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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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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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.

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

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

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

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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Jerry, AI0K

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

On 3/7/2015 11:35 AM, Jerry Stuckle wrote:

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.


I don't enjoy discussing things with you because you have to make
everything personal. But I will explain the fallacy of your argument on
the Nyquist sampling rate concept.

You pick a sampling point for the dual slope, integrating converter that
happens to give valid results. But if you shift the phase by 90° so
that this converter sees positive values half the integrating period and
negative values for the other half, it produces all zero samples as
well. So there is really no difference in the two converters regarding
Nyquist rate sampling. It merely depends on the phasing of the sample
clock to the input signal. It also depends on how you define the
"sample point" of an integrating converter, the start, the end or the
middle of the integration period.

I will finally point out that your use of the term "slope detecting ADC"
is invalid. Google returns exactly 4 hits when this term is entered
with quotes. The name of this converter may have slope in it, but that
is because the circuit generates a slope, not because it is detecting a
slope. Please look up the circuit and use a proper name for it such as
integrating ADC or dual slope ADC. The integrating converter is not at
all sensitive to the slope of the input signal, otherwise it would not
be able to measure a DC signal which has a slope of zero.

I'm only replying so that others are not confused by your misstatements.

--

Rick

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.

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

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).

I just got my Digital equipment back in operating condition (Had several
issues, New computer lacked an AUDIO IN jack (Now have a dongle) Dongle
needed audio on the RING, not the TIP, I had used a mono plug (no ring)
Bad transformer (replaced with a better one)

When I say JUST... I mean yesterday

But back before the old computer died this is what I observed with
digital text

1: Signals I Could not hear (and I have good hearing) It could often decode.

2: Most signals either decoded...Or not... But sometimes.... IT was broken.


From other expierence, both with digital video and audio

As the signal degrades with analog it is 10,9,8,7 and so on

With Digital it is 10,10,10,10,10,10,10,5,0

(NOTE: Number of 10's is not to scale)
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