Compressor: faster! via POW10 and LOG10 substitutions

pull/5/head
Chip Audette 8 years ago
parent 561d783e54
commit f8bc5a38f5
  1. 48
      AudioEffectCompressor_F32.h

@ -16,6 +16,47 @@
#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:
@ -78,7 +119,7 @@ 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] = log10f(wav_pow_block->data[i]);
level_dB_block->data[i] = LOG10_FUNC(wav_pow_block->data[i]);
}
//limit the amount that the state of the smoothing filter can go toward negative infinity
@ -105,8 +146,9 @@ class AudioEffectCompressor_F32 : public AudioStream_F32
calcSmoothedGain_dB(inst_targ_gain_dB_block,gain_dB_block);
//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] = powf(10.0f,gain_dB_block->data[i]); //do the 10^(x)
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)
//release memory and return
AudioStream_F32::release(gain_dB_block);

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