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Compressor: move pow10 and log10_approx into the class

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pull/5/head
Chip Audette 8 years ago
parent 59d474693a
commit aae97dc956
1 changed files with 54 additions and 46 deletions
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  1. 100
      AudioEffectCompressor_F32.h

100
AudioEffectCompressor_F32.h
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@ -16,47 +16,6 @@
#include <arm_math.h> //ARM DSP extensions. https://www.keil.com/pack/doc/CMSIS/DSP/html/index.html
#include <AudioStream_F32.h>
// //// Accelerate the powf(10.0,x) function???
//#define POW10_FUNC(x) powf(10.0f,x) //standard, but slower
#define POW10_FUNC(x) expf(2.302585092994f*x) //faster: exp(log(10.0f)*x)
// //// Accelerate the log10f(x) function?
//#define LOG10_FUNC(x) log10f(x) //standard, but slower
#define LOG10_FUNC(x) log2f_approx(x)*0.3010299956639812f; //faster: log2(x)/log2(10)
//https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
/* ----------------------------------------------------------------------
** Fast approximation to the log2() function. It uses a two step
** process. First, it decomposes the floating-point number into
** a fractional component F and an exponent E. The fraction component
** is used in a polynomial approximation and then the exponent added
** to the result. A 3rd order polynomial is used and the result
** when computing db20() is accurate to 7.984884e-003 dB.
** ------------------------------------------------------------------- */
float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
float log2f_approx(float X) {
float *C = &log2f_approx_coeff[0];
float Y;
float F;
int E;
// This is the approximation to log2()
F = frexpf(fabsf(X), &E);
// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
Y = *C++;
Y *= F;
Y += (*C++);
Y *= F;
Y += (*C++);
Y *= F;
Y += (*C++);
Y += E;
return(Y);
}
class AudioEffectCompressor_F32 : public AudioStream_F32
{
public:
@ -119,7 +78,7 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
prev_level_lp_pow = wav_pow_block->data[i];
//now convert the signal power to dB (but not yet multiplied by 10.0)
level_dB_block->data[i] = LOG10_FUNC(wav_pow_block->data[i]);
level_dB_block->data[i] = log10f_approx(wav_pow_block->data[i]);
}
//limit the amount that the state of the smoothing filter can go toward negative infinity
@ -147,7 +106,7 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
//finally, convert from dB to linear gain: gain = 10^(gain_dB/20); (ie this takes care of the sqrt, too!)
arm_scale_f32(gain_dB_block->data, 1.0f/20.0f, gain_dB_block->data, gain_dB_block->length); //divide by 20
for (int i = 0; i < gain_dB_block->length; i++) gain_block->data[i] = POW10_FUNC(gain_dB_block->data[i]); //do the 10^(x)
for (int i = 0; i < gain_dB_block->length; i++) gain_block->data[i] = pow10f(gain_dB_block->data[i]); //do the 10^(x)
//release memory and return
@ -254,13 +213,13 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
void enableHPFilter(boolean flag) { use_HP_prefilter = flag; };
//methods to return information about this module
float32_t getPreGain_dB(void) { return 20.0 * log10f(pre_gain); }
float32_t getPreGain_dB(void) { return 20.0 * log10f_approx(pre_gain); }
float32_t getAttack_sec(void) { return attack_sec; }
float32_t getRelease_sec(void) { return release_sec; }
float32_t getLevelTimeConst_sec(void) { return level_lp_sec; }
float32_t getThresh_dBFS(void) { return thresh_dBFS; }
float32_t getCompressionRatio(void) { return comp_ratio; }
float32_t getCurrentLevel_dBFS(void) { return 10.0* log10f(prev_level_lp_pow); }
float32_t getCurrentLevel_dBFS(void) { return 10.0* log10f_approx(prev_level_lp_pow); }
float32_t getCurrentGain_dB(void) { return prev_gain_dB; }
void setHPFilterCoeff_N2IIR_Matlab(float32_t b[], float32_t a[]){
@ -297,7 +256,7 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
float32_t comp_ratio_const, thresh_pow_FS_wCR; //used in calcGain(); set in updateThresholdAndCompRatioConstants()
void updateThresholdAndCompRatioConstants(void) {
comp_ratio_const = 1.0f-(1.0f / comp_ratio);
thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const); //powf() is much faster than pow()
thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const);
}
//settings
@ -312,6 +271,55 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
float32_t pre_gain = -1.0; //gain to apply before the compression. negative value disables
boolean use_HP_prefilter;
// Accelerate the powf(10.0,x) function
static float32_t pow10f(float x) {
//return powf(10.0f,x) //standard, but slower
return expf(2.302585092994f*x); //faster: exp(log(10.0f)*x)
}
// Accelerate the log10f(x) function?
static float32_t log10f_approx(float x) {
//return log10f(x); //standard, but slower
return log2f_approx(x)*0.3010299956639812f; //faster: log2(x)/log2(10)
}
/* ----------------------------------------------------------------------
** Fast approximation to the log2() function. It uses a two step
** process. First, it decomposes the floating-point number into
** a fractional component F and an exponent E. The fraction component
** is used in a polynomial approximation and then the exponent added
** to the result. A 3rd order polynomial is used and the result
** when computing db20() is accurate to 7.984884e-003 dB.
** ------------------------------------------------------------------- */
//https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
//float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
static float log2f_approx(float X) {
//float *C = &log2f_approx_coeff[0];
float Y;
float F;
int E;
// This is the approximation to log2()
F = frexpf(fabsf(X), &E);
// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
//Y = *C++;
Y = 1.23149591368684f;
Y *= F;
//Y += (*C++);
Y += -4.11852516267426f;
Y *= F;
//Y += (*C++);
Y += 6.02197014179219f;
Y *= F;
//Y += (*C++);
Y += -3.13396450166353f;
Y += E;
return(Y);
}
};
#endif

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