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429 lines
17 KiB
429 lines
17 KiB
/* From https://github.com/chipaudette/OpenAudio_ArduinoLibrary */
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/*
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AudioEffectCompressor
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Created: Chip Audette, Dec 2016 - Jan 2017
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Purpose; Apply dynamic range compression to the audio stream.
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Assumes floating-point data.
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This processes a single stream fo audio data (ie, it is mono)
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MIT License. use at your own risk.
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*/
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#ifndef _COMPRESSOR_H
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#define _COMPRESSOR_H
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#ifndef TEENSYDUINO
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#include <arm_math.h> //ARM DSP extensions. https://www.keil.com/pack/doc/CMSIS/DSP/html/index.html
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#include "synth.h"
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/*
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static const float zeroblock_f32[] = {
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0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
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#if AUDIO_BLOCK_SAMPLES > 16
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0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
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#endif
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#if AUDIO_BLOCK_SAMPLES > 32
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0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
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#endif
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#if AUDIO_BLOCK_SAMPLES > 48
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0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
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#endif
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#if AUDIO_BLOCK_SAMPLES > 64
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0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
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#endif
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#if AUDIO_BLOCK_SAMPLES > 80
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0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
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#endif
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#if AUDIO_BLOCK_SAMPLES > 96
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0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
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#endif
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#if AUDIO_BLOCK_SAMPLES > 112
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0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0,
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#endif
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#if AUDIO_BLOCK_SAMPLES > 128
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#error AUDIO_BLOCK_SAMPLES > 128 is a problem for this class
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#endif
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};
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*/
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class Compressor
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{
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//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
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public:
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//constructor
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Compressor(const float sample_rate_Hz) {
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//setDefaultValues(AUDIO_SAMPLE_RATE); resetStates();
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setDefaultValues(sample_rate_Hz);
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resetStates();
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};
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void setDefaultValues(const float sample_rate_Hz) {
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setThresh_dBFS(-20.0f); //set the default value for the threshold for compression
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setCompressionRatio(5.0f); //set the default copression ratio
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setAttack_sec(0.005f, sample_rate_Hz); //default to this value
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setRelease_sec(0.200f, sample_rate_Hz); //default to this value
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setHPFilterCoeff(); enableHPFilter(true); //enable the HP filter to remove any DC offset from the audio
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}
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//here's the method that does all the work
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void doCompression(float *audio_block, uint16_t len) {
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//Serial.println("AudioEffectGain_F32: updating."); //for debugging.
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if (!audio_block) {
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printf("No audio_block available for Compressor!\n");
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return;
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}
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//apply a high-pass filter to get rid of the DC offset
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if (use_HP_prefilter)
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arm_biquad_cascade_df1_f32(&hp_filt_struct, audio_block, audio_block, len);
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//apply the pre-gain...a negative gain value will disable
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if (pre_gain > 0.0f)
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arm_scale_f32(audio_block, pre_gain, audio_block, len); //use ARM DSP for speed!
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//calculate the level of the audio (ie, calculate a smoothed version of the signal power)
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float* audio_level_dB_block = float[len];
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if(!audio_level_dB_block)
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{
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printf("Cannot allocate memory for \"audio_level_dB_block\" - stopping\n");
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while(1);
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}
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//arm_copy_f32(zeroblock_f32,audio_level_dB_block,len);
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if(audio_level_dB_block)
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calcAudioLevel_dB(audio_block, audio_level_dB_block, len); //returns through audio_level_dB_block
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//compute the desired gain based on the observed audio level
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float* gain_block=new float[len];
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if(!gain_block)
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{
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printf("Cannot allocate memory for \"gain_block\" - stopping\n");
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while(1);
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}
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//arm_copy_f32(zeroblock_f32,gain_block,len);
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if(gain_block)
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{
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calcGain(audio_level_dB_block, gain_block, len); //returns through gain_block
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//apply the desired gain...store the processed audio back into audio_block
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arm_mult_f32(audio_block, gain_block, audio_block, len);
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}
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//release memory
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if(audio_level_dB_block)
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delete audio_level_dB_block;
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if(gain_block)
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delete gain_block;
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}
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// Here's the method that estimates the level of the audio (in dB)
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// It squares the signal and low-pass filters to get a time-averaged
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// signal power. It then
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void calcAudioLevel_dB(float *wav_block, float *level_dB_block, uint16_t len) {
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// calculate the instantaneous signal power (square the signal)
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float* wav_pow_block=new float[len];
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if(!wav_pow_block)
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{
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printf("Cannot allocate memory for \"wav_pow_block\" - stopping\n");
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while(1);
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}
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//arm_copy_f32(zeroblock_f32,wav_pow_block,len);
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arm_mult_f32(wav_block, wav_block, wav_pow_block, len);
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// low-pass filter and convert to dB
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float c1 = level_lp_const, c2 = 1.0f - c1; //prepare constants
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for (uint16_t i = 0; i < len; i++) {
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// first-order low-pass filter to get a running estimate of the average power
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wav_pow_block[i] = c1*prev_level_lp_pow + c2*wav_pow_block[i];
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// save the state of the first-order low-pass filter
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prev_level_lp_pow = wav_pow_block[i];
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//now convert the signal power to dB (but not yet multiplied by 10.0)
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level_dB_block[i] = log10f_approx(wav_pow_block[i]);
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}
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//limit the amount that the state of the smoothing filter can go toward negative infinity
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if (prev_level_lp_pow < (1.0E-13)) prev_level_lp_pow = 1.0E-13; //never go less than -130 dBFS
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//scale the wav_pow_block by 10.0 to complete the conversion to dB
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arm_scale_f32(level_dB_block, 10.0f, level_dB_block, len); //use ARM DSP for speed!
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//release memory and return
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if(wav_pow_block)
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delete wav_pow_block;
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return; //output is passed through level_dB_block
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}
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//This method computes the desired gain from the compressor, given an estimate
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//of the signal level (in dB)
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void calcGain(float *audio_level_dB_block, float *gain_block,uint16_t len) {
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//first, calculate the instantaneous target gain based on the compression ratio
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float* inst_targ_gain_dB_block=new float[len];
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if(!inst_targ_gain_dB_block)
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{
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printf("Cannot allocate memory for \"inst_targ_gain_dB_block\" - stopping\n");
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while(1);
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}
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//arm_copy_f32(zeroblock_f32,inst_targ_gain_dB_block,len);
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calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block,len);
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//second, smooth in time (attack and release) by stepping through each sample
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float *gain_dB_block = new float[len];
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if(!gain_dB_block)
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{
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printf("Cannot allocate memory for \"gain_dB_block\" - stopping\n");
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while(1);
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}
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//arm_copy_f32(zeroblock_f32,gain_dB_block,len);
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calcSmoothedGain_dB(inst_targ_gain_dB_block,gain_dB_block, len);
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//finally, convert from dB to linear gain: gain = 10^(gain_dB/20); (ie this takes care of the sqrt, too!)
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arm_scale_f32(gain_dB_block, 1.0f/20.0f, gain_dB_block, len); //divide by 20
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for (uint16_t i = 0; i < len; i++) gain_block[i] = pow10f(gain_dB_block[i]); //do the 10^(x)
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//release memory and return
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if(inst_targ_gain_dB_block)
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delete inst_targ_gain_dB_block;
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if(gain_dB_block)
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delete gain_dB_block;
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return; //output is passed through gain_block
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}
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//Compute the instantaneous desired gain, including the compression ratio and
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//threshold for where the comrpession kicks in
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void calcInstantaneousTargetGain(float *audio_level_dB_block, float *inst_targ_gain_dB_block, uint16_t len) {
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// how much are we above the compression threshold?
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float* above_thresh_dB_block=new float[len];
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if(!above_thresh_dB_block)
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{
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printf("Cannot allocate memory for \"above_thresh_dB_block\" - stopping\n");
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while(1);
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}
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//arm_copy_f32(zeroblock_f32,above_thresh_dB_block,len);
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arm_offset_f32(audio_level_dB_block, //CMSIS DSP for "add a constant value to all elements"
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-thresh_dBFS, //this is the value to be added
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above_thresh_dB_block, //this is the output
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len);
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// scale by the compression ratio...this is what the output level should be (this is our target level)
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arm_scale_f32(above_thresh_dB_block, //CMSIS DSP for "multiply all elements by a constant value"
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1.0f / comp_ratio, //this is the value to be multiplied
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inst_targ_gain_dB_block, //this is the output
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len);
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// compute the instantaneous gain...which is the difference between the target level and the original level
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arm_sub_f32(inst_targ_gain_dB_block, //CMSIS DSP for "subtract two vectors element-by-element"
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above_thresh_dB_block, //this is the vector to be subtracted
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inst_targ_gain_dB_block, //this is the output
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len);
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// limit the target gain to attenuation only (this part of the compressor should not make things louder!)
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for (uint16_t i=0; i < len; i++) {
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if (inst_targ_gain_dB_block[i] > 0.0f) inst_targ_gain_dB_block[i] = 0.0f;
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}
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// release memory before returning
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if(above_thresh_dB_block)
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delete above_thresh_dB_block;
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return; //output is passed through inst_targ_gain_dB_block
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}
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//this method applies the "attack" and "release" constants to smooth the
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//target gain level through time.
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void calcSmoothedGain_dB(float *inst_targ_gain_dB_block, float *gain_dB_block, uint16_t len) {
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float gain_dB;
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float one_minus_attack_const = 1.0f - attack_const;
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float one_minus_release_const = 1.0f - release_const;
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for (uint16_t i = 0; i < len; i++) {
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gain_dB = inst_targ_gain_dB_block[i];
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//smooth the gain using the attack or release constants
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if (gain_dB < prev_gain_dB) { //are we in the attack phase?
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gain_dB_block[i] = attack_const*prev_gain_dB + one_minus_attack_const*gain_dB;
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} else { //or, we're in the release phase
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gain_dB_block[i] = release_const*prev_gain_dB + one_minus_release_const*gain_dB;
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}
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//save value for the next time through this loop
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prev_gain_dB = gain_dB_block[i];
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}
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return; //the output here is gain_block
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}
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//methods to set parameters of this module
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void resetStates(void) {
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prev_level_lp_pow = 1.0f;
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prev_gain_dB = 0.0f;
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//initialize the HP filter. (This also resets the filter states,)
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arm_biquad_cascade_df1_init_f32(&hp_filt_struct, hp_nstages, hp_coeff, hp_state);
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}
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void setPreGain(float g) { pre_gain = g; }
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void setPreGain_dB(float gain_dB) { setPreGain(pow(10.0, gain_dB / 20.0)); }
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void setCompressionRatio(float cr) {
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comp_ratio = std::max(0.001f, cr); //limit to positive values
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updateThresholdAndCompRatioConstants();
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}
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void setAttack_sec(float a, float fs_Hz) {
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attack_sec = a;
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attack_const = expf(-1.0f / (attack_sec * fs_Hz)); //expf() is much faster than exp()
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//also update the time constant for the envelope extraction
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setLevelTimeConst_sec(std::min(attack_sec,release_sec) / 5.0, fs_Hz); //make the level time-constant one-fifth the gain time constants
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}
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void setRelease_sec(float r, float fs_Hz) {
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release_sec = r;
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release_const = expf(-1.0f / (release_sec * fs_Hz)); //expf() is much faster than exp()
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//also update the time constant for the envelope extraction
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setLevelTimeConst_sec(std::min(attack_sec,release_sec) / 5.0, fs_Hz); //make the level time-constant one-fifth the gain time constants
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}
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void setLevelTimeConst_sec(float t_sec, float fs_Hz) {
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const float min_t_sec = 0.002f; //this is the minimum allowed value
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level_lp_sec = std::max(min_t_sec,t_sec);
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level_lp_const = expf(-1.0f / (level_lp_sec * fs_Hz)); //expf() is much faster than exp()
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}
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void setThresh_dBFS(float val) {
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thresh_dBFS = val;
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setThreshPow(pow(10.0, thresh_dBFS / 10.0));
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}
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void enableHPFilter(boolean flag) { use_HP_prefilter = flag; };
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//methods to return information about this module
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float getPreGain_dB(void) { return 20.0 * log10f_approx(pre_gain); }
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float getAttack_sec(void) { return attack_sec; }
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float getRelease_sec(void) { return release_sec; }
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float getLevelTimeConst_sec(void) { return level_lp_sec; }
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float getThresh_dBFS(void) { return thresh_dBFS; }
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float getCompressionRatio(void) { return comp_ratio; }
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float getCurrentLevel_dBFS(void) { return 10.0* log10f_approx(prev_level_lp_pow); }
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float getCurrentGain_dB(void) { return prev_gain_dB; }
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void setHPFilterCoeff_N2IIR_Matlab(float b[], float a[]){
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//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
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//Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
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hp_coeff[0] = b[0]; hp_coeff[1] = b[1]; hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
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hp_coeff[3] = -a[1]; hp_coeff[4] = -a[2]; //the DSP needs the "a" terms to have opposite sign vs Matlab
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}
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private:
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//state-related variables
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float *inputQueueArray_f32[1]; //memory pointer for the input to this module
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float prev_level_lp_pow = 1.0;
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float prev_gain_dB = 0.0; //last gain^2 used
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//HP filter state-related variables
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arm_biquad_casd_df1_inst_f32 hp_filt_struct;
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static const uint8_t hp_nstages = 1;
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float hp_coeff[5 * hp_nstages] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
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float hp_state[4 * hp_nstages];
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void setHPFilterCoeff(void) {
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//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
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//Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
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float b[] = {9.979871156751189e-01, -1.995974231350238e+00, 9.979871156751189e-01}; //from Matlab
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float a[] = { 1.000000000000000e+00, -1.995970179642828e+00, 9.959782830576472e-01}; //from Matlab
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setHPFilterCoeff_N2IIR_Matlab(b, a);
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//hp_coeff[0] = b[0]; hp_coeff[1] = b[1]; hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
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//hp_coeff[3] = -a[1]; hp_coeff[4] = -a[2]; //the DSP needs the "a" terms to have opposite sign vs Matlab
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}
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//private parameters related to gain calculation
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float attack_const, release_const, level_lp_const; //used in calcGain(). set by setAttack_sec() and setRelease_sec();
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float comp_ratio_const, thresh_pow_FS_wCR; //used in calcGain(); set in updateThresholdAndCompRatioConstants()
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void updateThresholdAndCompRatioConstants(void) {
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comp_ratio_const = 1.0f-(1.0f / comp_ratio);
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thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const);
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}
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//settings
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float attack_sec, release_sec, level_lp_sec;
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float thresh_dBFS = 0.0; //threshold for compression, relative to digital full scale
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float thresh_pow_FS = 1.0f; //same as above, but not in dB
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void setThreshPow(float t_pow) {
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thresh_pow_FS = t_pow;
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updateThresholdAndCompRatioConstants();
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}
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float comp_ratio = 1.0; //compression ratio
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float pre_gain = -1.0; //gain to apply before the compression. negative value disables
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boolean use_HP_prefilter;
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// Accelerate the powf(10.0,x) function
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static float pow10f(float x) {
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//return powf(10.0f,x) //standard, but slower
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return expf(2.302585092994f*x); //faster: exp(log(10.0f)*x)
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}
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// Accelerate the log10f(x) function?
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static float log10f_approx(float x) {
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//return log10f(x); //standard, but slower
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return log2f_approx(x)*0.3010299956639812f; //faster: log2(x)/log2(10)
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}
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/* ----------------------------------------------------------------------
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** Fast approximation to the log2() function. It uses a two step
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** process. First, it decomposes the floating-point number into
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** a fractional component F and an exponent E. The fraction component
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** is used in a polynomial approximation and then the exponent added
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** to the result. A 3rd order polynomial is used and the result
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** when computing db20() is accurate to 7.984884e-003 dB.
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** ------------------------------------------------------------------- */
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//https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
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//float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
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static float log2f_approx(float X) {
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//float *C = &log2f_approx_coeff[0];
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float Y;
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float F;
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int32_t E;
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// This is the approximation to log2()
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F = frexpf(fabsf(X), &E);
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// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
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//Y = *C++;
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Y = 1.23149591368684f;
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Y *= F;
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//Y += (*C++);
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Y += -4.11852516267426f;
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Y *= F;
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//Y += (*C++);
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Y += 6.02197014179219f;
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Y *= F;
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//Y += (*C++);
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Y += -3.13396450166353f;
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Y += E;
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return(Y);
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}
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};
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#else
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#warning USING TEENSYDUINO SO INTERNAL COMPRESSOR IS DISABLED!
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#endif // TEENSYDUINO
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#endif
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