Single Sideband, or SSB, has been the predominant amateur radio voice mode for many decades now. It has bee traditionally generated by analogue means, generating a double sideband and filtering away the unwanted side, or generating 90 degree phase shifted quadrature signals and mixing them. More recent software-defined radios have taken this into the CPU, but here’s [Georg DG6RS] with another method. It uses SDR techniques and a combination of AM and FM to achieve polar modulation and generate SSB. He’s provided a fascinating in-depth technical explanation to help understand how it works.
The hardware is relatively straightforward; an SI5351 clock generator provides the reference for an ADF4351 PLL and VCO, which in turn feeds a PE4302 digital attenuator. It’s all driven from an STM32F103 microcontroller which handles the signal processing. Internally this means conventionally creating I and Q streams from the incoming audio, then an algorithm to generate the phase and amplitude for polar modulation. These are fed to the PLL and attenuator in turn for FM and AM modulation, and the result is SSB. It’s only suitable for narrow bandwidths, but it’s a novel and surprisingly simple deign.
We like being presented with new (to us at least) techniques, as it never pays to stand still. Meanwhile for more conventional designs, we’ve got you covered.
As Georg states himself, the modulation technique is based on usdx (and trusdx), which uses EER (envelope elimination and restauration). It allows using a highly efficient class e power amplifier for SSB on the trusdx.
I would be happy if I get in contact with a person who is interested to build a suitable nonlinear amplifier.
Why not go all the way and build a LINC amplifier (linear in compression)? You could even do the combining at 2.4 GHz, allowing the whole doubler chain and the power amplifier to be fully in compression (taking into account phase doubling as well).
sorry typo, 24 GHz.
Interesting Idea. However, still two chains of SHF doublers and amplifiers necessary. I have an idea to try to generate audible SSB without any AM, only phase modulation. This is because I noticed, when the AM part of the SSB generator did not work, the audio sounded distorted but still intelligible. Just the spectrum was awful. I do not have an idea how to use this effect to generate SSB with better sound quality but still without AM, only PM.
Cool appliance.
So far, all the implementations I have found are doing the FM part by (heroically!) updating the PLL registers as fast as possible over IIC. (The AM part is done either by the easy expedient of a variable attenuator before the PA or (more efficient) modulating the supply voltage of the PA, which can become especially efficient when said PA is class E.)
But has anyone actually implemented a VCXO-based FM modulator for this yet? It seems the logical thing to do, but I have not found one. Sure, it adds an analog step in an otherwise all-digital chain, but it easily increases bandwidth and decreases PLL complexity, jitter, and spur generation.
Could actually be done with the same ADF4351 board. If you modify the loop filter to be slow enough not the follow the modulation. Then add the modulation to vtune. I actually saw an FM-modulator built this way. Never did it myself, I wonder what the consquences are for phase noise, as these synthesizers typically rely heavily on the feedback loop to clean up the close-in phase noise.
You imply that you’re modulating the VFO itself, and keeping the loop slow enough that it doesn’t correct out your modulation input.
I’m thinking the opposite: frequency modulate the crystal reference clock, which would require the loop filter to be fast enough to follow the modulation.
Writing that makes me think that a bit of feed-forward to modulate the VFO at the same time as the XO reference might be the way to do it…
Nearly 8500 of the ‘B variants in stock at Mouser…
During the time of 9k6 packet radio this was widespread. A FM transmitter was modified to insert the 9k6 FSK modulation at the VCO (to be fast enough). And the FSK was inserted at the reference crystal as a second point, so the PLL did not try to get the frequency back to the nominal frequency. This was called “two point modulation”. You needed to get the FM deviation roughly the same for both entry points.
Only at the reference frequency would not have worked, because the PLL was not fast enough.
And the modulation only at the VCO would not have worked because the PLL was not slow enough ;-)
And there’s an Si5351 variant with a voltage input which has been used with some success to create FM.
Yeah, those versions were unobtainium last I looked, but that was a few years ago around the Skyworks transition. I have not checked recently.
What I did is to “modulate” the fractional counter of the frac-N-PLL. As I understand this is roughly the same effect. In both cases, the PLL loop filter needs to be fast enough to be able to follow the modulation.
Isn’t the easiest-cleanest way to get SSB (USB or LSB) by using DSP to impliment the Discrete Hilbert Transform?
As far I understand, the output of the Hilbert transformer is the I and Q component of the desired signal. The you still need to modulate this on an RF signal. You can use two mixers. Or, as I did, polar modulation.