LabVIEW uses integer PCM data for the sound data instead of a floating-point type between -1 and 1, so you'll have to convert numbers in the range [-1, +1] to [0, 255] for 8-bit or [-32768, 32767] for 16-bit sample size. (For example, the second case you can multiply by 32767 and then convert to I16.)
You can modify the Snd Write Wave File to include the new sampling rate that you want. Just changing the area near "rate" should automatically work, but I haven't tested it.
By the way, many sound cards do not support more than the frequencies mentioned, so even if you can create a WAV file, it doesn't mean your sound card can play it at that exact frequency.
Good Luck.