"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.

On Tue, 24 Feb 2015 16:57:04 +0000, gareth wrote:

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


I've had this dream before. Oh wait a minute, that was about Single Side
Band :-)

Charlie.

--
Hello Wisconsin!

En el artículo , FranK Turner-Smith
G3VKI escribió:

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.

One word: compression.

If we all took the same attitude as the OP we'd still be using Strowger
exchanges for the phone system. Thankfully, some people understand the
need for progress and aren't stuck in the past.

Times have changed and left me behind, but I've still got me beer so who
cares?

Good man

--
:: je suis Charlie :: yo soy Charlie :: ik ben Charlie ::

"Mike Tomlinson" wrote in message
...

If we all took the same attitude as the OP we'd still be using Strowger
exchanges for the phone system. Thankfully, some people understand the
need for progress and aren't stuck in the past.

There you go again with your one-sided infantile outbursts.

"gareth" wrote in message
...
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?

A very well informed and authoratative response to this has been made
by "Channel Jumper" over in rec.radio.amateur.equipment, although
he did not elect to post it here as well.
..

"Jerry Stuckle" wrote in message
...
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").

Thanks, Jerry, you've explained a lot to me, and in a manner that an old
fart like me can understand.
I appreciate that, and feel I owe you a pint, (not a compressed 1cc one).
--
;-)
..
73 de Frank Turner-Smith G3VKI - mine's a pint.
..
http://turner-smith.co.uk

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

A friend worked in sonar where the data was collected on ships and
transmitted via satellite to shore for signal processing rather than
doing any compression on the data and sending the useful info. As the
signal was nearly all "noise" trying to do any compression on it, even
the aspects that weren't "pure" white noise, would potentially have
masked the signals. Sonar is all about pulling the signal out of the
noise.

--

Rick

On 2/24/2015 12:10 PM, Brian Morrison wrote:
On Tue, 24 Feb 2015 16:32:21 -0000
FranK Turner-Smith G3VKI wrote:

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.

Have a look at David Rowe's web site about Codec 2 and his work on it.

http://rowetel.com

Most of the codec development effort goes into voice modelling that
allows redundant information to be thrown away without making the
encoded speech sound too horrible when decoded. And on working out
which bits in the encoded frame need to be better protected and which
don't, this is especially important when considering what encodes
voiced and non-voiced speech and ensuring it doesn't get mixed up.

Other than uLaw/Alaw, voice for telephony is not compressed in the same
ways that a zip file is. As you say, they model the vocal tract and
send the parameters for the sounds that are to be produced along with
error information to make it more intelligible. For sounds that aren't
voice or voice like, they are reproduced poorly. This is why low bit
rate compression on cell phones doesn't convey music very well and
background noise messes up the intelligibility much more than with uLaw
or ALaw compression which are just ways of compressing the waveform
without knowing anything about the content.

--

Rick

On 2/24/2015 1:28 PM, gareth wrote:
"Mike Tomlinson" wrote in message
...

If we all took the same attitude as the OP we'd still be using Strowger
exchanges for the phone system. Thankfully, some people understand the
need for progress and aren't stuck in the past.

There you go again with your one-sided infantile outbursts.

lol!

--

Rick

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"???

--

Rick