Re: AudioConverter gives non normalized samples?!
Re: AudioConverter gives non normalized samples?!
- Subject: Re: AudioConverter gives non normalized samples?!
- From: Jeff Moore <email@hidden>
- Date: Tue, 27 Jan 2009 09:33:15 -0800
I would add that the main reason we use floating point samples is
precisely this situation. Generally, the only time you need to clip
your sample buffers is if you are going to convert them to another
sample format. Otherwise, just let the out-of-range samples ride. More
often than not, they come back into range later after more processing
has been applied. And even if they don't, you still don't need to clip
the samples yourself as the driver will take care of that for you in
the final mixing stage when it is converting the floating point
samples into the hardware's native format.
On Jan 27, 2009, at 9:05 AM, tahome izwah wrote:
This is just the result of the signal exceeding the allowed range
which would lead to clipping. This can happen at any time and there is
no way to prevent this, as users might set a gain value > 0 at some
point in the processing chain. You either need to deal with this in
your code or leave it to the output unit.
--th
2009/1/27 Mike Kluev <email@hidden>:
I have a situation when the result data from AudioConverter/
AUConverter
contain sample values outside [-1.0 .. +1.0 range, e.g. 1.000072 or
-1.000320. Source data (the data being converted) has all sample
values
normalized to (-1 .. +1) range. Is that a bug? Or am I supposed to
normalize the result data myself to fit the -1..+1 range? Or should
I just
leave them as they are? I have reproducible case for this.
--
Jeff Moore
Core Audio
Apple
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