This type of encoder circuit is quite common, but the operation is perhaps often misunderstood. I have used an LM358, Dual OpAmp for the two input buffer stages, but even these could be eliminated and resistive filtering used. I used them since there are definite advantages to having them. By the way, the 10uf supply decoupling cap is not on the PCB. This is on the transmitter PCB, just after the 8v regulator. The A and B channel signals from the OpAmp buffers are each fed to a CD4016 (bilateral switch) to "chop" the audio up at 38kHz. The outputs of the two switches are passed to a simple resistive adder. The two switches need to be switched alternately, but the CD4060 oscillator divider only provides a single phase output. So I used IC2a switch to connect it's output to ground when turned ON. This effectively inverts the 38KHz, without any additional logic circuits. The two choppers are fed with two 180 degrees phase shifted clocks, so the 38kHz DSB sidebands are inverted for one channel, to get the difference signal. The two DSB signals being added, but the carriers (being of opposite phases) cancel. The two original channels are also passed through the switches, unaltered (other than chopped up a bit), but the chopping rate is higher than the highest modulating frequency, so it doesn't matter. The result is that the output of the two choppers (IC2b and IC2c) contains A+B and A-B(DSB) signals. It does NOT, however contain the needed 19kHz pilot tone. This comes from IC3 pin 14, and is passed through a simple RC filter to make it a (sort of) sine(ish) waveform. Everything is added in the resistive mixer, to form the composite stereo output signal. A single capacitor has been added, as a token filter, to limit the total output to under 60kHz. The CD4060 is a conventional oscillator, based on a 4860kHz crystal, divided down to 38kHz and 19kHz. As you can see, the circuit is quite crude, but it does give a very good, and surprisingly clean, stereo signal out. If I had continued with the original OpAmp design you would need an oudio spectrum analyser, or selective level meter to set up and balance the various parts of the circuit. This circuit is so crude that it needs no alignment. All you need to do is adjust RV1 for the output level needed for your FM transmitter. No audio filtering is included, but after investigating every audio device in my house (including the TV set) I could not find a need for an AF filter. The input levels are 1v Peak-to-peak, which is the standard "LINE OUT" level from nearly all HiFi systems and computers. It can even be driven from your Sony walkman, but you need to be a bit more careful with the input level. A simple voltage reading with an AC voltmeter is close enough. You want 300mV (0.3v) AC.