The DAC on the 6711 is extremely quick to hit each requested output value. If you output a value of 1 V to the ao channel it will hit 1 V and stay there until it receives the next update value. If the next value is 2 V, your output will not look like a smooth curve or a straight line from 1 to 2 V. Instead, it will look like a stair step where the step from 1 to 2 V will appear (looking at it on a scope) to happen 'immediately' (relative to the time between samples). This is nice if you are generating square waves and not so nice if you are generating a tone. It sounds like you want to generate 'smoother' signals. On the software side you could greatly oversample the waveform so that each step change in the waveform is minimized. For instance, for a full-scale (10 V pk), 1000 Hz sinusoid if you only have 10 samples per cycle (sample rate = 10 kHz) the maximum step change is ~6.3 V! If you have a 1000 samples per cycle (sample rate = 1 MHz), the maximum (digital) step change is ~63 mV. Oversampling in software also allows you to produce sharp square waves and more linear triangle waves. On the hardware side you could put an external lowpass filter on your output, or use a DSA output device (i.e. the PCI 4461) that has an embedded anti-imaging filter. A hardware solution will yield superior spectral purity for bandlimited signals such as a multitone or a tone, but it will also bandlimit your triangle and square wave (which might not be desired).
Doug
Enthusiast for LabVIEW, DAQmx, and Sound and Vibration