In Search Of... A Few Good Developers

Occasionally I get a chance to poke my head up and see what is new and different and occasionally I get asked if I know anyone that could help.  This is one of those cases.

I got a nice email from Mike Lebo, N6IEF, this afternoon, asking if I knew of any Open Source developers, interested in tackling a project that he has in mind, and he sent me along the guts of it. 

If you want to jump right into what he has in mind, he has posted it to GoogleDocs.  For the rest, let me paraphrase.  Mike is out to modify and write software needed to convert the analog voice of a ham radio operator into narrow band digital modulation and convert the received digital modulation into a selectable synthetic voice.

Now, I am sure you are asking: Why do this?

The bandwidth of voice is about 2,400 Hz. When speech is reduced to 125 Hz, the gain is 12.8 dB (19.2 times). Processing gain by a computer is cost free. This project receives weak signals 9 dB below the SSB (Single Side Band) noise floor of a radio.

 

OK, this is pretty geeky stuff!  (If you thought Amateur Radio was just a bunch of old white guys warming their hands over tubes, think again!)  Mike added, in his email: I want to do phone moon bounce and it is very hard with ssb.   Now that is really geeky.  (Yes, there is a whole subculture of operators that live to bounce signals off the moon to each other.)

So, if you are looking for a summer project and hacking the kernel just seems so ... yesterday, here is a perfect project for you to flex your muscles on.

Now, if you will excuse me, I have to crawl back into KVM.

 

______________________

David Lane, KG4GIY is a member of Linux Journal's Editorial Advisory Panel and the Control Op for Linux Journal's Virtual Ham Shack

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tdlhygiene's picture

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This isn't exactly a codec

KA8KPN's picture

A while ago, I imagined transmitting a voice signal by converting the voice into phonemes, and then reconstituting that voice at the far end by running the phoneme stream through a voice synthesizer. The idea being that, since the phoneme stream would be generated from an actual voice, it would be far easier to understand than the typical output of a text-to-speech system and, since it would not be intended to be read by a human, it wouldn't be near as hard to convert to the voice into the phoneme stream as it is to convert voice into plain text.

This seems to be a similar idea. I suppose it's a good idea, but I no longer think that generating the phoneme stream is particularly easy. You'd need to find someone who understands a little bit about the physics of speech, though, because J. Random Coder isn't going to know enough about the problem domain to make much headway.

You might be able to get the data rate down enough to do voice moonbounce with large amateur stations. I don't know. I do know that I'm not going to give up on my 10 GHz free-electron maser idea.

Digital speech within 125 Hz bandwidth

Mike Lebo's picture

Since I do not know to write software, I do not have any of the previous tried solutions which did not work. Voice is composed of at least 4 parts. Speech is the only one that communicates ideas. We make automatic gain controls to keep the amplitude at maximum level, so that should not be sent and use bandwidth. Pitch makes one person sound different from another, but should not use bandwidth when we do not know what the other person should sound like. Timing is critical. You cannot send a message faster then it takes a person to say it. Sending a message slower then a person says it waste bandwidth. My idea handles all 4 of these parts of voice. Please contact me or better yet help write software to hake this happen.

mike.lebo@gmail.com n6ief

thank you

voyance en directe's picture

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Amateur Radio Codec Development

K7VE's picture

The CODEC2 project (http://codec2.org/) might be a place to start.

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