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Re: fldigi psk waveform dropping samples? #fldigi #fldigi-parameters


 

Thanks for clarifying. I have seen the dropped bits with many fldigi modems. I don't know why they occur. There are several possibilities related to threading and OS not being real time. I find the signal error is usually in the same place for a set of circumstances, but does move around depending on the modem of choice and the waterfall "center freq".

Keep in mind that you are looking at what fldigi thinks it is sending the CODEC, so the SIG, FFT, and waterfall all have naught to do with anything further downstream...AFAIK. I don't think fldigi looks at the device, looks at its digitization freq (mine is set at 48kHz), and then creates a signal. I ~imagine~ that fldigi simply creates a digitized version of the sine wave it needs (actually combining them for more complex signals) and sends that to a CODEC. The CODEC takes the wave and samples it at the rate defined in the device table (Sound settings, in Windows), and sends that sampled signal to the next device. Someone on the dev team will need to correct me on that.

BTW, I also use 500 Hz or so for my "testing" when I want to look at signals.

As for the spectral artifacts...those audio sideband artifacts...good question. I assume that the SIG display and the FFT display show exactly what fldigi is producing. And there are all those "sidebands". I don't know the method or algorithm. "Filtering" an audio signal in real time is expensive. We all don't have spare FPGAs floating around our laptops. :-) So if the method for producing the signal produces audio side tones, they are transmitted. The question is...to what end? On the RX end, fldigi uses an FFT to find the real data in all the noise, so those very low signals don't "make the cut", so to speak, and don't figure into the final data for decoding what freq was the important freq. And people aren't really listening to the sidebands, are they? Even when we listen to the TX audio at the fldigi end, we focus on the part of the signal that is much "louder" than the sidebands.

We are used to thinking that the sounds are all clean audio but they aren't. If I play a piano, theoretically I can get a pure note (not IRL), but the instant I send that to a digitizing process, I change it. I have audio samples here that were recorded with NO digitizing, and the same signal sent in parallel to very expensive digitizing equipment. I can hear the difference. fldigi, the new DSP radios...they all are creating artifacts, and we just ignore them, often because we are so thrilled with what the digitizing is doing for us, such as filtering out "noise". The only way you are going to get signals without artifacts is by using analog all the way from your voice to the antenna...oops...we hams all know that's not true, right? Otherwise, why do we have an FCC spec on transmitted bandwidth? We have it because the instant we combine the audio stream with the carrier, we produce sidebands.

On 2024-09-12 16:54:, Curt Karnstedt wrote:
And when I say "carrier", I mean center frequency, understanding that no carrier is present.
Got it.

The picture of the signal scope is just transmit with no on data going out, while the spectrums are showing sending data.
Here I assume you mean that you hit TX with no data, while I usually test by hitting Tune.

73 Rich

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