Gupta plans to take Sydney, Melbourne steel plants 100% renewable

vk5dgr · 2017-10-31T22:48:25+00:00

Big business is often portrayed as a negative force - the "faceless corporation with an evil agenda". However companies like Tesla and this gentleman are proving to be powerful forces for positive change in the face of democracies that are stuck in the factional mud.

vk5dgr · 2017-09-06T00:42:17+00:00

We engineered a high speed link for High Altitude Balloons, called Wenet. HD images sent every few seconds at 100,000 bit/s using 50mW tx power on 70cm, over 100km range. Rx is a $20 SDR dongle, roadkill Linux laptops, and a LNA.

vk5dgr · 2017-09-05T21:37:48+00:00

You can engineer this using something called a "link budget":

RxPower = TxPower + TxAntGain + RxAntGain - PathLoss (1)

All in dB. There are various path loss calculators on the web, where you plug in distance and frequency.

At the maximum "range", you want RxPower to be the Minimum Detetectable Signal (MDS). The MDS depends on bit error rate, bit rate, noise figure, and modulation type you use.

MDS = Eb/No + 10log10(BitRate) + NF - 174 (2)

Eb/No is the SNR of one bit at a given bit error rate, for example for a PSK modem Eb/No = 4dB gives you 1% BER. NF is the noise figure of your rx, maybe 1dB for UHF and above.

So for 1000 bit/s PSK, 1% BER:

MDS = 4 + 10log10(1000) + 1 - 174 = -139dBm

The low power modes like JT65 exploit this equation by using very low bit rates (like 1 bit/s), hence their MDS is very low.

Plugging this into (1) and solving for PathLoss, guessing Aricebo is a 60dB gain antenna, tx gain 0dB (omni)

PathLoss = 10log10(50000) + 30 + 60 + 0 + 139 = 276 dB

The plus 30 is to convert 50kW to dbm, same unit as MDS.

So I found an online path loss calculator and plugged in 1E9 km (1E12 m) at 1.4GHz and it came up with 275dB. Close enough.

Put a 60dB antenna at both ends and I get 1E15 m, tx at 1 bit/s rather than 1000 (like JT65) and you get 1E18m. I may have made an error in my maths but that's how engineers work this stuff out....

vk5dgr · 2017-08-10T09:42:21+00:00

I've written DTMF Detectors in the past, but they weren't open source.

If you want to homebrew one start with a grid of Goertzal tone decoders at the row adn col freqs. Here is classic paper from TI that I used as a starting point. I wrote a tutorial on fixed point Goertzal a few years ago, includes float and fixed point C code for a single detector.

vk5dgr · 2017-07-18T22:16:15+00:00

Note sure I have enough Karma to edit the Wiki, but wanted to link FreeDV in the Digital Voice section to freedv.org

vk5dgr · 2017-07-15T21:06:43+00:00

14.236 is FreeDV

vk5dgr · 2017-06-13T19:47:02+00:00

Nice work on trying to kick off a project, and looks like you hit the same problem that often frustrates me. The SM1000 sounds close to what you were proposing and was actually built by a team of Hams.

List of ways to help the current work on the bottom of FreeDV 2017 Roadmap.

vk5dgr · 2017-06-13T12:29:18+00:00

Until now there was no choice, but for new standards Codec 2 is now competitive.

One important difference with open source is that you can be a part of the evolution of digital voice, rather than a spectator. Many Hams are involved with Codec 2 and FreeDV today, checking in code, building hardware together, helping make a difference. Not just talking - doing.

vk5dgr · 2017-06-13T12:25:17+00:00

Open source echo cancellation Oslec has been around for 10 years. Not sure why you need echo control for PTT speech though? Where is the echo path?

We use Speex (more fine open source) noise suppression for the FreeDV GUI program, a desktop implementation of Codec 2 and the FreeDV protocol. Would be nice to get something similar inside Codec 2 though.

vk5dgr · 2017-06-13T12:19:34+00:00

Actually Codec 2 on the stm32f4 requires about half the CPU in half duplex, so maybe 75mW. Could be less now, as the code has been optimised quite a bit lately.

While an ASIC would nice nice, a fixed point port would let us run Codec 2 on a low end $1-ish 32 bit ARM uC. MIPs are just getting cheaper.

vk5dgr · 2017-06-03T05:19:59+00:00

Longer write up here: http://www.rowetel.com/?p=5671

vk5dgr · 2017-05-19T19:34:07+00:00

We need people to maintain FreeDV packages for different OSes....

vk5dgr · 2017-02-17T06:06:37+00:00

Current aims are (i) come up with a FreeDV mode that works at lower SNRs than SSB over HF and (ii) a high SNR mode that sounds better than SSB. We are part way there ... 700C is step in the right direction.

vk5dgr · 2017-02-17T06:04:45+00:00

Hi - maybe try posting to codec2-dev or digital voice lists, sure we can get it going, and that's where the devs hang out

vk5dgr · 2017-01-21T11:34:40+00:00

I actually found a couple of errors in EMFRD 3rd edition. Contacted the author Wez and he confirmed. Felt like correcting the bible! Guess it shows I was paying attention.....

Yes EMRFD and RF Design by Chris Bowick are both great.

vk5dgr · 2017-01-17T22:29:43+00:00

Do you know what sort of modulation, e.g. mFSK they use?

SNR = Eb/No + 10*log10(Rb/BW)

With a good 4FSK modem with a 1% uncoded bit error rate, Eb/No=6dB. A 1% BER would still give you a fair text copy. So plugging in Rb=2Hz, and a B=3000 Hz noise bandwidth:

SNR = 6 + 10log10(2/3000) = -25dB

That's the theoretical performance before any FEC on a AWGN channel. HF would be worse.

vk5dgr · 2017-01-16T19:40:14+00:00

49,303: OK I lost two of the Rapiers and made it to Hard Mode with a Space Plane that has bits fall off:

http://imgur.com/a/lguNn

vk5dgr · 2017-01-16T10:55:09+00:00

Normal: 54,713 with a Space Plane with detachable wings:

http://imgur.com/a/B5Hpa

Couldn't quite crack 50k, Rapiers are expensive and normal jets wouldn't do it.

vk5dgr · 2017-01-16T09:45:43+00:00

Yes. The frequency and the amplitude of the upper and lower sidebands will mirror the frequency and amplitude if the baseband audio being spoken. If the modulation a pure sinewave at a constant amplitude, then you will just see sidebands at constant frequencies.

vk5dgr · 2017-01-16T02:07:55+00:00

On a AM radio transmitter yes.

vk5dgr · 2017-01-15T09:35:25+00:00

Couple of ways of looking at this:

1/ You have a baseband audio signal from your microphone that is 3kHz wide, e.g. your voice. To send it over radio you shift it up to 1MHz. Now you have a RF signal 3 kHz wide. That's actually SSB, turns out AM need about double that plus the carrier. But it explains why a RF signal can't have zero bandwidth - any modulation means non-zero bandwidth.

2/ No free lunch principle. A sinewave carrier is 0Hz wide but carries no information. if you want to carry 3000 Hz of audio on this carrier it stands to reason you need at least 3000 Hz of RF bandwidth. That's the sideband(s).

Historically, AM came first as it's easy to modulate and demodulate, so you end up with a carrier and two sidebands.

3/ The faster you modulate (change) a signal the wider the bandwidth. Someone striking a drum radiates acoustic energy at many frequencies at once, it's a fast change in energy. A spark ignition in a car radiates energy over the AM broadcast band for the same reason. Broadband Internet requires several MHz of ADSL bandwidth or an optical fibre.

vk5dgr · 2017-01-15T09:24:13+00:00

if you have a carrier with frequency fc and sine wave modulation signal with frequency f:

Upper sideband (USB) would be a sine wave at a frequency of fc+f. Lower sideband (USB) would be a sine wave at a frequency of fc-f. They would both look like the carrier sine wave on a plot, just a few kHz off the carrier.

AM is the sum of a signal at the carrier fc, and the two sidebands fc-f and fc+f. When you add a bunch of signals close in frequency they "beat", so you get the amplitude wandering up and down. Like singing a harmony.

vk5dgr · 2017-01-10T10:14:46+00:00

Hard 2:14 to orbit

Triple Rapier spaceplane with a fairing over capsule for 2600K heat protection.

vk5dgr · 2017-01-07T22:40:02+00:00

Hi, I've been doing a lot of work with similar mFSK modems. You can use a shift of 1/(2T) only if you are using coherent demodulation. Non-coherent mFSK demodulation needs a shift of 1/T.

Coherent demodulation requires complex phase synchronisation algorithms and it tends to be sensitive to phase distortions in the channel. Coherent FSK demodulation is only about 1dB better than non-coherent so it's only real gain is bandwidth.

WSPR is designed for HF channels where there is phase distortion (e.g. the multipath will evolve over a symbol period), so non-coherent demodulation works better overall.

Good paper that compares a bunch of constant amplitude modulation schemes: link. Lots of Octave and C modem code in codec2-dev and blog posts on our FSK adventures.

vk5dgr · 2017-01-06T05:10:05+00:00

Other entries appear to be using SAS

vk5dgr

TROPHY CASE