[Stk] FileWvIn question

greg kellum greg_kellum@hotmail.com
Sun, 04 Feb 2007 23:35:14 +0000


Hi Stephen,

Thanks for the advice.  I ended up multiplying the sample values by 1/32768 
and got the result I was looking for.

I did first try the chunking approach you suggested.  But I kept getting 
some strange clipped sample values during the second chunk.  I tried using 
different chunk sizes and chunk thresholds to try to correct the problem, 
but that usually caused the program to crash with a segmentation fault.  I 
have no idea why this chunking caused so many issues.  Maybe, Gary could 
take a look at it for a future release.  But in any case I found a 
work-around...

Best,
Greg


>From: Stephen Sinclair <stephen.sinclair@mail.mcgill.ca>
>To: greg kellum <greg_kellum@hotmail.com>
>CC: stk@ccrma.Stanford.EDU
>Subject: Re: [Stk] FileWvIn question
>Date: Sun, 04 Feb 2007 12:21:42 -0500
>
>It's probably one of 32768, 65536, or 2147483648 depending on your data 
>type. (16-bit or 32-bit, signed unsigned, etc).
>
>Unfortunately I think that through the FileWvIn interface there is no 
>function to provide the data type.  However, I _think_ that if you specify 
>to use normalization, and also to use "chunking" (that is, streaming from 
>disk instead of loading the whole file), then it will "normalize", but 
>without using the peak value.  In other words, the data stream will be 
>multiplied by 1/32768 for example, isntead of scaling the largest data 
>point to 1.  This is probably what you really want, since then the data 
>type is not important, and your relative amplitude will be unaffected.  I 
>haven't tested it, however..
>
>Steve
>
>
>greg kellum wrote:
>>Hi,
>>
>>I have been trying to use the classes FileWvIn and FileWvOut to test some 
>>signal processing code that I have been working on.  I have noticed that 
>>by default FileWvIn normalizes the sound file.  So, if I try to just make 
>>a copy of a sound file via FileWvIn and FileWvOut, the resulting sound 
>>file is quite a bit louder than the source file.  If I however turn off 
>>the normalization, FileWvIn outputs values that don't fall between -1 and 
>>1 but are instead in the thousands.  These values obviously need to be 
>>scaled by some factor to get useable audio output.  So my question is: by 
>>what factor would these values have to be scaled in order to make an 
>>identical copy of the original sound file?
>>
>>Thanks,
>>Greg
>>
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