Is there level limiting somewhere in the digital audio chain?
I’m working on a tower defence game with OpenAL, and there will be many towers firing at the same time, all using the same sound effect (at least for now). My concern is that triggering too many sounds at the same time could lead to speakers being blown, or at the very least a headache for the user.
It seems to me that there should be a level limiter either in software, or at the sound card hardware level to prevent fools like me from doing this.
Can anyone confirm this, and if so, tell me where this limiter exists? Thanks!
as it is, you’d be lucky if the signal were simply clipped in the software before it hit the DAC. you can easily implement this yourself. when i say ‘clipped’, i mean that amplitudes that exceed the maximum are set to the maximum, rather than allowed to overflow, wrap, or other less unpleasant results. clipping at this stage often sounds terrible, but the alternatives i mentioned sound worse.
there’s actually a big consideration to account for in this regard: are you rendering in float or int? if int, what is your headroom? with int, you could clip or overflow at practically any stage. with floating point, that will only happen as a serious design flaw. of course, you’ll often eventually have to convert to int, when interfacing with the DAC/hardware. the DAC will limit the output because it handles signals within very specific limits. at worst, this will be the equivalent of (sampled) white noise at 0 dB FS (which can be an awful experience for the user). so… the DAC serves as a limiter, although this stage only makes it significantly less probable that a signal will cause hearing or equipment damage.
at any rate, you can easily avoid this, and i recommend you do it yourself, since you’re directly in control of the number of sounds and their amplitude. at worst, samples with peaks of 0 dB FS will all converge at the same sample and you’ll need to multiply signal (the sum of shots) by the reciprocal of the sum of shots:
output[i] = numShots > 1 ? allThoseShots[i]*(1.0/numShots) : allThoseShots[i];that’s not ideal in many cases (because there will be an exaggerated ducking sound). so you should actually introduce a limiter in addition to overall reduction for the number of simultaneous shots. then you back off the shots signal by a lower factor since their peaks will not likely converge at the same point in time. a simple limiter with ~10 ms of lookahead should prevent you from doing something awful. it would also be a good idea to detect heavy limiting in debug mode, this catches upstream design issues.
in any case, you should definitely consider appropriate gain compensation your responsibility – you never want to clip the output dac. in fact, you want to leave some headroom (ref: intersample peaks).