----- 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)
...
