If the polarity is not preserved, then Mixbus will think there has been a "very long delay" from the output to the input, and this can result in nonsensical values from the calibration. Some devices internally flip the polarity between the input and output (which is an indication of lax testing, but probably not a problem in regular recording usage). It is very important that your polarity of the wiring between your device's output and input is preserved. The "calibrate audio" feature works by sending a complicated mix of low, mid, and high tones through your I/O device's output, and then using the frequency content of the tones to determine, with great accuracy, how long it takes for the signal to be sent & received by your device. Once measured, Mixbus should use this value to further refine the placement of recorded audio regions to match "what you heard". The "calibrate audio" feature (near the bottom of the Audio setup Window) is intended to measure these delays. When that happens, we have to measure the latency by sending a signal through the D/A, and returning it to an A/D. Sometimes the hardware latency introduced by your A/D and D/A hardware isn't reported correctly to Mixbus. If Mixbus is provided your hardware's inherent delays, the soundcard buffer size, and the latency of all plugins etc, then Mixbus should be able to place the audio correctly with "what you heard while you were playing". Particularly when you are adding new tracks to an existing recording, or "overdubbing" on parts of the existing recording, you want the timing to match the previously-recorded material. The value of this adjustment is the recording-offset latency, and Mixbus should manage this for you "automatically". When you finish a recording, Mixbus has to place the newly recorded region so it aligns correctly with the existing audio in the session. When you record audio into a computer, there are inherent delays caused by the conversion and buffering inside your audio system. Soundcard calibration + recording latency
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