uint32_t AUDMAudioFilterLimiter::fill(uint32_t max,float *output,AUD_Status *status)
{
uint32_t len,i;
shrink();
fillIncomingBuffer(status);
len=_tail-_head;
if(len>max) len=max;
if(len>=DELIM_WINDOW_SIZE) len=DELIM_WINDOW_SIZE-1;
// Count in full sample i.e. all channels
len=len-(len%_wavHeader.channels);
// Process..
if (mLastLevel == 0.0) {
int preSeed = mCircleSize;
if (preSeed > len)
preSeed = len;
for(i=0; i<preSeed; i++)
AvgCircle(_incomingBuffer[_head+i]);
}
for (i = 0; i < len; i++) {
Follow(_incomingBuffer[_head+i], &follow[i], i);
}
for (i = 0; i < len; i++) {
output[i] =DoCompression(_incomingBuffer[_head+i], follow[i]);
}
_head+=len;
return len;
}
void EffectCompressor::Follow(float *buffer, float *env, size_t len, float *previous, size_t previous_len)
{
/*
"Follow"ing algorithm by Roger B. Dannenberg, taken from
Nyquist. His description follows. -DMM
Description: this is a sophisticated envelope follower.
The input is an envelope, e.g. something produced with
the AVG function. The purpose of this function is to
generate a smooth envelope that is generally not less
than the input signal. In other words, we want to "ride"
the peaks of the signal with a smooth function. The
algorithm is as follows: keep a current output value
(called the "value"). The value is allowed to increase
by at most rise_factor and decrease by at most fall_factor.
Therefore, the next value should be between
value * rise_factor and value * fall_factor. If the input
is in this range, then the next value is simply the input.
If the input is less than value * fall_factor, then the
next value is just value * fall_factor, which will be greater
than the input signal. If the input is greater than value *
rise_factor, then we compute a rising envelope that meets
the input value by working bacwards in time, changing the
previous values to input / rise_factor, input / rise_factor^2,
input / rise_factor^3, etc. until this NEW envelope intersects
the previously computed values. There is only a limited buffer
in which we can work backwards, so if the NEW envelope does not
intersect the old one, then make yet another pass, this time
from the oldest buffered value forward, increasing on each
sample by rise_factor to produce a maximal envelope. This will
still be less than the input.
The value has a lower limit of floor to make sure value has a
reasonable positive value from which to begin an attack.
*/
int i;
double level,last;
if(!mUsePeak) {
// Update RMS sum directly from the circle buffer
// to avoid accumulation of rounding errors
FreshenCircle();
}
// First apply a peak detect with the requested decay rate
last = mLastLevel;
for(i=0; i<len; i++) {
if(mUsePeak)
level = fabs(buffer[i]);
else // use RMS
level = AvgCircle(buffer[i]);
// Don't increase gain when signal is continuously below the noise floor
if(level < mNoiseFloor) {
mNoiseCounter++;
} else {
mNoiseCounter = 0;
}
if(mNoiseCounter < 100) {
last *= mDecayFactor;
if(last < mThreshold)
last = mThreshold;
if(level > last)
last = level;
}
env[i] = last;
}
mLastLevel = last;
// Next do the same process in reverse direction to get the requested attack rate
last = mLastLevel;
for(i = len; i--;) {
last *= mAttackInverseFactor;
if(last < mThreshold)
last = mThreshold;
if(env[i] < last)
env[i] = last;
else
last = env[i];
}
if((previous != NULL) && (previous_len > 0)) {
// If the previous envelope was passed, propagate the rise back until we intersect
for(i = previous_len; i--;) {
last *= mAttackInverseFactor;
if(last < mThreshold)
last = mThreshold;
if(previous[i] < last)
previous[i] = last;
else // Intersected the previous envelope buffer, so we are finished
return;
}
// If we can't back up far enough, project the starting level forward
// until we intersect the desired envelope
last = previous[0];
for(i=1; i<previous_len; i++) {
last *= mAttackFactor;
if(previous[i] > last)
previous[i] = last;
else // Intersected the desired envelope, so we are finished
return;
}
// If we still didn't intersect, then continue ramp up into current buffer
for(i=0; i<len; i++) {
last *= mAttackFactor;
if(buffer[i] > last)
buffer[i] = last;
else // Finally got an intersect
return;
}
// If we still didn't intersect, then reset mLastLevel
mLastLevel = last;
}
}
void AUDMAudioFilterLimiter::Follow(float x, float *outEnv, int maxBack)
{
/*
"Follow"ing algorithm by Roger B. Dannenberg, taken from
Nyquist. His description follows. -DMM
Description: this is a sophisticated envelope follower.
The input is an envelope, e.g. something produced with
the AVG function. The purpose of this function is to
generate a smooth envelope that is generally not less
than the input signal. In other words, we want to "ride"
the peaks of the signal with a smooth function. The
algorithm is as follows: keep a current output value
(called the "value"). The value is allowed to increase
by at most rise_factor and decrease by at most fall_factor.
Therefore, the next value should be between
value * rise_factor and value * fall_factor. If the input
is in this range, then the next value is simply the input.
If the input is less than value * fall_factor, then the
next value is just value * fall_factor, which will be greater
than the input signal. If the input is greater than value *
rise_factor, then we compute a rising envelope that meets
the input value by working bacwards in time, changing the
previous values to input / rise_factor, input / rise_factor^2,
input / rise_factor^3, etc. until this new envelope intersects
the previously computed values. There is only a limited buffer
in which we can work backwards, so if the new envelope does not
intersect the old one, then make yet another pass, this time
from the oldest buffered value forward, increasing on each
sample by rise_factor to produce a maximal envelope. This will
still be less than the input.
The value has a lower limit of floor to make sure value has a
reasonable positive value from which to begin an attack.
*/
float level = AvgCircle(x);
float high = mLastLevel * mAttackFactor;
float low = mLastLevel * mDecayFactor;
if (low < _param.mFloor)
low = _param.mFloor;
if (level < low)
*outEnv = low;
else if (level < high)
*outEnv = level;
else {
// Backtrack
float attackInverse = 1.0 / mAttackFactor;
float temp = level * attackInverse;
int backtrack = 50;
if (backtrack > maxBack)
backtrack = maxBack;
float *ptr = &outEnv[-1];
int i;
bool ok = false;
for(i=0; i<backtrack-2; i++) {
if (*ptr < temp) {
*ptr-- = temp;
temp *= attackInverse;
}
else {
ok = true;
break;
}
}
if (!ok && backtrack>1 && (*ptr < temp)) {
temp = *ptr;
for (i = 0; i < backtrack-1; i++) {
ptr++;
temp *= mAttackFactor;
*ptr = temp;
}
}
else
*outEnv = level;
}
mLastLevel = *outEnv;
}
