pull/32/head
Holger Wirtz 5 years ago
parent cd750c4b37
commit d72caf668a
  1. 3
      MicroDexed.ino
  2. 2
      config.h
  3. 81
      dexed.cpp

@ -1814,6 +1814,9 @@ void _softRestart(void)
float pseudo_log_curve(float value)
{
// ideas from here: https://forum.pjrc.com/threads/56015-Changing-Volume-Levels-based-on-pot-wheel
const float _pseudo_log = 1048575 / (float)(1 << 20);
/*
#ifdef DEBUG

@ -84,7 +84,7 @@
//*************************************************************************************************
//* DEBUG OUTPUT SETTINGS
//*************************************************************************************************
#define DEBUG 1
//#define DEBUG 1
#define SERIAL_SPEED 230400
#define SHOW_XRUN 1
#define SHOW_CPU_LOAD_MSEC 5000

@ -149,7 +149,7 @@ void Dexed::getSamples(uint16_t n_samples, int16_t* buffer)
for (j = 0; j < _N_; ++j)
{
//sumbuf[i + j] += static_cast<float>(signed_saturate_rshift(audiobuf.get()[j] >> 5, 24, 9)) / 0x8000;
sumbuf[i + j] += signed_saturate_rshift(audiobuf.get()[j] >> 4, 24, 9) / 32768.0;
sumbuf[i + j] += signed_saturate_rshift(audiobuf.get()[j] >> 5, 24, 9) / 32768.0;
audiobuf.get()[j] = 0;
}
}
@ -800,3 +800,82 @@ void Dexed::setPortamentoMode(uint8_t portamento_mode, uint8_t portamento_glissa
controllers.refresh();
}
// https://www.musicdsp.org/en/latest/Effects/169-compressor.html#
void compress
(
float* wav_in, // signal
int n, // N samples
double threshold, // threshold (percents)
double slope, // slope angle (percents)
int sr, // sample rate (smp/sec)
double tla, // lookahead (ms)
double twnd, // window time (ms)
double tatt, // attack time (ms)
double trel // release time (ms)
)
{
typedef float stereodata[2];
stereodata* wav = (stereodata*) wav_in; // our stereo signal
threshold *= 0.01; // threshold to unity (0...1)
slope *= 0.01; // slope to unity
tla *= 1e-3; // lookahead time to seconds
twnd *= 1e-3; // window time to seconds
tatt *= 1e-3; // attack time to seconds
trel *= 1e-3; // release time to seconds
// attack and release "per sample decay"
double att = (tatt == 0.0) ? (0.0) : exp (-1.0 / (sr * tatt));
double rel = (trel == 0.0) ? (0.0) : exp (-1.0 / (sr * trel));
// envelope
double env = 0.0;
// sample offset to lookahead wnd start
int lhsmp = (int) (sr * tla);
// samples count in lookahead window
int nrms = (int) (sr * twnd);
// for each sample...
for (int i = 0; i < n; ++i)
{
// now compute RMS
double summ = 0;
// for each sample in window
for (int j = 0; j < nrms; ++j)
{
int lki = i + j + lhsmp;
double smp;
// if we in bounds of signal?
// if so, convert to mono
if (lki < n)
smp = 0.5 * wav[lki][0] + 0.5 * wav[lki][1];
else
smp = 0.0; // if we out of bounds we just get zero in smp
summ += smp * smp; // square em..
}
double rms = sqrt (summ / nrms); // root-mean-square
// dynamic selection: attack or release?
double theta = rms > env ? att : rel;
// smoothing with capacitor, envelope extraction...
// here be aware of pIV denormal numbers glitch
env = (1.0 - theta) * rms + theta * env;
// the very easy hard knee 1:N compressor
double gain = 1.0;
if (env > threshold)
gain = gain - (env - threshold) * slope;
// result - two hard kneed compressed channels...
float leftchannel = wav[i][0] * gain;
float rightchannel = wav[i][1] * gain;
}
}

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