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We have a karaoke app that records the user and later mixes the voice and playback in real time.
We are experiencing a synchronization problem when playing back the two streams.
It looks like when recording with FMODEX there is a 100 MS delay between the input and output signal of the sound card.
I put an example file of recording from the microphone recording channel.

http://staging.ksolo.com/plugintest/test.wav

You can see that there is a 100 MS of silence in the beginning before the actual signal of the microphone kicks in.
When I’m recording with MS sound recorder for example it looks different.
Is there anything that can be done about that? Is it dependent om buffer size definitions.

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It’s pretty much impossible to get instant playthrough, but using an audio interface like ASIO (or perhaps very small buffer sizes and a high process priority), you can probably get pretty close.

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Janus is right, you can achieve short latency between the two streams by using the smallest possible buffer sizes. ASIO enables this to be done, you have excellent examples of this using Cubase SX when you record an instrument : you can only have a reasonable latency if you setup your sound buffer to a small size.

If your sound card driver does not support ASIO, you can still look at http://www.asio4all.com/
to be able to test tiny buffer sizes.

Note that many cards perform properly with 15ms latencies.

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