nn / 4;
for (m = 1; (1 << m) < n; m++)
continue;
tpiovn = 2 * M_PI / nn;
ji = 3;
jl = 2;
jr = 2;
ni = (n + 1) / 2;
for (i0 = 0; i0 < ni; i0++) {
if (i0 == 0) {
for (k = 0; k < ii; k++) {
t0 = x0[k] + x2[k];
t1 = x1[k] + x3[k];
x2[k] = x0[k] - x2[k];
x3[k] = x1[k] - x3[k];
x0[k] = t0 + t1;
x1[k] = t0 - t1;
}
if (nn > 4) {
k0 = ii * 4;
kl = k0 + ii;
for (k = k0; k < kl; k++) {
pr = (float) (M_SQRT1_2 * (x1[k] - x3[k]));
pi = (float) (M_SQRT1_2 * (x1[k] + x3[k]));
x3[k] = x2[k] + pi;
x1[k] = pi - x2[k];
x2[k] = x0[k] - pr;
x0[k] += pr;
}
}
} else {
arg = tpiovn * bitrev(i0, m);
c1 = cosf(arg);
s1 = sinf(arg);
c2 = c1 * c1 - s1 * s1;
s2 = c1 * s1 + c1 * s1;
c3 = c1 * c2 - s1 * s2;
s3 = c2 * s1 + s2 * c1;
i4 = ii * 4;
j0 = jr * i4;
k0 = ji * i4;
jlast = j0 + ii;
for (j = j0; j < jlast; j++) {
k = k0 + j - j0;
r1 = x1[j] * c1 - x5[k] * s1;
r5 = x1[j] * s1 + x5[k] * c1;
t2 = x2[j] * c2 - x6[k] * s2;
t6 = x2[j] * s2 + x6[k] * c2;
t3 = x3[j] * c3 - x7[k] * s3;
t7 = x3[j] * s3 + x7[k] * c3;
t0 = x0[j] + t2;
t4 = x4[k] + t6;
t2 = x0[j] - t2;
t6 = x4[k] - t6;
t1 = r1 + t3;
t5 = r5 + t7;
t3 = r1 - t3;
t7 = r5 - t7;
x0[j] = t0 + t1;
x7[k] = t4 + t5;
x6[k] = t0 - t1;
x1[j] = t5 - t4;
x2[j] = t2 - t7;
x5[k] = t6 + t3;
x4[k] = t2 + t7;
x3[j] = t3 - t6;
}
jr += 2;
ji -= 2;
if (ji <= jl) {
ji = 2 * jr - 1;
jl = jr;
}
}
}
}
//-----------------------------------------------------------
static int
rcfft2(float *x, int m)
{
int ii, nn, m2, it, n;
n = 1 << m;;
m2 = m / 2;
// radix 2
if (m <= m2 * 2) {
nn = 1;
} else {
nn = 2;
ii = n / nn;
rad2(ii, x, x + ii);
}
// radix 4
if (m2 != 0) {
for (it = 0; it < m2; it++) {
nn = nn * 4;
ii = n / nn;
rcrad4(ii, nn, x, x + ii, x + 2 * ii, x + 3 * ii,
x, x + ii, x + 2 * ii, x + 3 * ii);
}
}
// re-order
reorder1(m, x);
reorder2(m, x);
for (it = 3; it < n; it += 2)
x[it] = -x[it];
x[n] = x[1];
x[1] = 0.0;
x[n + 1] = 0.0;
return (0);
}
/***********************************************************/
// rcfft
static void
crrad4(int jj, int nn,
float *x0, float *x1, float *x2, float *x3,
float *x4, float *x5, float *x6, float *x7)
{
double arg, tpiovn;
float c1, c2, c3, s1, s2, s3;
float t0, t1, t2, t3, t4, t5, t6, t7;
int ii, j, j0, ji, jr, jl, jlast, j4, k, k0, kl, m, n, ni;
tpiovn = 2 * M_PI / nn;
ji = 3;
jl = 2;
jr = 2;
n = nn / 4;
for (m = 1; (1 << m) < n; m++)
continue;
ni = (n + 1) / 2;
for (ii = 0; ii < ni; ii++) {
if (ii == 0) {
for (k = 0; k < jj; k++) {
t0 = x0[k] + x1[k];
t1 = x0[k] - x1[k];
t2 = x2[k] * 2;
t3 = x3[k] * 2;
x0[k] = t0 + t2;
x2[k] = t0 - t2;
x1[k] = t1 + t3;
x3[k] = t1 - t3;
}
if (nn > 4) {
k0 = jj * 4;
kl = k0 + jj;
for (k = k0; k < kl; k++) {
t2 = x0[k] - x2[k];
t3 = x1[k] + x3[k];
x0[k] = (x0[k] + x2[k]) * 2;
x2[k] = (x3[k] - x1[k]) * 2;
x1[k] = (float) ((t2 + t3) * M_SQRT2);
x3[k] = (float) ((t3 - t2) * M_SQRT2);
}
}
} else {
arg = tpiovn * bitrev(ii, m);
c1 = cosf(arg);
s1 = -sinf(arg);
c2 = c1 * c1 - s1 * s1;
s2 = c1 * s1 + c1 * s1;
c3 = c1 * c2 - s1 * s2;
s3 = c2 * s1 + s2 * c1;
j4 = jj * 4;
j0 = jr * j4;
k0 = ji * j4;
jlast = j0 + jj;
for (j = j0; j < jlast; j++) {
k = k0 + j - j0;
t0 = x0[j] + x6[k];
t1 = x7[k] - x1[j];
t2 = x0[j] - x6[k];
t3 = x7[k] + x1[j];
t4 = x2[j] + x4[k];
t5 = x5[k] - x3[j];
t6 = x5[k] + x3[j];
t7 = x4[k] - x2[j];
x0[j] = t0 + t4;
x4[k] = t1 + t5;
x1[j] = (t2 + t6) * c1 - (t3 + t7) * s1;
x5[k] = (t2 + t6) * s1 + (t3 + t7) * c1;
x2[j] = (t0 - t4) * c2 - (t1 - t5) * s2;
x6[k] = (t0 - t4) * s2 + (t1 - t5) * c2;
x3[j] = (t2 - t6) * c3 - (t3 - t7) * s3;
x7[k] = (t2 - t6) * s3 + (t3 - t7) * c3;
}
jr += 2;
ji -= 2;
if (ji <= jl) {
ji = 2 * jr - 1;
jl = jr;
}
}
}
}
//-----------------------------------------------------------
static int
crfft2(float *x, int m)
{
int n, i, it, nn, jj, m2;
n = 1 << m;
x[1] = x[n];
m2 = m / 2;
// re-order
for (i = 3; i < n; i += 2)
x[i] = -x[i];
reorder2(m, x);
reorder1(m, x);
// radix 4
if (m2 != 0) {
nn = 4 * n;
for (it = 0; it < m2; it++) {
nn = nn / 4;
jj = n / nn;
crrad4(jj, nn, x, x + jj, x + 2 * jj, x + 3 * jj,
x, x + jj, x + 2 * jj, x + 3 * jj);
}
}
// radix 2
if (m > m2 * 2) {
jj = n / 2;
rad2(jj, x, x + jj);
}
return (0);
}
/***********************************************************/
// real-to-complex FFT
//FUNC(void)
void cha_fft_rc(float *x, int n)
{
int m;
// assume n is a power of two
m = ilog2(n);
rcfft2(x, m);
}
// complex-to-real inverse FFT
//FUNC(void)
void cha_fft_cr(float *x, int n)
{
int i, m;
// assume n is a power of two
m = ilog2(n);
crfft2(x, m);
// scale inverse by 1/n
for (i = 0; i < n; i++) {
x[i] /= n;
}
}
};
//endif

19
utility/BTNRH_rfft.h
Unescape View File

@ -0,0 +1,19 @@
#ifndef _BTNRH_FFT_H
#define _BTNRH_FFT_H
#include <math.h>
//#include "chapro.h"
//#include "cha_ff.h"
/***********************************************************/
// FFT functions adapted from G. D. Bergland, "Subroutines FAST and FSST," (1979).
// In IEEE Acoustics, Speech, and Signal Processing Society.
// "Programs for Digital Signal Processing," IEEE Press, New York,
namespace BTNRH_FFT {
void cha_fft_cr(float *, int);
void cha_fft_rc(float *, int);
}
#endif

384
utility/rfft.c
Unescape View File

@ -1,384 +0,0 @@
#include <math.h>
//#include "chapro.h"
//#include "cha_ff.h"
/***********************************************************/
// FFT functions adapted from G. D. Bergland, "Subroutines FAST and FSST," (1979).
// In IEEE Acoustics, Speech, and Signal Processing Society.
// "Programs for Digital Signal Processing," IEEE Press, New York,
static __inline int
ilog2(int n)
{
int m;
for (m = 1; m < 32; m++)
if (n == (1 << m))
return (m);
return (-1);
}
static __inline int
bitrev(int ii, int m)
{
register int jj;
jj = ii & 1;
--m;
while (--m > 0) {
ii >>= 1;
jj <<= 1;
jj |= ii & 1;
}
return (jj);
}
static __inline void
rad2(int ii, float *x0, float *x1)
{
int k;
float t;
for (k = 0; k < ii; k++) {
t = x0[k] + x1[k];
x1[k] = x0[k] - x1[k];
x0[k] = t;
}
}
static __inline void
reorder1(int m, float *x)
{
int j, k, kl, n;
float t;
k = 4;
kl = 2;
n = 1 << m;
for (j = 4; j <= n; j += 2) {
if (k > j) {
t = x[j - 1];
x[j - 1] = x[k - 1];
x[k - 1] = t;
}
k -= 2;
if (k <= kl) {
k = 2 * j;
kl = j;
}
}
}
static __inline void
reorder2(int m, float *x)
{
int ji, ij, n;
float t;
n = 1 << m;
for (ij = 0; ij <= (n - 2); ij += 2) {
ji = bitrev(ij >> 1, m) << 1;
if (ij < ji) {
t = x[ij];
x[ij] = x[ji];
x[ji] = t;
t = x[ij + 1];
x[ij + 1] = x[ji + 1];
x[ji + 1] = t;
}
}
}
/***********************************************************/
// rcfft
static void
rcrad4(int ii, int nn,
float *x0, float *x1, float *x2, float *x3,
float *x4, float *x5, float *x6, float *x7)
{
double arg, tpiovn;
float c1, c2, c3, s1, s2, s3, pr, pi, r1, r5;
float t0, t1, t2, t3, t4, t5, t6, t7;
int i0, i4, j, j0, ji, jl, jr, jlast, k, k0, kl, m, n, ni;
n = nn / 4;
for (m = 1; (1 << m) < n; m++)
continue;
tpiovn = 2 * M_PI / nn;
ji = 3;
jl = 2;
jr = 2;
ni = (n + 1) / 2;
for (i0 = 0; i0 < ni; i0++) {
if (i0 == 0) {
for (k = 0; k < ii; k++) {
t0 = x0[k] + x2[k];
t1 = x1[k] + x3[k];
x2[k] = x0[k] - x2[k];
x3[k] = x1[k] - x3[k];
x0[k] = t0 + t1;
x1[k] = t0 - t1;
}
if (nn > 4) {
k0 = ii * 4;
kl = k0 + ii;
for (k = k0; k < kl; k++) {
pr = (float) (M_SQRT1_2 * (x1[k] - x3[k]));
pi = (float) (M_SQRT1_2 * (x1[k] + x3[k]));
x3[k] = x2[k] + pi;
x1[k] = pi - x2[k];
x2[k] = x0[k] - pr;
x0[k] += pr;
}
}
} else {
arg = tpiovn * bitrev(i0, m);
c1 = cosf(arg);
s1 = sinf(arg);
c2 = c1 * c1 - s1 * s1;
s2 = c1 * s1 + c1 * s1;
c3 = c1 * c2 - s1 * s2;
s3 = c2 * s1 + s2 * c1;
i4 = ii * 4;
j0 = jr * i4;
k0 = ji * i4;
jlast = j0 + ii;
for (j = j0; j < jlast; j++) {
k = k0 + j - j0;
r1 = x1[j] * c1 - x5[k] * s1;
r5 = x1[j] * s1 + x5[k] * c1;
t2 = x2[j] * c2 - x6[k] * s2;
t6 = x2[j] * s2 + x6[k] * c2;
t3 = x3[j] * c3 - x7[k] * s3;
t7 = x3[j] * s3 + x7[k] * c3;
t0 = x0[j] + t2;
t4 = x4[k] + t6;
t2 = x0[j] - t2;
t6 = x4[k] - t6;
t1 = r1 + t3;
t5 = r5 + t7;
t3 = r1 - t3;
t7 = r5 - t7;
x0[j] = t0 + t1;
x7[k] = t4 + t5;
x6[k] = t0 - t1;
x1[j] = t5 - t4;
x2[j] = t2 - t7;
x5[k] = t6 + t3;
x4[k] = t2 + t7;
x3[j] = t3 - t6;
}
jr += 2;
ji -= 2;
if (ji <= jl) {
ji = 2 * jr - 1;
jl = jr;
}
}
}
}
//-----------------------------------------------------------
static int
rcfft2(float *x, int m)
{
int ii, nn, m2, it, n;
n = 1 << m;;
m2 = m / 2;
// radix 2
if (m <= m2 * 2) {
nn = 1;
} else {
nn = 2;
ii = n / nn;
rad2(ii, x, x + ii);
}
// radix 4
if (m2 != 0) {
for (it = 0; it < m2; it++) {
nn = nn * 4;
ii = n / nn;
rcrad4(ii, nn, x, x + ii, x + 2 * ii, x + 3 * ii,
x, x + ii, x + 2 * ii, x + 3 * ii);
}
}
// re-order
reorder1(m, x);
reorder2(m, x);
for (it = 3; it < n; it += 2)
x[it] = -x[it];
x[n] = x[1];
x[1] = 0.0;
x[n + 1] = 0.0;
return (0);
}
/***********************************************************/
// rcfft
static void
crrad4(int jj, int nn,
float *x0, float *x1, float *x2, float *x3,
float *x4, float *x5, float *x6, float *x7)
{
double arg, tpiovn;
float c1, c2, c3, s1, s2, s3;
float t0, t1, t2, t3, t4, t5, t6, t7;
int ii, j, j0, ji, jr, jl, jlast, j4, k, k0, kl, m, n, ni;
tpiovn = 2 * M_PI / nn;
ji = 3;
jl = 2;
jr = 2;
n = nn / 4;
for (m = 1; (1 << m) < n; m++)
continue;
ni = (n + 1) / 2;
for (ii = 0; ii < ni; ii++) {
if (ii == 0) {
for (k = 0; k < jj; k++) {
t0 = x0[k] + x1[k];
t1 = x0[k] - x1[k];
t2 = x2[k] * 2;
t3 = x3[k] * 2;
x0[k] = t0 + t2;
x2[k] = t0 - t2;
x1[k] = t1 + t3;
x3[k] = t1 - t3;
}
if (nn > 4) {
k0 = jj * 4;
kl = k0 + jj;
for (k = k0; k < kl; k++) {
t2 = x0[k] - x2[k];
t3 = x1[k] + x3[k];
x0[k] = (x0[k] + x2[k]) * 2;
x2[k] = (x3[k] - x1[k]) * 2;
x1[k] = (float) ((t2 + t3) * M_SQRT2);
x3[k] = (float) ((t3 - t2) * M_SQRT2);
}
}
} else {
arg = tpiovn * bitrev(ii, m);
c1 = cosf(arg);
s1 = -sinf(arg);
c2 = c1 * c1 - s1 * s1;
s2 = c1 * s1 + c1 * s1;
c3 = c1 * c2 - s1 * s2;
s3 = c2 * s1 + s2 * c1;
j4 = jj * 4;
j0 = jr * j4;
k0 = ji * j4;
jlast = j0 + jj;
for (j = j0; j < jlast; j++) {
k = k0 + j - j0;
t0 = x0[j] + x6[k];
t1 = x7[k] - x1[j];
t2 = x0[j] - x6[k];
t3 = x7[k] + x1[j];
t4 = x2[j] + x4[k];
t5 = x5[k] - x3[j];
t6 = x5[k] + x3[j];
t7 = x4[k] - x2[j];
x0[j] = t0 + t4;
x4[k] = t1 + t5;
x1[j] = (t2 + t6) * c1 - (t3 + t7) * s1;
x5[k] = (t2 + t6) * s1 + (t3 + t7) * c1;
x2[j] = (t0 - t4) * c2 - (t1 - t5) * s2;
x6[k] = (t0 - t4) * s2 + (t1 - t5) * c2;
x3[j] = (t2 - t6) * c3 - (t3 - t7) * s3;
x7[k] = (t2 - t6) * s3 + (t3 - t7) * c3;
}
jr += 2;
ji -= 2;
if (ji <= jl) {
ji = 2 * jr - 1;
jl = jr;
}
}
}
}
//-----------------------------------------------------------
static int
crfft2(float *x, int m)
{
int n, i, it, nn, jj, m2;
n = 1 << m;
x[1] = x[n];
m2 = m / 2;
// re-order
for (i = 3; i < n; i += 2)
x[i] = -x[i];
reorder2(m, x);
reorder1(m, x);
// radix 4
if (m2 != 0) {
nn = 4 * n;
for (it = 0; it < m2; it++) {
nn = nn / 4;
jj = n / nn;
crrad4(jj, nn, x, x + jj, x + 2 * jj, x + 3 * jj,
x, x + jj, x + 2 * jj, x + 3 * jj);
}
}
// radix 2
if (m > m2 * 2) {
jj = n / 2;
rad2(jj, x, x + jj);
}
return (0);
}
/***********************************************************/
// real-to-complex FFT
//FUNC(void)
void cha_fft_rc(float *x, int n)
{
int m;
// assume n is a power of two
m = ilog2(n);
rcfft2(x, m);
}
// complex-to-real inverse FFT
//FUNC(void)
void cha_fft_cr(float *x, int n)
{
int i, m;
// assume n is a power of two
m = ilog2(n);
crfft2(x, m);
// scale inverse by 1/n
for (i = 0; i < n; i++) {
x[i] /= n;
}
}
Write Preview
Loading…
Cancel
Save