On 2/24/2015 7:12 PM, Jerry Stuckle wrote:
On 2/24/2015 7:03 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.

Why does it not "exist"? That is not at all clear. You don't
understand compression. Compression is a means of removing the part of
a signal that is unimportant and sending only the part that is
important. In most cases of "pure" noise, you can just send a statement
that the signal is "noise" without caring about the exact voltages over
time. So, yes, even noise can be compressed depending on your
requirements.


Pure white noise is a concept only. There is no perfect white noise
source, just as there is no pure resistor or capacitor.

And yes, I do understand compression. One of the things it depends on
is predictability and repeatability of the incoming signal. That does
not exist with white noise. The fact you don't understand that pure
white noise is only a concept and cannot exist in the real world shows
your lack of understanding.

This is not very productive. You make an assertion and the fact that I
don't agree means I am wrong. Ok, you have an idea in your mind and
can't explain it. I get that. The fact that you don't have a white
noise source in your lab doesn't mean it doesn't exist other than in the
same way that 100.1 doesn't exist. No one has ever made anything that
was *exactly* 100.1.

This is a pointless abstraction so I won't continue to debate it.


Some compression algorithms (i.e. mp3) remove what they consider is
"unimportant". However, the result after decompressing is a poor
recreation of the original signal.

That is a value judgement which most would disagree with not to mention
that your example is not valid. MP3 does not *remove* anything from the
signal. It is a form of compression that simply can't reproduce the
signal exactly. The use of the term "poor" is your value judgement.
Most people would say an MP3 audio sounds very much like the original.


But for perfect recreation, nothing is "unimportant". Voice/video
compression is no different than file compression on a computer. Can
you imaging what would happen if your favorite program was not perfectly
recreated?


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.


You mean the signal can't be compressed? No way. Any non-random signal
can be compressed to some extent. How much depends on the signal and
the amount of processing power required to compress it. However, in
your example, the processing power to compress the signal would probably
have been greater than that required to process the original signal. So
if there wasn't enough power to process the signal on the ship, there
wouldn't be enough power to compress the near-white noise signal, either.

You really like your all encompassing assumptions. No, all signals can
not be compressed, even non-noise signals can't be compressed if the
signal is not appropriate for the compressor. This is really a very
large topic and I think you are used to dealing with the special cases
without understanding the general case.


Which is just the opposite of what you claimed above. Please make up
your mind.

This is the sort of stuff that makes discussions with you unenjoyable.
You clearly don't understand compression or you would understand this
statement. Compression maps a combination of bits into a smaller number
of bits. By the counting theorem it is impossible for any compression
algorithm to compress all possible input sets. Whether it can be
compressed depends on a match between the input bits and the compression
algorithm. Even white noise (which can exist if you define "white
noise" adequately) can be compressed by the appropriate algorithm. That
algorithm won't compress much else though.


Try visiting comp.compression and offering them your opinions. There
are many there who are happy to explain the details to you.


I understand the details, thank you. Much better than you do,
obviously. But that's not surprising, either.

Ok, you have reverted into snarky mode. I'm done.

--

Rick

On 2/25/2015 1:41 AM, rickman wrote:
On 2/24/2015 7:12 PM, Jerry Stuckle wrote:
On 2/24/2015 7:03 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.

Why does it not "exist"? That is not at all clear. You don't
understand compression. Compression is a means of removing the part of
a signal that is unimportant and sending only the part that is
important. In most cases of "pure" noise, you can just send a statement
that the signal is "noise" without caring about the exact voltages over
time. So, yes, even noise can be compressed depending on your
requirements.


Pure white noise is a concept only. There is no perfect white noise
source, just as there is no pure resistor or capacitor.

And yes, I do understand compression. One of the things it depends on
is predictability and repeatability of the incoming signal. That does
not exist with white noise. The fact you don't understand that pure
white noise is only a concept and cannot exist in the real world shows
your lack of understanding.

This is not very productive. You make an assertion and the fact that I
don't agree means I am wrong. Ok, you have an idea in your mind and
can't explain it. I get that. The fact that you don't have a white
noise source in your lab doesn't mean it doesn't exist other than in the
same way that 100.1 doesn't exist. No one has ever made anything that
was *exactly* 100.1.

This is a pointless abstraction so I won't continue to debate it.


You obviously again have no idea what you're talking about. By
definition, white noise is a concept only and CAN'T EXIST in the real
world. It's similar to an isotropic source.


Some compression algorithms (i.e. mp3) remove what they consider is
"unimportant". However, the result after decompressing is a poor
recreation of the original signal.

That is a value judgement which most would disagree with not to mention
that your example is not valid. MP3 does not *remove* anything from the
signal. It is a form of compression that simply can't reproduce the
signal exactly. The use of the term "poor" is your value judgement.
Most people would say an MP3 audio sounds very much like the original.


That is a value judgement that all experts agree with - and an area I
have been intimately involved with for the last 13 years. You also
don't understand how mp3 works.

All experts agree that when comparing mp3 to the original, there is a
significant difference.


But for perfect recreation, nothing is "unimportant". Voice/video
compression is no different than file compression on a computer. Can
you imaging what would happen if your favorite program was not perfectly
recreated?


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.


You mean the signal can't be compressed? No way. Any non-random signal
can be compressed to some extent. How much depends on the signal and
the amount of processing power required to compress it. However, in
your example, the processing power to compress the signal would
probably
have been greater than that required to process the original
signal. So
if there wasn't enough power to process the signal on the ship, there
wouldn't be enough power to compress the near-white noise signal,
either.

You really like your all encompassing assumptions. No, all signals can
not be compressed, even non-noise signals can't be compressed if the
signal is not appropriate for the compressor. This is really a very
large topic and I think you are used to dealing with the special cases
without understanding the general case.


Which is just the opposite of what you claimed above. Please make up
your mind.

This is the sort of stuff that makes discussions with you unenjoyable.
You clearly don't understand compression or you would understand this
statement. Compression maps a combination of bits into a smaller number
of bits. By the counting theorem it is impossible for any compression
algorithm to compress all possible input sets. Whether it can be
compressed depends on a match between the input bits and the compression
algorithm. Even white noise (which can exist if you define "white
noise" adequately) can be compressed by the appropriate algorithm. That
algorithm won't compress much else though.


I understand compression much better than you do. And not everything
can be compressed - there is a limit. White noise is one of the things
which cannot be compressed.


Try visiting comp.compression and offering them your opinions. There
are many there who are happy to explain the details to you.


I understand the details, thank you. Much better than you do,
obviously. But that's not surprising, either.

Ok, you have reverted into snarky mode. I'm done.


That's good. Trying to educate you is like trying to teach a pig to sing.

--
==================
Remove the "x" from my email address
Jerry, AI0K

==================

That's good. Trying to educate you is like trying to teach a pig to sing.


and I thought brian's put downs were good.........

On 25/02/2015 13:45, Jerry Stuckle wrote:
On 2/25/2015 1:41 AM, rickman wrote:

That is a value judgement which most would disagree with not to mention
that your example is not valid. MP3 does not *remove* anything from the
signal. It is a form of compression that simply can't reproduce the
signal exactly. The use of the term "poor" is your value judgement.
Most people would say an MP3 audio sounds very much like the original.


That is a value judgement that all experts agree with - and an area I
have been intimately involved with for the last 13 years. You also
don't understand how mp3 works.

All experts agree that when comparing mp3 to the original, there is a
significant difference.

I think that there is a semantics issue here.

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.

Andy

On 2/26/2015 3:55 AM, AndyW wrote:
On 25/02/2015 13:45, Jerry Stuckle wrote:
On 2/25/2015 1:41 AM, rickman wrote:

That is a value judgement which most would disagree with not to mention
that your example is not valid. MP3 does not *remove* anything from the
signal. It is a form of compression that simply can't reproduce the
signal exactly. The use of the term "poor" is your value judgement.
Most people would say an MP3 audio sounds very much like the original.


That is a value judgement that all experts agree with - and an area I
have been intimately involved with for the last 13 years. You also
don't understand how mp3 works.

All experts agree that when comparing mp3 to the original, there is a
significant difference.

I think that there is a semantics issue here.

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.

Andy


Andy,

You are really trying to split hairs here. The data are lost because
they are "removed" during compression. It is an active decision as to
what is compressed and what is ignored.

And yes, the term "removed" is used when describing the technical
aspects of MP3 compression.

--
==================
Remove the "x" from my email address
Jerry, AI0K

==================

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).
--
Home, is where I park it.

---
This email has been checked for viruses by Avast antivirus software.
http://www.avast.com

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.

--
==================
Remove the "x" from my email address
Jerry, AI0K

==================

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

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.

--
==================
Remove the "x" from my email address
Jerry, AI0K

==================

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

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.


You are partially right.. CD uses a specific sample rate and bit size,
And I believe you are correct as to what they are

MP3 can use a very wide range of sample rates, and different bit sizes
as well.... Crank the bit rate and sample size up enough and yes, I am
not going to be able to tell the difference. (A golden ear I'm not).

But the bandwith needed goes way up.

Some "Cred"info..My daughter, in whom I am well pleased, IS, among other
things, a Classical Musician and music teacher...In the past I have been
her "Recording Engineer" I also do recording in other venues as well...
Usualy ATRAC, but sometimes CD quality.

I have played a lot with Bit Rates, sample sizes and sample rates.using
Total Recorder PRO and the LAME codec.

Fun Fact: Remember Spaceship One, the one that won the X-Prize? Well,
Dr Space (David Livingston) complained during the first flight about
having to change cassettes in his audio recorder... I tossed Total
Recorder on the job and within minutes of the program end he had an MP3
in his mail box.

Next flight he mentioned his new log recorder (Total Recorder PRO) and
credited me with the suggestion.

I have hundreds (IF not thousands) of hours of live recordings lying
about here.

--
Home, is where I park it.

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