Hi all, Just looking at the source code, it hit me. When in analogue mode, we run 16kHz sample rate, fair enough.
But in DV mode, we sample the ADC at 16kHz, downconvert to 8kHz, pass it through the modem and codec, then upconvert back to 16kHz for the DAC. We're dealing with voice frequencies, with SSB transmit bandwidths of less than 3kHz. Why not do the whole lot at 8kHz and save some CPU time? Maybe some rate-switching when in analogue or DV mode might be an option too, so we run ADC/DAC at full-rate in analogue (for higher fidelity), then switch to half-rate when we're doing DV. Regards, -- Stuart Longland (aka Redhatter, VK4MSL) I haven't lost my mind... ...it's backed up on a tape somewhere.
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