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 11:56:48 -0800
Yes. The float-to-int converters all use saturation instructions
during the conversion.
BTW, if you have out-of-range samples in the raw recording, after
analog-to-digital conversion, it is generally too late to do anything
about it. You need to watch your headroom on the signal feeding into
the ADC and keep it from clipping before it gets into the computer.
That's the only way to get a clean signal.
On Jan 27, 2009, at 9:55 AM, Mike Kluev wrote:
On Tue, 27 Jan 2009 09:33:15 Jeff Moore wrote:
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.
Does AudioConverter correctly clips overflown samples when it
converts from floats to shorts? I do some mic recording +
postprocessing, putting the result into PCM file (samples are
shorts) and noticed that the wave form is not "nice" due to
overflow on the said out-of-bounds samples. If I manually
clip sample values to the -1 .. +1 range the wave form is ok.
--
Jeff Moore
Core Audio
Apple
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