Ailunce HD1 using ATAK on Android phone. Please help how to setup

Im looking to run ATAK with an Ailunce HD1 DMR radio with a Motorola M3 interface. Can you list the steps and hardware to achieve this please.

From what I see, I’ll need your DigiRig dongle and leads from radio thru to Android phone.

Please help!

I have had zero success using digital radio. I have used Retivis RT3S radios that are digital, but only in an analog mode. Digital just does not work. And I have no idea why.

the retevis has analog and dmr operation.

yes, dmr is digital, but it not the same thing as using a smartphone or pc to modulate/demodulate digital data.

the ‘digital’ operation of any DMR/DSTAR/etc radio is that it will take your voice and digitize it according to the BUILT IN CODEC. then reverse the operation on receive.

if you feed a digital signal from phone/pc into the radio when it is in the ‘digital’ mode, it will try to encode (modulate) a signal already encoded and well outside the input expectations.

:slight_smile:
so that’s why using phone/pc to do ‘digital’ requires the radio to be in analog mode.

good luck
kb0wlf

So the radio is set up to digitize a certain “type” of sound like voice, and the digital fax tone is basically out of spec. and doesnt translate well?

that’s it more or less.
i wish i could explain better.

say you have an icom D-STAR capable radio.
when you put it in d-star, or ‘digital’ mode, it is only for communicating with other d-star radios in d-star mode.
nothing else.
now d-star has data transmission as well as voice but again, only with other d-star radios.

same for the retevis except it uses ‘DMR’ which many brands of radios support.
it is still only for ‘talking’ to other DMR radios.

and yes, any digital ‘noises’ from a pc program/smartphone program would be not be reassembled on the other end :slight_smile:

now, in case any hf guys are lurking this thread.
many hf rigs these days have USB-D and LSB-D(igital) selections.
these simply switch the audio path from mic/spk to the accessory port OR built in soundcard. they are not the same as the vhf/uhf radio ‘digital’ mode selection.

clear as mud isn’t it?
:wink:

kb0wlf

Yes, and most digital modulation protocols don’t send your entire voice, for lack of a better way to explain it. They sample the (analog) sound and send a digital representation of it. If the sampling rate is fast enough (Nyquist limit) , and the original digital audio signal is entirely within the passband of the transmitter and whatever protocol IT is using to encode the audio signal, it should work. I don’t know what encoding Hammer uses to digitize its output, or what the bit rate is, but it’s likely that it’s incompatible with DMR for a reason similar to the above. If I ever get a cable sorted, I’ll try it on P25, but I would expect a similar result. This is why I want a cable that will connect to the actual data in on the Motorola XTS radios; it will bypass the audio vocoder section of the AMBE chip and put the input data directly into P25 data format.

This is not going to be an option on Chinee commercial and amateur radios.

Edit: I think I just figured it out: in the other Hammer thread, there’s a pic posted showing the Hammer bit rate set at 16kbps. DMR is much slower than that, because it has to fit in a 6.25khz channel:

“The DMR standard operates within the existing 12.5 kHz channel spacing used in land mobile frequency bands globally, but achieves two voice channels through two-slot TDMA technology built around a 30 ms structure. The modulation is 4-state FSK, which creates four possible symbols over the air at a rate of 4,800 symbols/s, corresponding to 9,600 bit/s. After overhead, forward error correction, and splitting into two channels, there is 2,450 bit/s left for a single voice channel using DMR, compared to 4,400 bit/s using P25 and 64,000 bit/s with traditional telephone circuits.”

So, you’re trying to force a 16kbps signal down a 2.4kbps pipe…which ain’t going to work. In fact, due to Nyquist limits, you’d need to have those numbers almost reversed to have a chance of this working. As it is, the DMR vocoder is missing probably 75% of the symbols Hammer is sending.

As I said in the other thread… this is, unfortunately, not going to work.

Yeah, sometimes it feels like it’s all “theory” not just “antenna theory”.

I don’t understand everything you said perfectly, but I got enough of the gist of it so say… 1. That’s the best explanation I’ve yet heard, and 2. It makes the most sense. If I’m understanding correctly. The vocoder, is a middleman that seems to be screwing up the interpretation of a fax tone (ATAK data) being turned into a digital signal (DMR). And to piggyback on a previous explanation the radio (vocoder I’m assuming) is looking to turn voice into digital and doesn’t do a good job with input that isn’t voice. Side note I have always hated the way voice sounds over digital. It just sounds like only half the sound is really there. Like looking at a picture with a third of pixels missing. You can still tell what it is, but it takes more work to interprate it than it should. I find it difficult to listen to.

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Hell, I don’t understand half what I said🤣 but yes, you got it.

And re your comment about voice sounding unpleasant over digital, you’re definitely not the first person to say that… but as always there are a lot of variables. A PROPERLY set up P25 radio sounds much better TO ME than DMR usually does, and there are a lot of possible explanations for that, but it all comes back to the fact that a digital voice communication system is constrained by the bandwidth the RF signal is permitted to occupy, in the case of VHF/UHF bands (disregarding amateur bands), that limit is 12.5 kHz. And the digital protocols in use take up a slice of that bandwidth for data signaling and other overhead, so the pipe gets even smaller. Then, you chop off all the audio information below 800hz and above 3000hz so your codec/digitizer/vocoder/whatever only has to work on that much bandwidth, you start to see how it could sound different on the receiver.

If you could run a 96khz direct sampling rate, on a 250-20000hz input audio signal and use a 50mhz wide channel to send it down, you’d have fine intelligible audio indeed. But there isn’t that much room in VHF and UHF low bands for more than one or two channels that size, which is why it isn’t done that way.

Hammer is neat, but it’s forcing a Porsche to try a lap at Nurburgring on Hoo Flung Poo bias-ply trailer tires. You might get away with it a few times, but you aren’t getting the experience Dr. Porsche intended…

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