Re: Convolution Audio Unit how to?
Re: Convolution Audio Unit how to?
- Subject: Re: Convolution Audio Unit how to?
- From: Aran Mulholland <email@hidden>
- Date: Tue, 22 Nov 2011 10:44:40 +1100
this question might be better asked here -
http://music.columbia.edu/cmc/music-dsp/
On Tue, Nov 22, 2011 at 10:36 AM, tahome izwah <email@hidden> wrote:
> I'd recommend you check the literature for implementation of
> overlap-add and overlap-save processing. These are common
> implementations for algorithms like convolution that deal with
> edge-effects.
>
> HTH
> --th
>
> 2011/11/22 Mark Heath <email@hidden>:
>> Hi guys,
>>
>> I've spent the last week searching google for information on how to do this
>> but have not found anything.
>> Even searching this mailing list archive returns an error. So forgive me if
>> this has been asked before.
>>
>> I'm trying to implement an audio unit filter that behaves similar to a
>> convolution matrix filter (my background is in image processing so I may use
>> the wrong terminology)
>>
>> To calculate the new value of the current sample I need a window of samples
>> either side of current sample. (from the future and past)
>> I have implemented this (using AUBaseEffect) without processing the samples
>> near the edge of the supplied sample frame. However I am getting some
>> strange distortion that I could only attribute to not processing these edge
>> samples.
>>
>> So I'm looking for a correct implementation of a convolution filter.
>>
>> My original thought was to buffer the last frame and process the sample that
>> are near the edge this way but this has 2 problems;
>> 1) the first sample buffer passed into the filter must output less samples
>> than passed in and then I would need a tail to process the remaining
>> samples.
>> 2) as the filter is only receiving 1 channel, I do not know if my stored
>> sample buffer is from a different channel.
>>
>> Could anyone help?
>>
>> Thanks
>> Mark
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