Easiest seems to be to route the two signals to pins and add them with a simple resistor network.
Bob's right, the easiest way is two filters and a resistive adder externally. Feed back into a PGA to get a buffered low-impedance output. You could put a cap on the resistor summing junction to filter out most of the sampling aliases.
IF . . . . you want zero external parts. Put one LPF in ASC10, ASD11, feed filter from digital input on P2. Put second LPF in ASC23, ASD22, feed filter from digital input on P2. Sum the outputs with an MDAC6 in ASC21; you'll have to set the B-input on block ASC21 in your code. Cool part this way is that you can scale the levels by changing the A and B caps. This results in levels that are dependent on power supply. If you want absolute levels, connect filter inputs to RefHi, then modulator bits to your digital signals. Set RefHi to something other than Vdd. You'll get the best result if the filters have the same column clock.
IF . . . . you are software-centric you can follow the method of the DTMF dialer user module and do with direct digital synthesis into a DAC. This is probably limited to a few kHz.
The limit on your output frequency is the slew rate of the analog column output buffer, roughly 40 kHz for rail to rail swing with no distortion caused by slew rate limiting.