DSD FME

lwvmobile

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AFAIK, SDR# doesn't do rigctrl, so it won't automagically tune to voice channels. SDR++ will.

You got to use those plugins for the TCP Audio and GpredictConnector for the Ricgtl in SDR# to work. Don't know how far they are compatible though, but people keep insisting on using SDR# for various reasons and not SDR++, where it is natively integrated.
 

TheButcher

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I prefer SDR# for it's ability to show alpha tags better than what SDR++ does. And also, people were using SDR# as SDR++ wasn't about back when.

This is a feature request in disguise !
 

Marco192

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Is there a way to setup tcp audio input from SDR++ that is located on another PC. I have setup a VPN network, so I do have a local network address.
I was trying with something like this:
dsd-fme.exe -fs -i tcp:x.x.x.x:7356 -U 4532 -T -C dmr_t3_chan_cn1.csv -G group.csv -N -Z 2> NUL
but dsd-fme would not start. Please help with commands and SDR++ settings.
I am using the last precompiled windows version on Win10
 

lwvmobile

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Is there a way to setup tcp audio input from SDR++ that is located on another PC. I have setup a VPN network, so I do have a local network address.
I was trying with something like this:
dsd-fme.exe -fs -i tcp:x.x.x.x:7356 -U 4532 -T -C dmr_t3_chan_cn1.csv -G group.csv -N -Z 2> NUL
but dsd-fme would not start. Please help with commands and SDR++ settings.
I am using the last precompiled windows version on Win10

I honestly don't see why not, just make sure the address in SDR++ is also set to the IP4 address assigned to it that it should be listening on and not to localhost. Just keep in mind though, if you are using a VPN or over the Internet, bandwidth, stability, latency, etc may play a role in decoding and so on.

Another thing that we have done in the past is run SDR++ server on one PC on a remote location, and had the local SDR++ instance connect to it over the internet and do remote decoding and trunking that way. Not going to lie though, needs tons of bandwidth and stability, it can get weird form time to time if your ISP or the remote ISP sucks or the route tanks for whatever reason.
 

Marco192

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I honestly don't see why not, just make sure the address in SDR++ is also set to the IP4 address assigned to it that it should be listening on and not to localhost. Just keep in mind though, if you are using a VPN or over the Internet, bandwidth, stability, latency, etc may play a role in decoding and so on.

Another thing that we have done in the past is run SDR++ server on one PC on a remote location, and had the local SDR++ instance connect to it over the internet and do remote decoding and trunking that way. Not going to lie though, needs tons of bandwidth and stability, it can get weird form time to time if your ISP or the remote ISP sucks or the route tanks for whatever reason.
Thanks for the assist ... it's connected now, waiting for some signals to check how it's decoding.
The correct command for connecting to rigctl server is -U tcp:x.x.x.x:port (it may help someone)
The problem is that the connection on the remote PC is very slow, but I am hoping that the tcp audio will not take too much bandwidth.

btw. dsd-fme works wonders - it correctly decodes some group calls in this DMR tier3 network, that DSDPlus does not follow.
 

ki4hyf

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I downloaded a compiled release from GitHub.
I really don't like GitHub. I am not a programmer and sometimes cannot make heads or tails of the way stuff is structured there.
I found the latest release version from May. That is the only obvious link to anything I can recognize. I don't see anything that says "source code" or any later version than the one that is labeled as "Latest".
You may want to look at GitHub - arancormonk/dsd-neo: A modern, modular, and performance enhanced C/C++ decoder for digital voice. DMR, P25, NXDN, YSF, and more.. It is a fork of DSD-FME. It is NOT dsd-fme, so please don't post questions about neo here.
 

DaveNF2G

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You may want to look at GitHub - arancormonk/dsd-neo: A modern, modular, and performance enhanced C/C++ decoder for digital voice. DMR, P25, NXDN, YSF, and more.. It is a fork of DSD-FME. It is NOT dsd-fme, so please don't post questions about neo here.
So who is handling the Neo question and where?

In any event, neither program (FME or NEO) displays anything after the first two lines in the Latest Event History segment. Also, the symbols at the top of that segment look like settings. How does one change them?
bandicam 2025-10-31 15-20-17-752.jpg
 

DaveNF2G

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Is this information sufficient for enhanced use?

2025-11-01 13:28:31 DMR TGT: 00005400; SRC: 05407550; CC: 13; ENC; ALG: 21; KID: 01; Group; Slot 2;
 

sh7411usa

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Two questions:
1) How do I compile the latest version for cygwin/windows with color output?
2) How do I direct audio input from different sources to different instances of DSD-FME, eg. VAC Line 1 to 1st instance of DSD-FME, VAC Line 2 to second instance of DSD-FME in windows? If the answer is pulseaudio, how do I get that up and running on windows. I am aware I can use TCP with separate ports to separate instances of DSD-FME, but I am asking specifically about using audio devices, as this allow me to use multiple decoders, (I.E. DSD-FME for audio decoding, DSDPlus for logging) and to further propagate the audio to streams easily, as they can all listen to the same input device, whereas SDR# or SDR++ TCP only supports one listener at a time.
 

sh7411usa

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Two questions:
1) How do I compile the latest version for cygwin/windows with color output?
2) How do I direct audio input from different sources to different instances of DSD-FME, eg. VAC Line 1 to 1st instance of DSD-FME, VAC Line 2 to second instance of DSD-FME in windows? If the answer is pulseaudio, how do I get that up and running on windows. I am aware I can use TCP with separate ports to separate instances of DSD-FME, but I am asking specifically about using audio devices, as this allow me to use multiple decoders, (I.E. DSD-FME for audio decoding, DSDPlus for logging) and to further propagate the audio to streams easily, as they can all listen to the same input device, whereas SDR# or SDR++ TCP only supports one listener at a time.
Figured out the answer to question #2, using the windows sound mixer. Its been talked about already here. Question #1, still pending though...
 

Marco192

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Figured out the answer to question #2, using the windows sound mixer. Its been talked about already here. Question #1, still pending though...
Please share the answer to #2 ... because after several attempts I did not manage to start and setup pulseaudio on windows o_O
 

sh7411usa

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Please share the answer to #2 ... because after several attempts I did not manage to start and setup pulseaudio on windows o_O
I gave up on pulseaudio, couldn't figure it out. Here's how to route the audio in windows:

1) Just start DSD-FME using -i /dev/dsp, or don't specify `-i` at all.
2) Open the windows sound mixer. (Windows 11: Settings > System > Sound > Volume Mixer).
3) Find DSD-FME and set the input device:
1762873073574.png
Change it from default to whatever input you'd like to use.
4) Important: Restart DSD-FME!
This worked for me.


For Multiple instances of DSD-FME: (Examples using 2 lines: VAC Line 1 and VB-Audio Cable)
1) Copy the DSD-FME executable, and give it another name.
1762873575333.png
2) Run both copies, using -i /dev/dsp. Here called dsd-fme.exe and dsd-fme-2.exe.
3) Open the volume mixer and set the desired input device. Here, dsd-fme.exe is set to VB-Audio Cable, and dsd-fme-2.exe is set to VAC Line 1.
1762873419063.png
4) Restart both instances of dsd-fme, and you should get separate input to each instance.

Here is a screenshot of the setup, using SDR++ with two VFOs. `Radio` is parked on a P25 channel, and is sending its audio to dsd-fme.exe via VB-Audio Cable. `Radio 2` is parked on a Con+ control channel and is sending its audio to dsd-fme-2.exe via VAC Line 1.
1762873379236.png
Hope this helps!
 

sh7411usa

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I'll add one more question:
3) How can I achieve per call recording (-P) on a busy Con+ system - for only one specific talkgroup? I'm using rigctl (-U) together with a freqency_map.csv (-C) and group_list.csv (-G). All undesired talkgroups are set to B (Block) except the one I am interested, and while they don't play audio, they DO generate per call wav files. Tried (-I) to hold on a specific talkgroup, but this has two undesired effects. It seems that it blocks trunk-tracking/rigctl, and it still allows per call wav files to be generated for other talkgroups. So, bottom line, how can I use dsd-fme with rigctl to properly trunk track and play audio and generate per call wav recordings for only one specific talkgroup?
 

sh7411usa

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You may want to try this:
lol, that was me he was responding to... ;-) i posted the question here, and then made the feature request on github after discerning that what I wanted wasn't possible (yet).
 

vselic

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Hello, i have two simple questions:
from dsd-fme, how would I decript sms/tms
from dsd-fme, how would i decript lrrp

key is KNOWN, no illegal or hacking! I have own duplex frequency rpt and would like to monitor 5 radios in the system, arc 4 encrypted on PC.
 

hey

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Hello everyone,
I'm trying to monitor a NXDN system but for some reason, I can't get it to work for me. I'm using DSD-FME with SDR++. Here's what i have so far if anyone can correct me, I greatly appreciate.


dsd-fme -fi -i tcp -U 4532 -N -Z 2> log.txt -C FleetCallSite1.csv -T

ChannelNumber(dec),frequency(Hz) (do not delete this line or won't import properly)
127,451787500
199,452237500
668,463300000
752,463825000
874,464587500
 

lwvmobile

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Hello everyone,
I'm trying to monitor a NXDN system but for some reason, I can't get it to work for me. I'm using DSD-FME with SDR++. Here's what i have so far if anyone can correct me, I greatly appreciate.


dsd-fme -fi -i tcp -U 4532 -N -Z 2> log.txt -C FleetCallSite1.csv -T

ChannelNumber(dec),frequency(Hz) (do not delete this line or won't import properly)
127,451787500
199,452237500
668,463300000
752,463825000
874,464587500

That should be fine as far as I can tell, assuming the channels are matched properly to all the frequencies, making sure to include the control channel as well so it'll tune to it.

One thing to mention though, for NXDN, use the squelch box in SDR++ and set it so block out the noise floor, but allow the signal to pass, and turn off the low pass filter if its on. You may need to tweak the BW a bit from anywhere to 6-10 on NXDN48 for optimal decoding, depending on signal strength. Just play around with it and see what works best.

Also, just as a general aside, I may not be around much for the rest of the year due to holidays, etc, so I may or may not reply depending on what's going on. Happy Holidays folks.
 

Ubbe

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I can't get it to work for me. I'm using DSD-FME with SDR++.
What exactly are not working? Can you hear the NXDN modulation in SDR++, if not then you have issues to receive those frequencies and needs to first look at SDR++ what can be done there, or its antenna.

If SDR++ can be heard receiving datasignals are DSD-FME then successfully connecting to the TCP link?
What does the window of DSD-FME indicate?

/Ubbe
 

hey

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OK gentlemen, I tried everything that you said and I’m coming up short still has anybody that can take a screenshot of your SDR plus plus to make sure I everything set right
Sorry to be a bother, but I appreciate everyone helping me
 
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