Grumpy_Mike:
It is going to be a bit nosier the faster it goes but with just speeding it up that much it should be OK. The thing is you are not switching between analogue channels and that is what takes the time. Letting the signal stabilise after the switch.
So what actually makes it faster? By ignoring the other inputs or skipping the "stabilization time"?
Hey... and by the way, I want to take this even further. I'm going to try to produce stereo audio; but first, clear my doubts:
- Due to small memory, I must cut down the sampling rate to 8 KHz, but shall I keep treating the sound as if it has a 16 KHz sampling rate?
- Correct me if I'm wrong. Digital stereo audio stores its samples by alternating all its channels. For instance: if the data is stored in an array, then the even-numbered indexes correspond to the samples of the left channel, and the odd-numbered indexes correspond to the samples of the right channel.
- By using analogWrite twice per loop (in order to update the duty cycle of both channels), will this create a sort of "phase shift" to the right channel? (if the left one is updated first)
Ok, that's all I need to know. Now I'm going to create the sound for testing, and also spoil you what it will be.
It will be a 1 kHz sine wave that goes throughout the channels! (around and around if looped)