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LF: Re: Re: Bandlimited SSB test on 136 kHz

To: <[email protected]>
Subject: LF: Re: Re: Bandlimited SSB test on 136 kHz
From: "Marco Cadeddu" <[email protected]>
Date: Sun, 6 Jan 2013 13:45:06 -0000
References: <871180718.807959.1357202851097.JavaMail.open-xchange@email.1und1.de> <[email protected]> <4D45055C381D4BECA4B1BEF1254E1059@Black> <004a01cdec0e$965b1020$0201a8c0@marco09cqcdi12>
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Sorry Markus...
I see from your grabber that you started.. but nothing here to hear..just a steady tone on 137.5
 
will qsy back on 472 wspr (rx only!)
 
73 de Marco, IK1HSS
----- Original Message -----
Sent: Sunday, January 06, 2013 1:06 PM
Subject: LF: Re: Bandlimited SSB test on 136 kHz

Hi Markus,
 
stucked of bad cnd on 472 (Stefan arrives irregularly with -30 dB S/N....) I'm now in rx usb on 136
 
GL
73 Marco, IK1HSS
----- Original Message -----
Sent: Sunday, January 06, 2013 12:50 PM
Subject: LF: Bandlimited SSB test on 136 kHz

Dear LF,
 
around midday I have been conducting some test transmissions using 136.0 kHz USB voice. To comply with German regulations, the audio spectrum has been cut off sharply below 300 Hz and above 1100 kHz, occupying 136.3 to 137.1 kHz RF. Despite this rather mediocre speech quality, it is still possible to hear the voice and copy a simple message. 
 
To my surprise, the speech was quite audible on the Twente WebSDR (434 km), squeezed between the bursts of DCF and HGA. There's a recording and screenshot at
Of course it does get boring speaking only to myself ;-). If you'd like to listen yourself, I intend to do another test transmission starting 13:30 this afternoon. As the audio is realtime and standard SSB (apart from the filtering), no software postprocessing is needed..
 
Slow voice transmission (ie the audio deceleration/acceleration method originally used by DK8KW and myself) would be nicer as it can fit a full SSB channel into 800 Hz. I have been working on a semi-automatic control, with a fixed one minute raster similar to JT9-1. This will hopefully allow us to comfortably exchange one 20 second voice message per time slot (speak during seconds 0 to 20, concurrent transmit and receive at 1/3 speed from 0 to 60, replay starting 40 to 60). Anyone interested?
 
Best 73,
Markus (DF6NM)
   
----- Original Message -----
Sent: Thursday, January 03, 2013 12:33 PM
Subject: Re: LF: SSB - why not go digital?

Hi Geri, all,
 
I like the idea of trying voice on MF or even LF, and I was fascinated by actually hearing Gus talking on 507 kHz, and Geri on LF slow voice. I admit that I even tried (low-power daytime) SSB above 135.7 kHz once, and Stefan was actually able to hear and record my voice, albeit at marginal SNR at 180 km. As my LF TX antenna is fairly narrowband (Q ~ 200), I had inserted a phase shifter to transform the low output impedance of the PA into a current source at the feedpoint.
 
The good thing about slow voice is that we also get the SNR gain associated with the narrower bandwidth. But the non-realtime operating procedure is sort of difficult. We did use "roger beeps" to mark the end of message, but there is always an inconvenient delay for recording and replaying the messages. So some sort of realtime narrowband voice transmission would indeed be desirable.
 
However I would NOT fancy digital voice modes at all. For one, like all digital modes it's all-or-nothing. Either the message decodes well, or you get garbage, and you don't know what has been going on in the channel. In linear analog SSB, you hear if someone else is calling on frequency, or what type of interference came up, or whether there is selective fading. The other thing I don't like about digital modes is that they tend to occupy the whole channel permanently with high average power - just look at the spectra of DRM vs AM modulated BC transmissions. It may well turn out that a 2 kHz linear SSB transmission is much more friendly than a 1 kHz digital channel. 
 
I have been making some attempts on analog quasi-realtime narrowband voice transmissions. The principle was is much the same that has long been used for changing audio speed without changing pitch, ie cut the audio into ~ 20 ms grains. To accellerate or reduce bandwidth, you either leave out or average several grains, using a sliding window. On replay, each grain is repeated to extend it in time. This is not difficult to implement. In my own trials, I could maintain speech readability with a factor four bandwidth reduction, but it did sound very "robotic" because the fixed timing for the grains impressed itself onto the voice pitch. I think that there are better ways of adapting the timing by tracking the fundamental frequency to preserve the pitch modulation. Using Windows media player for accelerating and decelerating seemed to provide quite natural speech quality at a reduction factor of four.
 
Regarding the frequency allocations, like Graham I would think that on MF the narrowband beacons should best be placed in narrow slots near the band edges. To avoid blocking, we would preferably again use split band for west-east vs east-west intercontinental work, as we do on LF. Parts of these slots should also be utilized for the slow versions of Opera or WSPR. The wider modes would better be placed in the in the middle, perhaps with 2 kHz each for CW, middle-range digital modes, and one full-width SSB channel.
 
Best 73,
Markus (DF6NM)
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