How can I handle processing live audio data?
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I am using the "audiorecorder" object and the "record" function to process live audio data using techniques like FFT, binning, and filtering. I am using a timer callback to process the data every so often. However, the time it takes to process a chunk of data is longer than the time it took to record that data, which means that over time the amount of data I need to process when I call "getaudiodata" or use a callback also increases. It gets to the point where the time it takes to process the data is longer than it takes for the next callback to trigger. Trying to use "recordblocking" or a time parameter does not help in fixing the issue either.
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도움말 센터 및 File Exchange에서 Audio and Video Data에 대해 자세히 알아보기
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