/* * 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 DSP extensions. for speed! #include #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