> --- You can peruse (and that is the correct word.. Lots of detailed  
> examination required) the equivalent of the schematics: the source  
> code for PowerSDR and dttsp.

Certainly one could do that. I suspect that it would be a long climb  
up the learning curve for a small amount of information. A logical  
block diagram would probably be useful. whether one performs functions  
in DSP or in analog hardware, one still performs those functions.

> There is no "power control" per-se inside the physical radio.

Ah. I thought that there might be a way to vary the gain of an analog  
driver stage so that you are operating at an optimum level of output  
from the DAC and yet are still able to change the gain of the transmit  
chain. After all, there is not an insignificant amount of analog  
hardware in the F5K itself.

> The output of the QSE (vector mod) gets amplified in an essentially  
> fixed gain chain, so the thing that determines output power level is  
> the audio level fed to the modulator, which in turn is probably  
> entirely determined by the digital data stream. (I don't recall  
> seeing any code to change the DAC gain, but, OTOH, I haven't looked  
> for it)
>
>> In more traditional SSB radios one has no choice but to control  
>> transmit power level by varying the mic (low-level) gain. Gain  
>> settings for the rest of the stages are fixed. With the Elecraft K2  
>> we have, I found it works better to optimize the baseband signal  
>> level (audio) and then use the power output control to vary the  
>> transmitter power.
>>
>> So, how does the F5K work? I would naturally lean toward fixing the  
>> baseband signal level and then use the transmitter power level  
>> control to set the TX power output but now you are saying that we  
>> should manage power using the mic gain.
>>
> -----  Frank or Bob would be the ideal folks to answer the details,  
> but, I suspect it doesn't make much difference.  All the DSP is  
> floating point, so it's not like you have the traditional analog  
> problem of "saturating an amplifier".

Ah, but you do. Floating point or not, there is some point at which  
you clip the DAC on one end and some point where noise becomes a  
problem at the other. FP simply makes the intermediate results of  
calculations more palatable and easier for the programmer.

> There are almost certainly some interesting interactions with things  
> like compressors and limiters, though.

Yes, they potentially play hob with dynamic range.

> In general, you want enough (analog) gain on the mic input to get  
> the signal well up into the dynamic range of the audio A/D that's  
> digitizing it, but after that, it's all floating point.

Until you reach the DAC.

Brian Lloyd
Granite Bay Montessori School          9330 Sierra College Bl
brian AT gbmontessori DOT com          Roseville, CA 95661
+1.916.367.2131 (voice)                +1.791.912.8170 (fax)

PGP key ID:          12095C52A32A1B6C
PGP key fingerprint: 3B1D BA11 4913 3254 B6E0  CC09 1209 5C52 A32A 1B6C





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