Ex-ADIer here, we had a couple of copies kicking around in our office. Shortly after I joined I asked my team lead what an FFT is, he explained "it's a way to transform a signal from the time domain to the frequency domain..." and when he saw that my eyes had glazed over he pointed to this book and said "read that". It's really readable and an excellent intro to DSP
From the blog quoted above: "Advanced DSP can certainly get complicated with quadrature processing, Hilbert transforms, and cascaded infinite impulse response filters, but this series on DSP isn't going to get that complicated."
Gulp. I'm new to DSP, so I'm obviously not going to jump into "quadrature processing," but what would you say are the basic mathematical prerequisites for getting into this kind of thing? What sort of math should you know before you even start?
I'm afraid I don't know that info - I worked with the compiler toolchain and associated libraries so strictly speaking it probably wasn't even necessary to know the basics of signal processing. However it gets -complicated- interesting enough to be challenging, and whatever the next steps are this will at least put the necessary groundwork in place :-)