Audio cuts out - overflowing floats?

Software Development
14

Ok so I managed to write an envelope follower class, based on this:
http://sam-koblenski.blogspot.com/2015/10/everyday-dsp-for-programmers-signal.html

#ifndef ENV_FOLLOW_H
#define ENV_FOLLOW_H
#include <math.h>

namespace daisysp
{

class EnvFollow
{
    private:
        
    float avg;      //exp average of input
    float pos_sample;   //positive sample
    float sample_noDC;  //no DC sample
    float avg_env;  //average envelope
    float w;        //weighting
    float w_env;    //envelope weighting

    public:

    EnvFollow() //default constructor
    {
        avg = 0.0f;      //exp average of input
        pos_sample = 0.0f;   //positive sample
        avg_env = 0.0f;  //average envelope
        w = 0.0001f;        //weighting
        w_env = 0.0001f;    //envelope weighting
        sample_noDC = 0.0f;
    }  
    ~EnvFollow() {}

    float GetEnv(float sample)
    {
        //remove average DC offset:
        avg = (w * sample) + ((1-w) * avg);
        sample_noDC = sample - avg;

        //take absolute
        pos_sample = fabsf(sample_noDC);

        //remove ripple
        avg_env = (w_env * pos_sample) + ((1-w_env) * avg_env);

        return avg_env;
    }
};

} // namespace daisysp
#endif

This works pretty well, perhaps I could tweak the weighting values a little to improve the response, but it’s not bad. I then output this on the Daisy DAC, so I could view it on my scope as a DC voltage. I noticed that the range of my envelope follower float doesn’t go much above a value of 1.0. I know this won’t pick up all the peaks as it is averaging the signal, but shouldn’t this be in the ballpark of the magnitude of the audio signal? I was surprised, because I assumed the floats would have a maximum value of around 3 *10^38!

Anyhow, I managed to implement an inverse feedback using the envelope follower signal, and this works, managing the runaway feedback effectively. However if I put a really hot input signal in I can still get the audio to cutout.

What would the best way to hard limit the audio signal? I tried this:

out[i] = (mixL > HardLimit) ? HardLimit : mixL;

But still the audio cuts out. I used a value of 0.7f for the HardLimit.

Any tips would be much appreciated!

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