What is the point of digital voice?
 

On 2/25/2015 5:37 PM, gareth wrote:

Here is your big chance to prove your superiority of knowledge about
the super-regrenerative method, but you've gone strangely silent, which
is a bit bizarre when you consider how many times you have oft
repeated your childish sneer?



Perhaps you will listen to the voice of expierence.

My first receiver was a Knight Kit Star Roamer.. now this is a superhet,
true, but as it turns out it had a REGEN control in one stage, that
stage could be made super regenerative,, You used this to receive CW or
SSB,, i used that radio for many years.

But the fact is.. It worked,, NOT as well as a modern well filtered
Superhet,, But that has a lot to do with the Filters more than the
receiver's other parts.

I would not mind getting another of those.. Nostalga value and all that.
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What is the point of digital voice?
 

On 2/26/2015 3:55 AM, AndyW wrote:

MP3 is lossy, it cannot be used to reproduce the original but it does
not 'remove' signal, they get lost.

IIRC some sound encoding deliberately removes some frequencies if the
are low amplitude and are close to a higher amplitude frequency.

Loses is passive, the data just gets lost. Remove implies some active
removal of data.

All of what you type is true yet MP3 is good enough for most music
lovers (The true "Golden Ears" do not like it but not many are that
good) I can occasionaly hear the difference but not always.

The major advantage of digital over analog modulation is that the
computer's "ears" (The de-mod unit) are way more discreaning than my ears.

First. Under noisy low signal conditions,,, Most of the noise is lost
simply because it is not present at the proper time,, With analog none
of it is lost you need to spend heavy duty effort to filter it out.. But
with DSP you look for 1 or zero at the right time, noise that happens
when you are not looking... is ignored.. And with protocol some errors
caused by noise get corrected, others can not be but in some cases a
re-peat of the packet is requested and delivered.

Far less power is needed to make the trip,, Digital signals can travel
farther on less power all because of the above. It truly is an amazing
way to chat,, I have used both digital and analog or many years, and
where as with analog, as the sigal goes down the amount of operator
skill to hear the voice goes up, way up, and more and more folks start
wonering what it is I am hearing, cause they sure can not hear it, but I
seem to be writing down good inormation.

With digital you are there, or you are not, and "There" means it sounds
like you are sitting beside me. (Perhaps that is why I operate SSB, I
like to keep the skills honed a bit).
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What is the point of digital voice?
 

"John Davis" wrote in message
...
On 2/25/2015 5:37 PM, gareth wrote:

Here is your big chance to prove your superiority of knowledge about
the super-regrenerative method, but you've gone strangely silent, which
is a bit bizarre when you consider how many times you have oft
repeated your childish sneer?

Perhaps you will listen to the voice of expierence.

My first receiver was a Knight Kit Star Roamer.. now this is a superhet,
true, but as it turns out it had a REGEN control in one stage, that stage
could be made super regenerative,, You used this to receive CW or SSB,, i
used that radio for many years.

I fear that you will be incorrect and confusing regeneration and
super-regeneration.

What is the point of digital voice?
 

On Fri, 6 Mar 2015, gareth wrote:

"John Davis" wrote in message
...
On 2/25/2015 5:37 PM, gareth wrote:

Here is your big chance to prove your superiority of knowledge about
the super-regrenerative method, but you've gone strangely silent, which
is a bit bizarre when you consider how many times you have oft
repeated your childish sneer?

Perhaps you will listen to the voice of expierence.

My first receiver was a Knight Kit Star Roamer.. now this is a superhet,
true, but as it turns out it had a REGEN control in one stage, that stage
could be made super regenerative,, You used this to receive CW or SSB,, i
used that radio for many years.

I fear that you will be incorrect and confusing regeneration and
super-regeneration.

I almost missed it. No, he's talking about a superhet with standard
455Khz IF, where some feedback was added around an IF stage (usually a
"gimmick" capacitor so one can adjust it), and with control of the
cathode, one could increase selectivity and put it into oscillation so
there was something to beat against the incoming signals to demodulate CW
and SSB. But that's really just a more complicated method of regeneration
and superregeneration.

One of the problems with superregenerative receivers is that they were
long treated as a black box. ONce they fell out of leading edge circuity
(where they helped to homestead the higher bands), people forgot how they
worked and the book descriptions were pretty uninformative. I remember
one ARRL Handbook going into how the same active device could be the
receiver and the quenching oscillator, without explaining what the
quenching oscillator did.

That said, a superregenerative receiver is just a superset of a
regenerative receiver. Armstrong came up with the latter early on,
patented in 1914. It showed not only how to make a better receiver, but
how to make a tube oscillate, real cutting edge. Then later, when he was
on the eve of a court case over that regen patent, he went back to the
regen to remind himself about its operation, and came across a phenomena
that he'd noticed almost a decade earlier, but hadn't pursued. This was
superregeneration, and it happened with a regular regen receiver. It's
just kicking things further along. I'm sure some circuits are better to
get the quenching, but if you view the superregen as a regen receiver with
exteral quenching oscillator, it's all so much easier to visualize. The
quenching modulates the regen. If it's one device, the one device does
both, it's just a matter of getting the quenching going.

So the same receiver can be both. Indeed, in the late fifties or early
sixties, the ARRL had a popular VHF station construction series, using a
14MHz regen and converters. And they even say by adjuting regen, you can
use the receiver as a superregen.

You can't use superregeneration for receiving SSB and CW, but you can use
the same circuit, so long as it can be adjusted through regeneration to
actual feedback and beyond.

Michael

What is the point of digital voice?
 

On 3/6/2015 1:06 PM, John Davis wrote:
On 2/26/2015 3:55 AM, AndyW wrote:

MP3 is lossy, it cannot be used to reproduce the original but it does
not 'remove' signal, they get lost.

IIRC some sound encoding deliberately removes some frequencies if the
are low amplitude and are close to a higher amplitude frequency.

Loses is passive, the data just gets lost. Remove implies some active
removal of data.

All of what you type is true yet MP3 is good enough for most music
lovers (The true "Golden Ears" do not like it but not many are that
good) I can occasionaly hear the difference but not always.


Both MP3 and CD use 16/44 (16 bits, 44kHz sample rate) formats. The
difference is that the CD will have the entire signal stored, while MP3
will remove some of the signal which is not as important as others.

If you play an MP3 and a CD on any decent (not even audiophile)
equipment, the difference is noticeable, even to a non-audiophile. And
the difference between MP3 and high resolution 24/192 is even greater if
you're playing music with wide frequency and volume ranges, such as much
classical music. But you won't hear that much of a difference between
MP3 and 24/192 on a many rock songs :)

The major advantage of digital over analog modulation is that the
computer's "ears" (The de-mod unit) are way more discreaning than my ears.


Computers are lousy playback mechanisms. The frequency response of the
amplifier is nowhere near flat, and the speakers generally stink. It
would be better if you hooked up a decent set of stereo speakers - but
even then a cheap amplifier will outperform virtually any computer.

First. Under noisy low signal conditions,,, Most of the noise is lost
simply because it is not present at the proper time,, With analog none
of it is lost you need to spend heavy duty effort to filter it out.. But
with DSP you look for 1 or zero at the right time, noise that happens
when you are not looking... is ignored.. And with protocol some errors
caused by noise get corrected, others can not be but in some cases a
re-peat of the packet is requested and delivered.


Noise is like any other part of the signal. If you have a 1kHz noise
spike, it will be present for approximately 1ms. That is plenty long
for any ADC to detect it. And if the noise pulse is shorter than the
sampling time, it would be of too high of a frequency to hear, anyway.

Plus, DSPs do not look at amplitude. They measure the instantaneous
slope of the signal and store it as a digital value depending on the
number of bits, i.e. 16 bit samples would have 2^15 negative slope
values and 2^15-1 positive slope values (plus zero slope). By
recreating the instantaneous slope that is stored digitally, the DAC
converts the digital signal back to an analog signal.

And digital error-correction protocols have nothing to do with the
digital signal itself - only the transmission of it from one system to
another. But that is an entirely different subject.

Far less power is needed to make the trip,, Digital signals can travel
farther on less power all because of the above. It truly is an amazing
way to chat,, I have used both digital and analog or many years, and
where as with analog, as the sigal goes down the amount of operator
skill to hear the voice goes up, way up, and more and more folks start
wonering what it is I am hearing, cause they sure can not hear it, but I
seem to be writing down good inormation.


Yes, I understand that. I was working RTTY back in the 60's, and it was
amazing how you could get good copy on a signal you couldn't even hear
in the noise. Of course, the narrow filters used on the audio signal
made a big difference - just like a narrow filter helps pull a CW signal
out of the mud.

With digital you are there, or you are not, and "There" means it sounds
like you are sitting beside me. (Perhaps that is why I operate SSB, I
like to keep the skills honed a bit).

Yes and no. Digital does for the most part work or not work. However,
when you get into marginal conditions, it can get iffy, with some
packets lost and not recoverable.

Probably the easiest way to see this is watching a digital TV signal.
When the signal becomes marginal, the picture will start to display junk
in random small spots on the screen, similar to snow (known as
pixelation). Satellite TV users have seen it during heavy rain, and
even cable TV users can see it when a network's satellite link suffers
from a marginal signal.

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What is the point of digital voice?
 

On 3/6/2015 3:11 PM, Jerry Stuckle wrote:

Plus, DSPs do not look at amplitude. They measure the instantaneous
slope of the signal and store it as a digital value depending on the
number of bits, i.e. 16 bit samples would have 2^15 negative slope
values and 2^15-1 positive slope values (plus zero slope). By
recreating the instantaneous slope that is stored digitally, the DAC
converts the digital signal back to an analog signal.

This is just plain wrong. I'm not sure why you make a distinction
between DSP's [sic] and any other digital device since a DSP is not
needed at all to digitize or compress a signal, but the sample produced
by an ADC *is* the instantaneous value of the signal and not the slope.
If you were to compare adjacent ADC samples and calculate the slope
that would be a form of ADPCM. The DAC in turn converts this
instantaneous value back into analog followed by filtering to remove the
higher frequency images if important.

--

Rick

What is the point of digital voice?
 

"Michael Black" wrote in message
news:alpine.LNX.2.02.1503061451360.32579@darkstar. example.org...
On Fri, 6 Mar 2015, gareth wrote:
I fear that you will be incorrect and confusing regeneration and
super-regeneration.

I almost missed it. No, he's talking about a superhet with standard
455Khz IF, where some feedback was added around an IF stage (usually a
"gimmick" capacitor so one can adjust it), and with control of the
cathode, one could increase selectivity and put it into oscillation so
there was something to beat against the incoming signals to demodulate CW
and SSB. But that's really just a more complicated method of regeneration
and superregeneration.


He is discussing a regenerative IF detector, but not a superregenerative one
where the feedback is increased well past the point of oscillation to give
very high gain. There would not have been a quenching oscillator in what
he described.

The quencher acts like a balanced modulator onto the oscillatory stage to
remove the presence of the on-channel carrier out to two sidebands distanced
away by the quench frequency, which is why the super-regenerative technique
does not resolve SSB and CW.

What is the point of digital voice?
 

On 3/6/2015 3:48 PM, rickman wrote:
On 3/6/2015 3:11 PM, Jerry Stuckle wrote:

Plus, DSPs do not look at amplitude. They measure the instantaneous
slope of the signal and store it as a digital value depending on the
number of bits, i.e. 16 bit samples would have 2^15 negative slope
values and 2^15-1 positive slope values (plus zero slope). By
recreating the instantaneous slope that is stored digitally, the DAC
converts the digital signal back to an analog signal.

This is just plain wrong. I'm not sure why you make a distinction
between DSP's [sic] and any other digital device since a DSP is not
needed at all to digitize or compress a signal, but the sample produced
by an ADC *is* the instantaneous value of the signal and not the slope.
If you were to compare adjacent ADC samples and calculate the slope
that would be a form of ADPCM. The DAC in turn converts this
instantaneous value back into analog followed by filtering to remove the
higher frequency images if important.


Once again you are wrong, Rick. Integrating ADCs have been used at
least since the 70's and are much more accurate and noise immune than a
simple level ADC. ADPCM isn't even closely related.

And I mentioned DSPs because that is what John was discussing.

--
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Remove the "x" from my email address
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==================

What is the point of digital voice?
 

On 3/6/2015 5:13 PM, Jerry Stuckle wrote:
On 3/6/2015 3:48 PM, rickman wrote:
On 3/6/2015 3:11 PM, Jerry Stuckle wrote:

Plus, DSPs do not look at amplitude. They measure the instantaneous
slope of the signal and store it as a digital value depending on the
number of bits, i.e. 16 bit samples would have 2^15 negative slope
values and 2^15-1 positive slope values (plus zero slope). By
recreating the instantaneous slope that is stored digitally, the DAC
converts the digital signal back to an analog signal.

This is just plain wrong. I'm not sure why you make a distinction
between DSP's [sic] and any other digital device since a DSP is not
needed at all to digitize or compress a signal, but the sample produced
by an ADC *is* the instantaneous value of the signal and not the slope.
If you were to compare adjacent ADC samples and calculate the slope
that would be a form of ADPCM. The DAC in turn converts this
instantaneous value back into analog followed by filtering to remove the
higher frequency images if important.


Once again you are wrong, Rick. Integrating ADCs have been used at
least since the 70's and are much more accurate and noise immune than a
simple level ADC. ADPCM isn't even closely related.

I'm only going to point out your error and then I won't argue with you
further. No one is talking about integrating ADCs. You said, "They
measure the instantaneous slope of the signal and store it as a digital
value". That is not what an integrating ADC does, nor does any other ADC.

The integrating ADC uses the input to charge up a capacitance (the
integrator) for some period of time, then a reference is used to
discharge the "integrated" voltage and the time this takes is measured.
This is *not* measuring the "instantaneous slope" of the input signal.
In fact "integrating" and "instantaneous" are contradictory since
"integrating" takes time and "instantaneous" is... well, instantaneous.

Also I will mention that although integrating ADCs are good for noise
rejection, they are *very* slow and only used in such low sample rate
apps as volt meters and the like. More accurate systems like weight
scales typically use sigma-delta converters for low noise, low power and
high resolution or in the case of and high end audio sigma-delta
converters offer high linearity and low distortion.

I think one reason integrating converters are used in volt meters is
that they can be designed to always display 0 for a 0 input voltage
which is important to consumer confidence.

ADPCM is a form of compression comparing adjacent ADC samples to
calculate the differential of the signal which is the closest thing to
what you are describing by the "instantaneous slope".

--

Rick

What is the point of digital voice?
 

On 3/7/2015 2:55 AM, rickman wrote:
On 3/6/2015 5:13 PM, Jerry Stuckle wrote:
On 3/6/2015 3:48 PM, rickman wrote:
On 3/6/2015 3:11 PM, Jerry Stuckle wrote:

Plus, DSPs do not look at amplitude. They measure the instantaneous
slope of the signal and store it as a digital value depending on the
number of bits, i.e. 16 bit samples would have 2^15 negative slope
values and 2^15-1 positive slope values (plus zero slope). By
recreating the instantaneous slope that is stored digitally, the DAC
converts the digital signal back to an analog signal.

This is just plain wrong. I'm not sure why you make a distinction
between DSP's [sic] and any other digital device since a DSP is not
needed at all to digitize or compress a signal, but the sample produced
by an ADC *is* the instantaneous value of the signal and not the slope.
If you were to compare adjacent ADC samples and calculate the slope
that would be a form of ADPCM. The DAC in turn converts this
instantaneous value back into analog followed by filtering to remove the
higher frequency images if important.


Once again you are wrong, Rick. Integrating ADCs have been used at
least since the 70's and are much more accurate and noise immune than a
simple level ADC. ADPCM isn't even closely related.

I'm only going to point out your error and then I won't argue with you
further. No one is talking about integrating ADCs. You said, "They
measure the instantaneous slope of the signal and store it as a digital
value". That is not what an integrating ADC does, nor does any other ADC.

The integrating ADC uses the input to charge up a capacitance (the
integrator) for some period of time, then a reference is used to
discharge the "integrated" voltage and the time this takes is measured.
This is *not* measuring the "instantaneous slope" of the input signal.
In fact "integrating" and "instantaneous" are contradictory since
"integrating" takes time and "instantaneous" is... well, instantaneous.

Also I will mention that although integrating ADCs are good for noise
rejection, they are *very* slow and only used in such low sample rate
apps as volt meters and the like. More accurate systems like weight
scales typically use sigma-delta converters for low noise, low power and
high resolution or in the case of and high end audio sigma-delta
converters offer high linearity and low distortion.

I think one reason integrating converters are used in volt meters is
that they can be designed to always display 0 for a 0 input voltage
which is important to consumer confidence.

ADPCM is a form of compression comparing adjacent ADC samples to
calculate the differential of the signal which is the closest thing to
what you are describing by the "instantaneous slope".


Sorry - I used the wrong term. The integration is done by the DAC, to
invert the actions of the ADC.

But no, if you understood ANY calculus, you would understand that
"integrating" and "instantaneous" are not contradictory. But then
"instantaneous" is only a theoretical concept, not possible in the real
world. But the word is still in common usage. I wonder why that is?

ADPCM (Adaptive Differential Pulse Code Modulation) is something
completely different.

Slope ADCs are used because they can more accurately recreate the
waveform. To make it simple - let's see the ADC is sampling at twice
the frequency being sampled, i.e. 10kHz signal and 20kHz sampling rate.
If the sample happens to be at the zero crossing point, your ADC will
show zero volts - IOW, no signal. But a slope detecting ADC will show a
fairly high positive slope on one sample and an equally negative slope
on the next sample. By integrating these, the DAC can closely recreate
the signal because it can estimate the maximum amplitude by the slopes
of the samples. No, it won't be perfect - but it will be a lot closer
than your simple ADC.

Now I know you're going to find all kinds of problems with this example
- but I made the example simple so that even you might be able to
understand it. As you increase the sample rate relative to the
frequency of the signal being sampled, the difference becomes less. But
the slope detecting ADC will always provide a more accurate signal
(until you get to an infinitely small sample anyway). The math is
somewhat complex, and I'm sure beyond anything you could possibly
understand. But it can be proven.

As for them not existing. I guess the whole quarter we spent on ADCs in
my EE classes were wrong then. Of course, this was over 40 years ago.
But I doubt physics has changed in that time.

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