On Mar 30, 2023, at 10:43 AM, Evan Hand <elhandjr@...> wrote:

Hi Michael,

I am not sure what you mean by "Digita SSB."

There is the FreeDV effort which is open-source digital voice software.
The Hermes Light 2 is available from Makerfabs. It uses an FPGA device to capture the RF directly and a PC to process the I/Q audio into the audio voice range.
And then, there are multiple combinations of superhet to convert to a lower frequency to allow the use of a less costly DAC/ADC chip that uses a microprocessor to do the signal processing.

There is also the (Tr)uSDX effort that uses the QCX as a base platform to create the SSB signal using software in the MCP to modulate both the frequency and the amplitude to generate the SSB analog signal.

Here are some of the links:
https://freedv.org/freedv-specification/
https://www.makerfabs.com/hermes-lite-2.html
https://www.hfsignals.com/index.php/sbitx/
http://www.4sqrp.com/T41main.php
https://dl2man.de/

There is also a book by the developers of the T41 on Amazon:
https://www.amazon.com/Software-Defined-Radio-Transceiver-Construction/dp/B09WYP1ST8/

73
Evan
AC9TU

Obviously, I am not explaining the method properly. An audio signal from a micrphone has the voice frequencies from, say 100Hz to 2700 Hz, coming out to a BOX. We want this to come out of the BOX at, say on 20 meters at 14.1001 to 14.1027 MHz, just the upper sideband. The BOX looks at slices of the incoming audio signal in VERY narrow slices, determines the audio frequency and the amplitude of each VERY narrow slice and does the following things. It tells the Direct Digital Synthesis chip to generate a signal at that SSB frequency AND at that amplitude, directly at 20 meters. Now , of course, we tailor the actual amplitude to the wattage out that we require to send out a Upper Sideband Signal.  Since this has to be done at nanosecond intervals to capture the audio and convert, and since we probably need a DDS chip with an add on A/D capability, we are talking at much higher speeds than the normal DDS chips that I have seen so far. Other than the incoming microphone audio, everything is done digitally, except for the final D/A conversion of each slice at the output. No mixing, heterodyning, bandwidth filtering, phasing, frequency conversion as in 'normal' SSB transmitters.
If I seem to be talking foolishness, then my mind is going - possibly true. But I see this as a viable way of making a totally compatible SSB signal able to be used with our existing radios, NOT as some kind of another Digital Voice Mode. It would certainly require a computer chip, probably 32 bit bus, running at high speed do do all this. I don't know if we have DDS chips that fast yet, but I'm sure it won't be long. I won't beat this horse any longer but I hope to see pocket sized HF SSB boxes someday soon. I also apologize if this concept  doesn't make sense, but it's the best I can explain it.
73 - Mike Pupeza VE3EQP .....>

_._,_._,_

This is an excerpt from the uSDX project on Github.

uSDX is a simple and experimental (Class-E driven) SSB and CW SDR transceiver. It can be used to make QRP SSB contacts, or (in combination with a PC) used for the digital modes such as FT8, JS8, FT4. It can be fully-continuous tuned through bands 80m-10m in the LSB/USB-modes with a 2400Hz bandwidth has up to 5W PEP SSB output and features a software-based full Break-In VOX for fast RX/TX switching in voice and digital operations.

The SSB transmit-stage is implemented entirely in digital and software-based manner: at the heart the ATMEGA328P is sampling the input-audio and reconstructing a SSB-signal by controlling the SI5351 PLL phase (through tiny frequency changes over 800kbit/s I2C) and controlling the PA Power (through PWM on the key-shaping circuit). In this way a highly power-efficient class-E driven SSB-signal can be realized; a PWM driven class-E design keeps the SSB transceiver simple, tiny, cool, power-efficient and low-cost (ie. no need for power-inefficient and complex linear amplifier with bulky heat-sink as often is seen in SSB transceivers).

Steve KY4GX.

Glory to God in the highest, and on earth peace, good will toward men. Luke 2:14

On Friday, March 31, 2023 at 01:12:14 AM EDT, Michael Pupeza <mpupeza@...> wrote:

Obviously, I am not explaining the method properly. An audio signal from a micrphone has the voice frequencies from, say 100Hz to 2700 Hz, coming out to a BOX. We want this to come out of the BOX at, say on 20 meters at 14.1001 to 14.1027 MHz, just the upper sideband. The BOX looks at slices of the incoming audio signal in VERY narrow slices, determines the audio frequency and the amplitude of each VERY narrow slice and does the following things. It tells the Direct Digital Synthesis chip to generate a signal at that SSB frequency AND at that amplitude, directly at 20 meters. Now , of course, we tailor the actual amplitude to the wattage out that we require to send out a Upper Sideband Signal.  Since this has to be done at nanosecond intervals to capture the audio and convert, and since we probably need a DDS chip with an add on A/D capability, we are talking at much higher speeds than the normal DDS chips that I have seen so far. Other than the incoming microphone audio, everything is done digitally, except for the final D/A conversion of each slice at the output. No mixing, heterodyning, bandwidth filtering, phasing, frequency conversion as in 'normal' SSB transmitters.
If I seem to be talking foolishness, then my mind is going - possibly true. But I see this as a viable way of making a totally compatible SSB signal able to be used with our existing radios, NOT as some kind of another Digital Voice Mode. It would certainly require a computer chip, probably 32 bit bus, running at high speed do do all this. I don't know if we have DDS chips that fast yet, but I'm sure it won't be long. I won't beat this horse any longer but I hope to see pocket sized HF SSB boxes someday soon. I also apologize if this concept  doesn't make sense, but it's the best I can explain it.
73 - Mike Pupeza VE3EQP .....>