/*
* AudioCalcGainWDRC_F32
*
* Created: Chip Audette, Feb 2017
* Purpose: This module calculates the gain needed for wide dynamic range compression.
* Derived From: Core algorithm is from "WDRC_circuit"
* WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro
* As of Feb 2017, CHAPRO license is listed as "Creative Commons?"
*
* This processes a single stream fo audio data (ie, it is mono)
*
* MIT License. use at your own risk.
*/
#ifndef _AudioCalcGainWDRC_F32_h
#define _AudioCalcGainWDRC_F32_h
#include <arm_math.h> //ARM DSP extensions. for speed!
#include <AudioStream_F32.h>
#include "BTNRH_WDRC_Types.h"
class AudioCalcGainWDRC_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:calc_WDRCGain
public:
//constructors
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
void update(void) {
//get the input audio data block
audio_block_f32_t *in_block = AudioStream_F32::receiveReadOnly_f32(); // must be the envelope!
if (!in_block) return;
//prepare an output data block
audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
if (!out_block) return;
// ////////////////////// do the processing here!
calcGainFromEnvelope(in_block->data, out_block->data, in_block->length);
out_block->length = in_block->length; out_block->fs_Hz = in_block->fs_Hz;
//transmit the block and be done
AudioStream_F32::transmit(out_block);
AudioStream_F32::release(out_block);
AudioStream_F32::release(in_block);
}
void calcGainFromEnvelope(float *env, float *gain_out, const int n) {
//env = input, signal envelope (not the envelope of the power, but the envelope of the signal itslef)
//gain = output, the gain in natural units (not power, not dB)
//n = input, number of samples to process in each vector
//prepare intermediate data block
audio_block_f32_t *env_dB_block = AudioStream_F32::allocate_f32();
if (!env_dB_block) return;
//convert to dB 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
// apply wide-dynamic range compression
WDRC_circuit_gain(env_dB_block->data, gain_out, n, tkgn, tk, cr, bolt);
AudioStream_F32::release(env_dB_block);
}
//original call to WDRC_circuit
//void WDRC_circuit(float *x, float *y, float *pdb, int n, float tkgn, float tk, float cr, float bolt)
//void WDRC_circuit(float *orig_signal, float *signal_out, float *env_dB, int n, float tkgn, float tk, float cr, float bolt)
//modified to output just the gain instead of the fully processed signal
void WDRC_circuit_gain(float *env_dB, float *gain_out, const int n,
const float tkgn, const float tk, const float cr, const float bolt)
//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;
int k;
float *pdb = env_dB; //just rename it to keep the code below unchanged (input SPL dB)
float tk_tmp = tk; //temporary, threshold for start of compression (input SPL dB)
if ((tk_tmp + tkgn) > bolt) { //after gain, would the compression threshold be above the output-limitting threshold ("bolt")
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); //intermediate calc
pblt = cr * (bolt - tkgo); //calc input level (dB) where we need to start limiting, no just compression
const float cr_const = ((1.0f / cr) - 1.0f); //pre-calc a constant that we'll need later
for (k = 0; k < n; k++) { //loop over each sample
if ((pdb[k] < tk_tmp) && (cr >= 1.0f)) { //if below threshold and we're compressing
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 {
gdb = cr_const * pdb[k] + tkgo;
}
gain_out[k] = undb2(gdb);
//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) { //set as limiter
BTNRH_WDRC::CHA_WDRC gha = {
5.0f, // attack time (ms)
50.0f, // release time (ms)
24000.0f, // fs, sampling rate (Hz), THIS IS IGNORED!
115.0f, // maxdB, maximum signal (dB SPL)...assumed SPL for full-scale input signal
0.0f, // tkgain, compression-start gain (dB)
55.0f, // tk, compression-start kneepoint (dB SPL)
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_from_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
}
void setParams(float _maxdB, float _tkgain, float _cr, float _tk, float _bolt) {
maxdB = _maxdB;
tkgn = _tkgain;
tk = _tk;
cr = _cr;
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 db2(const float &x) { return 6.020599913279623f*log2f_approx(x); } //faster: 20*log2_approx(x)/log2(10); this is approximate
/* ----------------------------------------------------------------------
** Fast approximation to the log2() function. It uses a two step
** process. First, it decomposes the floating-point number into
** a fractional component F and an exponent E. The fraction component
** is used in a polynomial approximation and then the exponent added
** to the result. A 3rd order polynomial is used and the result
** when computing db20() is accurate to 7.984884e-003 dB.
** ------------------------------------------------------------------- */
//https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
static float log2f_approx(float X) {
//float *C = &log2f_approx_coeff[0];
float Y;
float F;
int E;
// This is the approximation to log2()
F = frexpf(fabsf(X), &E);
// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
Y = 1.23149591368684f; //C[0]
Y *= F;
Y += -4.11852516267426f; //C[1]
Y *= F;
Y += 6.02197014179219f; //C[2]
Y *= F;
Y += -3.13396450166353f; //C[3]
Y += E;
return(Y);
}
private:
audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
float maxdB, tkgn, tk, cr, bolt;
float last_gain = 1.0; //what was the last gain value computed for the signal
};
#endif