On 2/24/2015 7:22 PM, Jerry Stuckle wrote:
On 2/24/2015 7:07 PM, rickman wrote:
On 2/24/2015 6:37 PM, Jerry Stuckle wrote:
On 2/24/2015 5:47 PM, rickman wrote:
On 2/24/2015 12:00 PM, Jerry Stuckle wrote:
On 2/24/2015 11:32 AM, FranK Turner-Smith G3VKI wrote:
"AndyW" wrote in message
...
On 24/02/2015 12:47, gareth wrote:
What is the point of digital voice when there are already AM, SSB
and FM for those who want to appear indistinguishable from CBers?

Perhaps it is cynicism from the manufacturers who introduce such
things
as they see their traditional highly-priced corner of the market
being wiped away by SDR technologies?

Bandwidth reduction for one.
If you can encode and compress speech sufficiently then you can use
less bandwidth in transmission.

That's the bit I have trouble getting my head around. Back in the
1970s
and 1980s digital transmissions used a much greater bandwidth than
their
analogue equivalents. Sampling at 2.2 x max frequency x number of bits
plus housekeeping bits etc. etc.
A UK standard 625 line PAL video transmission would have used a
bandwidth of over 400MHz!
Times have changed and left me behind, but I've still got me beer
so who
cares?

But you forget compression. For instance, unless there is a scene
change, the vast majority of a television picture does not change from
frame to frame. Even if the camera moves, the picture shifts but
doesn't change all that much. Why waste all of that bandwidth
resending
information the receiver already has?

And voice isn't continuous; it has lots of pauses. Some are very
noticeable, while others are so short we don't consciously hear them,
but they are there.

And once you've compressed everything you can out of the original
signal, you can do bit compression, similar to zipping a file for
sending.

There are lots of ways to compress a signal before sending it
digitally.
About the only one which can't be compressed is pure white noise -
which, of course, is only a concept (nothing is "pure").

I think that depends on what you mean by "pure". Sounds very
non-technical to me. Even noise can be compressed since if it is truly
noise, you don't need to send the data, just send the one bit that says
there is no signal, just noise. lol


Pure white noise is a random distribution of signal across the entire
spectrum, with an equal distribution of frequencies over time. Like a
pure resistor or capacitor, it doesn't exist. But the noise IS the
signal. To recreate the noise, you have to sample the signal and
transmit it. However, since it is completely random, by definition no
compression is possible.

Here is a white noise signal... 4. That number was chosen at random,
courtesy of XKCD.com. http://xkcd.com/221/


No, that is not a white noise signal. And the number, by definition,
being computer generated, is only pseudo-random.

You didn't even read the damn reference. The number was *not* computer
generated.

--

Rick

On 2/24/2015 7:32 PM, Brian Reay wrote:
Jerry Stuckle wrote:
On 2/24/2015 5:47 PM, rickman wrote:
On 2/24/2015 12:00 PM, Jerry Stuckle wrote:
On 2/24/2015 11:32 AM, FranK Turner-Smith G3VKI wrote:
"AndyW" wrote in message
...
On 24/02/2015 12:47, gareth wrote:
What is the point of digital voice when there are already AM, SSB
and FM for those who want to appear indistinguishable from CBers?

Perhaps it is cynicism from the manufacturers who introduce such
things
as they see their traditional highly-priced corner of the market
being wiped away by SDR technologies?

Bandwidth reduction for one.
If you can encode and compress speech sufficiently then you can use
less bandwidth in transmission.

That's the bit I have trouble getting my head around. Back in the 1970s
and 1980s digital transmissions used a much greater bandwidth than their
analogue equivalents. Sampling at 2.2 x max frequency x number of bits
plus housekeeping bits etc. etc.
A UK standard 625 line PAL video transmission would have used a
bandwidth of over 400MHz!
Times have changed and left me behind, but I've still got me beer so who
cares?

But you forget compression. For instance, unless there is a scene
change, the vast majority of a television picture does not change from
frame to frame. Even if the camera moves, the picture shifts but
doesn't change all that much. Why waste all of that bandwidth resending
information the receiver already has?

And voice isn't continuous; it has lots of pauses. Some are very
noticeable, while others are so short we don't consciously hear them,
but they are there.

And once you've compressed everything you can out of the original
signal, you can do bit compression, similar to zipping a file for
sending.

There are lots of ways to compress a signal before sending it digitally.
About the only one which can't be compressed is pure white noise -
which, of course, is only a concept (nothing is "pure").

I think that depends on what you mean by "pure". Sounds very
non-technical to me. Even noise can be compressed since if it is truly
noise, you don't need to send the data, just send the one bit that says
there is no signal, just noise. lol


Pure white noise is a random distribution of signal across the entire
spectrum, with an equal distribution of frequencies over time. Like a
pure resistor or capacitor, it doesn't exist. But the noise IS the
signal. To recreate the noise, you have to sample the signal and
transmit it. However, since it is completely random, by definition no
compression is possible.


To be more accurate, it has an infinite bandwidth and constant power
density/Hz.

As you say, it doesn't really exist.

In practice, lab noise sources are specified over a bandwidth and to be
within a given limit of power variation across that.

Darn useful devices to have around.

So if I have a string of random numbers they can not represent a white
noise source of infinite bandwidth and constant power density?

--

Rick

On 2/24/2015 7:32 PM, Brian Reay wrote:
rickman wrote:
On 2/24/2015 6:35 PM, gareth wrote:
"rickman" wrote in message
...
On 2/24/2015 12:37 PM, gareth wrote:
"Spike" wrote in message
...
Get a CW signal peaked on the 20 c/s nose of the HRO crystal filter,
with
the phasing notching out any nearby signal, and you realise that DSP
just
isn't necessary due to the quality of the 80-year-old technology
employed.

WHS.

The Eddystone EA12 does not have a phasing control as that part of the
cct
is fixed-tuned, but it does have a tunable notch in the 100kHz IF to
achieve the same effect.

Mind you, there seems to be a diminishing band of people who know how to
do this, so the simplistic approach of using someone else's
ever-upgraded
software to do something less effective is about as far as the tick-box
Amateur seems to go. Heavens - they even buy ready-made wire aerials!

And going from previous threads, there are even fewer who understand that
setting up for single-signal reception means that the notional carrier
frequency has
to lie half-way between the peak of the Xtal and the notch of the phasing
control.

We should not forget that he who sneers loud and long about others' grasp
of
the mathematics of DSP maintains that changing the direction of a
rotating
vector
(A Phasor, and not related to the weapons of Star Trek!) causes it to
decrease in sixe.

What is "sixe"???

Typo - adjacent key - size

I thought it might be that, but it still makes no sense to me. Who or
how does changing the direction of rotation of a rotating vector change
its "size". Are you defining size as the rotation so that going from a +
to a - is like reversing the direction of a vector? I think most people
would consider the "size" of a vector to be the magnitude which is
independent of phase angle and so rotation, no?

Perhaps you can explain this with a little math?


He is (deliberately) misrepresenting the discussion. The point was made
that the phasor was rotating clockwise, thus the angle decreasing, ie
becoming negative.

This has been repeatedly explained to him but he continues to churn out his
bilge.

His maths (or math) isn't up to it, it is too complex for him (pun
intended).

If you look in the archives you will see him referring to 'negative
frequency', not to mention questioning basic DSP theory, the use of the
Dirac Delta, .....

Best just to ignore him, he is simply trying to start a row.

Maybe I don't understand the issue. Isn't that a valid example of a
negative frequency? There are some DSP experts in comp.dsp who talk
about negative frequency often.

--

Rick

On 24/02/2015 16:32, FranK Turner-Smith G3VKI wrote:
"AndyW" wrote in message

Bandwidth reduction for one.
If you can encode and compress speech sufficiently then you can use
less bandwidth in transmission.

That's the bit I have trouble getting my head around. Back in the 1970s
and 1980s digital transmissions used a much greater bandwidth than their
analogue equivalents. Sampling at 2.2 x max frequency x number of bits
plus housekeeping bits etc. etc.

But then you add compression on top. As technology increases and the
ability to process data quickly advances you can real-time encode and
decode data at a frightening rate. Back when I started playing about
with digital sound we had enough speed to run-length encode in real
time, now with dedicated number cruncher chips you can carry out very
complex lossless sound compression in real time and for lo-fi sound you
can use lossy compression and have a lot of the band left over for a
time-slice share.
One of my final dissertation for university was on digital compression
techniques (lossy and lossless) and I get a bit geeky about it all :-)
Surprised I still remember it all....


Andy

On 24/02/2015 16:57, gareth wrote:
"Iain Young, G7III" wrote in message
...
On 24/02/15 12:47, gareth wrote:

What is the point of digital voice when there are already AM, SSB
and FM for those who want to appear indistinguishable from CBers?

Technical developments of new encoding techniques that reduce required
bandwidth, just as SSB improved over AM. You can get CODEC2 down to
way less than an SSB signal quite easily.

Those who subscribe to these digital voice apparatuses lack a single clue
about any underlying technical development

Sweeping statement.

I have used digital voice and I have built my own kit (non-ham but
basically the same as a digital voice front end) and wrote my own codecs
(both fractal lossy and various lossless codecs).
If you have trouble sleeping at night I am sure I can dig out my
dissertation and forward it. The technology is dated (I built my first
one using thick-film technology - that dates it a little) but the
underlying work is still valid.

Andy

On 24/02/2015 17:00, Jerry Stuckle wrote:

But you forget compression. For instance, unless there is a scene
change, the vast majority of a television picture does not change from
frame to frame. Even if the camera moves, the picture shifts but
doesn't change all that much. Why waste all of that bandwidth resending
information the receiver already has?

Which is why, on cheaper televisions, the picture tesselates when
showing random images such as rain, fire, waterfalls etc.
The true test of a quality television is to watch a waterfall or flames
and see it pin-sharp.
Cheaper TVs use cheap lower-powered decoding systems and for complex
images they do not have enough time to fully decode the image before the
next frame arrives.

Andy

Thank you - you are much more accurate in describing it than I was able
to.


don't say that....he can barely get his head through a door as it is.......

On 25/02/15 06:43, rickman wrote:
On 2/24/2015 7:32 PM, Brian Reay wrote:
Jerry Stuckle wrote:
On 2/24/2015 5:47 PM, rickman wrote:
On 2/24/2015 12:00 PM, Jerry Stuckle wrote:
On 2/24/2015 11:32 AM, FranK Turner-Smith G3VKI wrote:
"AndyW" wrote in message
...
On 24/02/2015 12:47, gareth wrote:
What is the point of digital voice when there are already AM, SSB
and FM for those who want to appear indistinguishable from CBers?

Perhaps it is cynicism from the manufacturers who introduce such
things
as they see their traditional highly-priced corner of the market
being wiped away by SDR technologies?

Bandwidth reduction for one.
If you can encode and compress speech sufficiently then you can use
less bandwidth in transmission.

That's the bit I have trouble getting my head around. Back in the
1970s
and 1980s digital transmissions used a much greater bandwidth than
their
analogue equivalents. Sampling at 2.2 x max frequency x number of
bits
plus housekeeping bits etc. etc.
A UK standard 625 line PAL video transmission would have used a
bandwidth of over 400MHz!
Times have changed and left me behind, but I've still got me beer
so who
cares?

But you forget compression. For instance, unless there is a scene
change, the vast majority of a television picture does not change from
frame to frame. Even if the camera moves, the picture shifts but
doesn't change all that much. Why waste all of that bandwidth
resending
information the receiver already has?

And voice isn't continuous; it has lots of pauses. Some are very
noticeable, while others are so short we don't consciously hear them,
but they are there.

And once you've compressed everything you can out of the original
signal, you can do bit compression, similar to zipping a file for
sending.

There are lots of ways to compress a signal before sending it
digitally.
About the only one which can't be compressed is pure white noise -
which, of course, is only a concept (nothing is "pure").

I think that depends on what you mean by "pure". Sounds very
non-technical to me. Even noise can be compressed since if it is truly
noise, you don't need to send the data, just send the one bit that says
there is no signal, just noise. lol


Pure white noise is a random distribution of signal across the entire
spectrum, with an equal distribution of frequencies over time. Like a
pure resistor or capacitor, it doesn't exist. But the noise IS the
signal. To recreate the noise, you have to sample the signal and
transmit it. However, since it is completely random, by definition no
compression is possible.


To be more accurate, it has an infinite bandwidth and constant power
density/Hz.

As you say, it doesn't really exist.

In practice, lab noise sources are specified over a bandwidth and to be
within a given limit of power variation across that.

Darn useful devices to have around.

So if I have a string of random numbers they can not represent a white
noise source of infinite bandwidth and constant power density?


Not perfectly. The string would need to be infinitely long and truly
random. That is why the term Pseudo Random is generally used for strings
used in such applications.

Michael Black wrote:
On Tue, 24 Feb 2015, gareth wrote:

What is the point of digital voice when there are already AM, SSB
and FM for those who want to appear indistinguishable from CBers?

Perhaps it is cynicism from the manufacturers who introduce such things
as they see their traditional highly-priced corner of the market
being wiped away by SDR technologies?

Because it's something new, at least to amateur radio.

The phasing method of sideband was common in the early days of amateur
SSB (I recall reading the first rigs were filter type, but with really
low IFs, then phasing, then crystal and mechanical filters took over from
phasing). It offered up a lot on transmit and receive, though not perfection.

But now phasing is used a lot, because digital circuitry has made it
viable. I remember seeing some of the potential when phasing was still
analog, but I also remember reading articles where it was clear others
didn't see the potential. Sometimes ideas become lost when something becomes commonplace.

Who knows what would come from digital voice. But I remember 30 years
ago one local ham being interested in it, not to the extent of putting
something on the air, but as information from the computer world started
flowing in, the potential started being there. YOu can't resist new
things and say "they have no use", you have to embrace the new and see
what can be done with it. Maybe not as initially seen, but maybe it fits
in somewhere else.

Amateur radio has never done much with envelope elimination and
restoration (was that what it was called? I now forget). It's in one of
the sideband books, and Karl Meinzer of AMSAT fame wrote about it in QST
about 1970. Break the SSB signal into two components, so you can
multiply it up to a higher frequency, then modulate the output stage. If
you have an efficient modulator, you can do away with linear amplifiers
(which is why it was in that SSB book). I gather he used the scheme in
at least one of the amateur satellites after Oscar 6.

But what happens in the digital age? Can you generate the two streems,
in essence but not so simple an FM component and an AM component, without
needing to generate SSB and then extract the two streams? I don't know,
but so much digital processing is being done now, it may be something to
look into. With solid state devices and class D amplifiers, modulating
high level class C amplifiers can't be as much trouble as in the old
days. Maybe it amounts to nothing, but maybe it overall becomes more
efficient, if it can be done.

Maybe there's no value to digital voice, except that in the process of
learnign about it, and implementing it, one can learn something. Maybe
something merely new to the person learning, but maybe something
completely new. No advances are made without learning, the learning triggers new advances.

Michael

You do realise that you're responding to a troll post, right?

--
STC // M0TEY // twitter.com/ukradioamateur

Brian Reay wrote:
rickman wrote:
On 2/24/2015 6:35 PM, gareth wrote:
"rickman" wrote in message
...
On 2/24/2015 12:37 PM, gareth wrote:
"Spike" wrote in message
...
Get a CW signal peaked on the 20 c/s nose of the HRO crystal filter,
with
the phasing notching out any nearby signal, and you realise that DSP
just
isn't necessary due to the quality of the 80-year-old technology
employed.

WHS.

The Eddystone EA12 does not have a phasing control as that part of the
cct
is fixed-tuned, but it does have a tunable notch in the 100kHz IF to
achieve the same effect.

Mind you, there seems to be a diminishing band of people who know how to
do this, so the simplistic approach of using someone else's
ever-upgraded
software to do something less effective is about as far as the tick-box
Amateur seems to go. Heavens - they even buy ready-made wire aerials!

And going from previous threads, there are even fewer who understand that
setting up for single-signal reception means that the notional carrier
frequency has
to lie half-way between the peak of the Xtal and the notch of the phasing
control.

We should not forget that he who sneers loud and long about others' grasp
of
the mathematics of DSP maintains that changing the direction of a
rotating
vector
(A Phasor, and not related to the weapons of Star Trek!) causes it to
decrease in sixe.

What is "sixe"???

Typo - adjacent key - size

I thought it might be that, but it still makes no sense to me. Who or
how does changing the direction of rotation of a rotating vector change
its "size". Are you defining size as the rotation so that going from a +
to a - is like reversing the direction of a vector? I think most people
would consider the "size" of a vector to be the magnitude which is
independent of phase angle and so rotation, no?

Perhaps you can explain this with a little math?


He is (deliberately) misrepresenting the discussion. The point was made
that the phasor was rotating clockwise, thus the angle decreasing, ie
becoming negative.

This has been repeatedly explained to him but he continues to churn out his
bilge.

His maths (or math) isn't up to it, it is too complex for him (pun
intended).

If you look in the archives you will see him referring to 'negative
frequency', not to mention questioning basic DSP theory, the use of the
Dirac Delta, .....

Best just to ignore him, he is simply trying to start a row.

Or seek attention. He's bean almost universally ignored over in ukra for
some considerable time now, and is flailing around desperately trying to
get the spotlight on him. Our American friends will soon realise shunning
him is the wisest path.

--
STC // M0TEY // twitter.com/ukradioamateur