mirror of https://github.com/probonopd/MiniDexed
remove unused effect_compressor (#949)
The compressor of Synth_Dexed is used, so it is not necessary to link this into the binary.main
soyer
7 days ago
committed by
GitHub
parent
d750330eb2
commit
5cd9782a6a
@ -1,326 +0,0 @@ |
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/* 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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#include <algorithm> |
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#include <circle/logger.h> |
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#include <cstdlib> |
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#include "effect_compressor.h" |
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LOGMODULE ("compressor"); |
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Compressor::Compressor(const float32_t 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 Compressor::setDefaultValues(const float32_t 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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//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 Compressor::calcInstantaneousTargetGain(float32_t *audio_level_dB_block, float32_t *inst_targ_gain_dB_block, uint16_t len) |
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{ |
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// how much are we above the compression threshold?
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float32_t above_thresh_dB_block[len]; |
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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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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 Compressor::calcSmoothedGain_dB(float32_t *inst_targ_gain_dB_block, float32_t *gain_dB_block, uint16_t len) |
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{ |
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float32_t gain_dB; |
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float32_t one_minus_attack_const = 1.0f - attack_const; |
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float32_t 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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// 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 Compressor::calcAudioLevel_dB(float32_t *wav_block, float32_t *level_dB_block, uint16_t len) {
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// calculate the instantaneous signal power (square the signal)
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float32_t wav_pow_block[len]; |
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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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float32_t 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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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 Compressor::calcGain(float32_t *audio_level_dB_block, float32_t *gain_block,uint16_t len) |
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{
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//first, calculate the instantaneous target gain based on the compression ratio
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float32_t inst_targ_gain_dB_block[len]; |
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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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float32_t gain_dB_block[len]; |
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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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return; //output is passed through gain_block
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} |
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//here's the method that does all the work
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void Compressor::doCompression(float32_t *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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LOGERR("No audio_block available for Compressor!"); |
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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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float32_t audio_level_dB_block[len]; |
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//arm_copy_f32(zeroblock_f32,audio_level_dB_block,len);
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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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float32_t gain_block[len]; |
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//arm_copy_f32(zeroblock_f32,gain_block,len);
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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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//methods to set parameters of this module
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void Compressor::resetStates(void) |
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{ |
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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 Compressor::setPreGain(float32_t g) |
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{ |
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pre_gain = g; |
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} |
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void Compressor::setPreGain_dB(float32_t gain_dB) |
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{ |
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setPreGain(pow(10.0, gain_dB / 20.0)); |
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} |
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void Compressor::setCompressionRatio(float32_t cr) |
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{ |
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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 Compressor::setAttack_sec(float32_t a, float32_t fs_Hz) |
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{ |
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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 Compressor::setRelease_sec(float32_t r, float32_t fs_Hz) |
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{ |
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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 Compressor::setLevelTimeConst_sec(float32_t t_sec, float32_t fs_Hz) |
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{ |
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const float32_t 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 Compressor::setThresh_dBFS(float32_t val) |
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{
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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 Compressor::enableHPFilter(boolean flag) |
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{ |
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use_HP_prefilter = flag; |
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} |
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void Compressor::setHPFilterCoeff_N2IIR_Matlab(float32_t b[], float32_t a[]) |
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{ |
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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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void Compressor::setHPFilterCoeff(void) |
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{ |
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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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float32_t b[] = {9.979871156751189e-01, -1.995974231350238e+00, 9.979871156751189e-01}; //from Matlab
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float32_t 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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void Compressor::updateThresholdAndCompRatioConstants(void) |
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{ |
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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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void Compressor::setThreshPow(float32_t t_pow) |
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{
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thresh_pow_FS = t_pow; |
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updateThresholdAndCompRatioConstants(); |
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} |
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// Accelerate the powf(10.0,x) function
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static float32_t pow10f(float32_t x) |
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{ |
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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 float32_t log10f_approx(float32_t x) |
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{ |
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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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//float32_t log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
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static float32_t log2f_approx(float32_t X) |
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{ |
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//float32_t *C = &log2f_approx_coeff[0];
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float32_t Y; |
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float32_t F; |
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int 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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@ -1,84 +0,0 @@ |
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/* 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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#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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class Compressor |
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{ |
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public: |
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//constructor
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Compressor(const float32_t sample_rate_Hz); |
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void doCompression(float32_t *audio_block, uint16_t len); |
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void setDefaultValues(const float32_t sample_rate_Hz); |
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void setPreGain(float32_t g); |
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void setPreGain_dB(float32_t gain_dB); |
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void setCompressionRatio(float32_t cr); |
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void setAttack_sec(float32_t a, float32_t fs_Hz); |
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void setRelease_sec(float32_t r, float32_t fs_Hz); |
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void setLevelTimeConst_sec(float32_t t_sec, float32_t fs_Hz); |
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void setThresh_dBFS(float32_t val); |
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||||||
void enableHPFilter(boolean flag); |
|
||||||
float32_t getPreGain_dB(void); |
|
||||||
float32_t getAttack_sec(void); |
|
||||||
float32_t getRelease_sec(void); |
|
||||||
float32_t getLevelTimeConst_sec(void); |
|
||||||
float32_t getThresh_dBFS(void); |
|
||||||
float32_t getCompressionRatio(void); |
|
||||||
float32_t getCurrentLevel_dBFS(void); |
|
||||||
float32_t getCurrentGain_dB(void); |
|
||||||
|
|
||||||
protected: |
|
||||||
void calcAudioLevel_dB(float32_t *wav_block, float32_t *level_dB_block, uint16_t len); |
|
||||||
void calcGain(float32_t *audio_level_dB_block, float32_t *gain_block,uint16_t len); |
|
||||||
void calcInstantaneousTargetGain(float32_t *audio_level_dB_block, float32_t *inst_targ_gain_dB_block, uint16_t len); |
|
||||||
void calcSmoothedGain_dB(float32_t *inst_targ_gain_dB_block, float32_t *gain_dB_block, uint16_t len); |
|
||||||
void resetStates(void); |
|
||||||
void setHPFilterCoeff_N2IIR_Matlab(float32_t b[], float32_t a[]); |
|
||||||
|
|
||||||
//state-related variables
|
|
||||||
float32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
|
|
||||||
float32_t prev_level_lp_pow = 1.0; |
|
||||||
float32_t prev_gain_dB = 0.0; //last gain^2 used
|
|
||||||
|
|
||||||
//HP filter state-related variables
|
|
||||||
arm_biquad_casd_df1_inst_f32 hp_filt_struct; |
|
||||||
static const uint8_t hp_nstages = 1; |
|
||||||
float32_t hp_coeff[5 * hp_nstages] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
|
|
||||||
float32_t hp_state[4 * hp_nstages]; |
|
||||||
void setHPFilterCoeff(void); |
|
||||||
//private parameters related to gain calculation
|
|
||||||
float32_t attack_const, release_const, level_lp_const; //used in calcGain(). set by setAttack_sec() and setRelease_sec();
|
|
||||||
float32_t comp_ratio_const, thresh_pow_FS_wCR; //used in calcGain(); set in updateThresholdAndCompRatioConstants()
|
|
||||||
void updateThresholdAndCompRatioConstants(void); |
|
||||||
//settings
|
|
||||||
float32_t attack_sec, release_sec, level_lp_sec;
|
|
||||||
float32_t thresh_dBFS = 0.0; //threshold for compression, relative to digital full scale
|
|
||||||
float32_t thresh_pow_FS = 1.0f; //same as above, but not in dB
|
|
||||||
void setThreshPow(float32_t t_pow); |
|
||||||
float32_t comp_ratio = 1.0; //compression ratio
|
|
||||||
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(float32_t x); |
|
||||||
// Accelerate the log10f(x) function?
|
|
||||||
static float32_t log10f_approx(float32_t x); |
|
||||||
static float32_t log2f_approx(float32_t X); |
|
||||||
#endif |
|
||||||
Loading…
Reference in new issue