Re: AudioConverterConvertBuffer & sampleRate
Re: AudioConverterConvertBuffer & sampleRate
- Subject: Re: AudioConverterConvertBuffer & sampleRate
- From: Christof Faller <email@hidden>
- Date: Wed, 22 Jan 2003 10:59:10 +0200
For sampling-rate conversion you need to use:
AudioConverterFillBuffer
This is a pulling data model. You specify how many samples you need at
the output, and your input-samples-providing-function then is called as
many times as possible.
I couple of days ago I posted on this list a source code snipped for
re-sampling the audio output. This is quite straight forward (see
posting).
But for re-sampling audio input, this is a bit more tricky, one needs
to keep track of number of samples. Let me know if you are trying to
re-sample the input, and I can forward you some example code.
Chris
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