How to avoid audio sampling 16-bit clipping after balance? - audio, balance, bit, Cut, how to avoid, sampling, wave

I have samples from ffmpeg, usually it is a 16-bit sample (short type), I used an iir bandpass filter with dbGain, as described here, after filtering , I sometimes get a short overflow, and the result is that some noise calculated sample values come out from 32767 / -32767, is there a way to escape audio pcm sample editing. May there be any methods?

I searched on Google but did not find any useful examples?

UPDATE

When I convert the transfer function calculation result to an integer and check for overflow, noise still occurs::

 int result = A1 * ((int) Rx) + A2 * ((int) Rxx) + A3 * ((int) Rxxx)
-B1 * ((int) Ryy)-B2 * ((int) Ryyy );
if (result> 32767)
 result = 32767;
if (result <-32700)
 result = -32700;
y = (short) result ;

16-bit PCM sampling must be within the range of [-32768 .. 32767]. If you add Mathematics (biquad filter in your case) is applied to the input signal, there is no guarantee that the output stays within that range, if you apply positive gain, this is an inevitable result.

< p>Since hitting the boundary of the range with this processing is a natural side effect, you should use one of these methods to deal with it (the list should not be complete):

>Make sure your input signal is sufficient Quiet and/or shift the value a few places to the right to provide margin for large values on the output>Use higher number of bits for the output signal, such as 24-bit PCM>Use floating-point PCM output signal to outside the PCM sampling range Loss of accuracy

I have samples from ffmpeg, usually it is a 16-bit sample (short type), I used iir bandpass filter with dbGain, as here As mentioned, after filtering, I sometimes get a short overflow. The result is that some noise calculated sample values come out from 32767 / -32767. Is there a way to escape audio pcm sample editing. Is there any way?

I searched on Google but did not find any useful examples?

UPDATE

When I convert the transfer function calculation result to an integer and check for overflow, noise still occurs::

 int result = A1 * ((int) Rx) + A2 * ((int) Rxx) + A3 * ((int) Rxxx)
-B1 * ((int) Ryy)-B2 * ((int) Ryyy );
if (result> 32767)
 result = 32767;
if (result <-32700)
 result = -32700;
y = (short) result ;

The 16-bit PCM samples must be in the range of [-32768 .. 32767]. If you use math (in your case, biquad filter If you apply a positive gain, this is an inevitable result.

Because of the use of this processing, the hit range boundary is one This is a natural side effect, so you should use one of these methods to deal with it (the list should not be complete):

>Make sure your input signal is quiet enough and/or shift the value a few places to the right, In order to provide margin for large values on the output>Use a higher number of bits for the output signal, such as 24-bit PCM>Use floating-point PCM to output the signal to the loss of precision outside the PCM sampling range

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