DSD FME

doriboni

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Oct 31, 2023
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31
It was purely a display bug where the last packed octet was printed but not collected properly, thus the random value in it. I honestly hadn't noticed because I never bother looking at that bit. Regardless, the bug has been fixed, its been pushed to the aw_dev branch for now pending further testing. Further testing by somebody else, won't be me.

The bug only affected the printed AMBE value, it didn't affect the bit array ambe_d, where the 49-bit decoded voice frame is usually held, nor did it in any way impact the late entry MI/IV value held by the ambe frame, that MI extraction process is carried out separately from the mbe voice portion, and has both Golay applied to it and a CRC check.

If you are getting late entry errors often, then it means that for one thing, you aren't getting the initial link control headers nor the PI header, and it probably means that you are also having tons of reception errors if it is often bad or having to assert itself over the current internal LFSR generated MI/IV value. If you are getting tons of decode errors, then you should look at either cleaning up your signal into DSD-FME, or there is some interference on that channel, or the signal is just too weak, or maybe even too strong if its nearby to you.

Another thing to keep in mind is that if that system is also TXI enabled, the late entry key/alg has to compete for time with the TXI information, so its possible that on super short talk spurts with TXI enabled, and encryption enabled, if you miss the PI header, or have bad signal, it may have to wait multiple superframes, and by that time, the call could very well be over. Using the -0 and/or -3 option may be required to both disable the late entry alg/key and also force it to attempt RC4 decryption at all times, but it can only use one key value if you go that route, no key loading.
GREAT ! thanks !
 

CanesFan95

Analog already is interoperable.
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How do we make the screen stop scrolling like mad?

And how do we tell it to tune SDR++ to a new frequency for trunking? I have a 2-channel Cap+ system in a .csv, but SDR++ always stay on one frequency.

When I do get a voice call, no audio comes through.
 
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lwvmobile

DSD-FME
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Apr 26, 2020
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1,297
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Lafayette County, FL
How do we make the screen stop scrolling like mad?

And how do we tell it to tune SDR++ to a new frequency for trunking? I have a 2-channel Cap+ system in a .csv, but SDR++ always stay on one frequency.

When I do get a voice call, no audio comes through.

Oh, I don't know, let me read your mind and figure out what command you are running, and exactly what your csv file looks like, and the output to the console/log, and whether or not you've enabled RIGCTL and Network Audio in SDR++.
 

rabrol

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Dec 19, 2009
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260
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Edmonton, Alberta
Alright guys, sorry I haven't been on lately... I'm just a bit burned out and have realized that I'd like to spend my free time doing other things. I've also realized, that I think I've taken the original DSD source code as far as I can take it in its current form, and just jamming more and more things into it isn't going to help much or 'make it better' at this point.
I just wanted to pop by and say thanks for all your hard work on this over the years. Your contribution of dsd-fme to the hobbyist community is very much appreciated by myself, an I expect very many others.

Thank you for all the time, bug squashing, late night coding, and everything else. You have provided a major gift to us all.

I wish you well as you refocus your time and energy, rest, and explore new things.
 

ki4hyf

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Messages
203
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Jackson, TN
How do we make the screen stop scrolling like mad?

And how do we tell it to tune SDR++ to a new frequency for trunking? I have a 2-channel Cap+ system in a .csv, but SDR++ always stay on one frequency.

When I do get a voice call, no audio comes through.
Post your command line and .csv file contents. Then we will cross the no audio bridge once we get the other straightened out.
 

CanesFan95

Analog already is interoperable.
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Ok, sorry. This is what I'm running:

dsd-fme.exe -fs -F -i tcp -C capacity_plus_chan.csv -T -N
 

ki4hyf

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Jackson, TN
Ok, sorry. This is what I'm running:

dsd-fme.exe -fs -F -i tcp -C capacity_plus_chan.csv -T -N
A couple of things, I would try without the "-F" unless it really needs it.
You must have "-U 4532" to control frequency switching.
You must redirect STDOUT, like "2> log.ans" or it will scroll and look crazy.
So, probably a good command line would be:
Code:
dsd-fme.exe -fs -F -i tcp -U 4532 -C capacity_plus_chan.csv -T -N 2> log.ans
 

ki4hyf

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The .csv is as follows:
Your .csv file may have extra crap from excel, causing DSD-FME to not recognize the intended content. Edit with a text editor like notepad. It should look like this:
Code:
ChannelNumber(dec),frequency(Hz) (do not delete this line or won't import properly)
1,463600000
2,463600000
3,463950000
4,463950000
5,464162500
6,464162500
If you still have audio issues, check your sound manager first.
 

CanesFan95

Analog already is interoperable.
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Messages
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That got rid of the scrolling, but with -U 4532, it's still not following the rest channel changes. .csv looks fine in Notepad. This system uses a mixture of Enhanced Privacy (EP) and Basic Privacy (BP), even on the same TGs. Will I hear the digital noise, or does it only unmute for calls in the clear?

Running this now:

dsd-fme.exe -fs -F -i tcp -U 4532 -C capacity_plus_chan.csv -T -N 2> log.ans
 
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lwvmobile

DSD-FME
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Apr 26, 2020
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Location
Lafayette County, FL
Do you have the RIGCTL server added and enabled in SDR++? Should look like this once everything is connect up properly. If you haven't added the rigctl server module yet, just find the module manager as seen in the bottom of this image, open the menu box and add it and start it up.

Screenshot from 2024-07-23 06-13-30.png


Also, for what its worth, in SDR++, I would set your Radio to NFM, Bandwidth to 12000, and disable any Low Pass, High Pass, and Noise Reduction as the default settings for DMR. Might also be worthwhile to use -t 2 if on a Cap+ system, sometimes those short beacon bursts don't trigger fast enough, so we need to tell DSD-FME to increase the hangtime to 2 seconds before hunting for a new rest channel.
 

lwvmobile

DSD-FME
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Apr 26, 2020
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Location
Lafayette County, FL
Will I hear the digital noise, or does it only unmute for calls in the clear?

It will mute itself if it knows the call is encrypted, and you haven't supplied the appropriate type of key. It is, however, possible that some encrypted garble will get through while trunking if you miss the initial VLC and/or PI header, if its EP.
 
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Jul 8, 2024
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Hello, good afternoon, greetings to everyone. I just moved to Kali Linux, I already have Sdr++ and DSD-FME

Now a question, how do I do the subject of the audio? To switch it from SDR++ to DSD-FME, and then since the audio is coming out clean decoded, play it on speaker. In Windows it was with virtual cable, how do you do this in KALI? I appreciate your help!!
 

ki4hyf

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Jackson, TN
For me, using Ubuntu, the best way is to include -itcp on the command line. It should use pulseaudio output by default. I'm not sure if Kali is much different, so YMMV. In SDR++ you should make your settings similar to the attached picture:
 

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ki4hyf

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Jackson, TN
hi i have an old version of dsd fme installed on dragon os based on ubuntu how can i upgrade mbelib 1.3.0 to new version 1.3.4
I don't know about 1.3.4, but you should be able to upgrade from 1.3.0 to 1.3.2 (latest on lwvmobile's GitHub page) by running git pull from the mbelib directory.
 
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