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; }