Hi Darkseed!
This method is not going too far from original analog spectrum inversion scheme. There is mixer, inversion frequency generator and filters on input and output. But instead of constant inversion frequency we use permanently changing inversion frequency. Let say it is Frequency Modulated signal, or even Phase Modulated signal of Inversion Frequency.
On picture they used some linear intervals of phase changing. But maybe it is better to change phase of Inversion Frequency by some pseudo-aleatory number within some limits?
I do not know how to make conversion (multiplexing) of voice signal with inversion frequency by microprocessor, maybe it is symple? But I am not familiar with DSP methods, so I plan to make analog mixing, but to use processor for generating of inversion frequency signal. I will try to synthesize this frequency by making addition of phase in summator and taking sinus from a table and then produse sample by DAC. I know this method is named as DDS (direct digital synthesis) (once I have tried this method for RX heterodine, for this method is known as making signal with very low phase-noise). By changing adding phase we will make digital phase modulation. It is very symple to make such calculations by processor while frequency is very low up to 4 kHz. So, processor will calculate pseudo-aleatory number sequence, then adds this number to phase, then will add this phase to summator, takes sinus from table and sends it to DAC.
By using this digital method of synthesis we will obtain absolutely the same pseudo-noise signal in transmiter and receiver sides of channel, so we will decode signal with big precision. The only thing we have to do is narrowing of voice band and precise synchronization of coder and decoder.
What possible mistakes are hiding here? Or maybe there is DSP method to make the same conversion?
73! klug.