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OpenAudio_ArduinoLibrary/AudioCalcEnvelope_F32.h

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4.0 KiB

/*
* AudioCalcEnvelope_F32
*
* Created: Chip Audette, Feb 2017
* Purpose: This module extracts the envelope of the audio signal.
* Derived From: Core envelope extraction algorithm is from "smooth_env"
* 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 _AudioCalcEnvelope_F32_h
#define _AudioCalcEnvelope_F32_h
#include <arm_math.h> //ARM DSP extensions. for speed!
#include <AudioStream_F32.h>
class AudioCalcEnvelope_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:calc_envelope
public:
//default constructor
AudioCalcEnvelope_F32(void) : AudioStream_F32(1, inputQueueArray_f32),
sample_rate_Hz(AUDIO_SAMPLE_RATE) { setDefaultValues(); };
AudioCalcEnvelope_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32),
sample_rate_Hz(settings.sample_rate_Hz) { 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();
if (!in_block) return;
//check format
if (in_block->fs_Hz != sample_rate_Hz) {
Serial.println("AudioComputeEnvelope_F32: *** WARNING ***: Data sample rate does not match expected.");
Serial.println("AudioComputeEnvelope_F32: Changing sample rate.");
setSampleRate_Hz(in_block->fs_Hz);
}
//prepare an output data block
audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
if (!out_block) return;
// //////////////////////add your processing here!
smooth_env(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);
}
//compute the smoothed signal envelope
//compute the envelope of the signal, not of the signal power)
void smooth_env(float x[], float y[], const int n) {
float xab, xpk;
int k;
// find envelope of x and return as y
//xpk = *ppk; // start with previous xpk
xpk = state_ppk;
for (k = 0; k < n; k++) {
xab = (x[k] >= 0.0f) ? x[k] : -x[k];
if (xab >= xpk) {
xpk = alfa * xpk + (1.f-alfa) * xab;
} else {
xpk = beta * xpk;
}
y[k] = xpk;
}
//*ppk = xpk; // save xpk for next time
state_ppk = xpk;
}
//convert time constants from seconds to unitless parameters, from CHAPRO, agc_prepare.c
void setAttackRelease_msec(const float atk_msec, const float rel_msec) {
given_attack_msec = atk_msec;
given_release_msec = rel_msec;
// convert ANSI attack & release times to filter time constants
float ansi_atk = 0.001f * atk_msec * sample_rate_Hz / 2.425f;
float ansi_rel = 0.001f * rel_msec * sample_rate_Hz / 1.782f;
alfa = (float) (ansi_atk / (1.0f + ansi_atk));
beta = (float) (ansi_rel / (10.f + ansi_rel));
}
void setDefaultValues(void) {
float32_t attack_msec = 5.0f;
float32_t release_msec = 50.0f;
setAttackRelease_msec(attack_msec, release_msec);
state_ppk = 0; //initialize
}
void setSampleRate_Hz(const float &fs_Hz) {
//change params that follow sample rate
sample_rate_Hz = fs_Hz;
}
void resetStates(void) { state_ppk = 1.0; }
float getCurrentLevel(void) { return state_ppk; }
private:
audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
float32_t sample_rate_Hz;
float32_t given_attack_msec, given_release_msec;
float32_t alfa, beta; //time constants, but in terms of samples, not seconds
float32_t state_ppk = 1.0f;
};
#endif