Daniel PEACOCK wrote:
Hi,
When ever you are processing audio in a way that means there is a chance
for a sample to exceed the range of signed short, then you need to
temporarily convert the samples into some other format (e.g signed longs or
floats). After processing is completed, you should clamp the values into
the allowable range (-32768 to 32767) and convert the samples back to
signed shorts. This may introduce clipping, but that is better than the
alternative of overflows where the most significant bits of the magnitude
are lost.
Dan
Creative Labs (UK) Ltd.
Fruskus
Hi, this is the first time I post here.
First of all, I'd like to thank all of you, because this forum has been
really helpful to me.
I'm doing a real time filtering application. So I'm using the capturing and
playing with queue routine. But when I modify the buffer, values may exceed
the short max values.
Does anyone know how to normalizate buffers in realtime? I've tried to use
a
normalization loop, but I think this only work for a whole file, not for
streaming.
Thanks in advance.
Thanks Dan, that was what I was doing, and it works. But I was just wondering If there was a better way to do this, and avoid the variations between the differents buffers signal level.
