----- Original Message -----
Sent: Sunday, January 06, 2013 7:50
AM
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)
...
