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Hello,

I’m just playing with the dsp_custom example, and was curious, how do you determine which samples you are currently editing in the inbuffer, i.e. I want to know the range x_i -> x_i+n where n is the length of the buffer.

In case it makes no sense why I want to do that, it is because I’m processing speech, and already know which segments of the speech are voiced and unvoiced and thus want to react differently depending on which segment of the speech is being processed.

Many thanks,

Chris.

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A DSP doesn’t really know about what data is being fed to it, it only knows that data is flowing through and you should do some kind of processing on it. As a DSP can be anywhere in the DSP network, positions don’t always relate to a particular sound, it may be some arbitrary submix.

You can always provide any information you know yourself to the callback though, for instance if you know a DSP will only be active when the sound is playing, you can keep track of how many samples have passed through. Be aware that DSP samples are at the system rate, usually 48KHz, so if this doesn’t match your source wave you will need to do some calculation to work out where you are.

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