Re: AudioQueueProcessingTap broken in 6.1? (was: Re: NewTimePitch Audio Unit Distorts Signal)
Re: AudioQueueProcessingTap broken in 6.1? (was: Re: NewTimePitch Audio Unit Distorts Signal)
- Subject: Re: AudioQueueProcessingTap broken in 6.1? (was: Re: NewTimePitch Audio Unit Distorts Signal)
- From: Kyle Sluder <email@hidden>
- Date: Tue, 05 Feb 2013 10:41:19 -0800
On Feb 5, 2013, at 10:25 AM, James McCartney <email@hidden> wrote:
> AUNewTimePitch is a format converter. Internally it is doing overlapped FFTs, so it pulls to get the data it needs. The extra buffer is to satisfy this extra pulling. Ultimately the answer is for us to release an AUNewPitch that is an effect. Same algorithm but since rate is not a parameter, the upstream pull size can be guaranteed to match the amount pulled for.
I understand that this ultimate solution is superior. But I have still have a question about the interim solution: is it possible to derive the requisite buffer size from some property exposed by the audio unit? For example, is it a known function of the _Latency property and the sample format?
Thanks for the info,
--Kyle Sluder
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