K3 2.7kHz filter DSP FC

[Barry N1EU]

"If the center frequency was set to 1.225 KHz (or close) as you suggest, then
the 2.7 KHz filter would let though too much opposite-sideband energy. We
want some stopping power down close to zero Hertz, so the filter edge is
kept 250 Hz above zero."

I don't understand a need to lock the lower filter edge above 250hz
for opposite sideband rejection.  The dsp filtering slope is very
steep.  You can set the Orion lower filter edge to 50hz and it sounds
great.

73,
Barry N1EU
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[P.B. Christensen]

> "If the center frequency was set to 1.225 KHz (or close) as you suggest,
> then
the 2.7 KHz filter would let though too much opposite-sideband energy. We
want some stopping power down close to zero Hertz, so the filter edge is
kept 250 Hz above zero."

Bruce, as Barry points out, the K3's audio response really needs the ability
to extend down significantly lower than the current ~ 250 Hz high-pass
cut-off, especially when the 6 kHz filter is employed.  I would recommend 50
Hz, although 70 Hz would be fine if there is some inherent limitation that
precludes a lower value on Rx and Tx.  Since we're not concerned with the
broadcast of music in our service, then the male voice (lowest relevant
fundamental of 70 Hz) and certain digital modes in SSB mode (e.g., PSK31)
may be used to establish the lower response threshold.

Transceivers such as the Orion, Omni VII, Icom Pro Series, Yaesu FT-2K/950
series, IC-7800...all provide reasonably good Rx and TX response to at least
100 Hz, some of these mentioned go even lower.  However, the benchmark of
all time goes to the Kenwood TS-870S, where I've measured flat low-end Rx
and Tx response down to 10 Hz (@ -3dB) and the opposite sideband remains
completely filtered out.   For example, if you tune into a carrier in SSB
mode, you will hear that carrier down to near DC, then when you approach the
other side of zero, absolutely nothing is heard.  That's one Helluva' design
accomplishment.  But obviously, that kind of low-end response is overkill.
On Tx, I've actually heard room rumble coming from my modified D-104.

So, to be competitive with these other radios, this is an area that needs
further refinement.  And, it is really my only significant adverse issue
with the K3 at the moment (okay, other than placement of the REV button).
Once you've listened to SSB or CW on a clear, open band, the audible
difference between a lower limit of 250 Hz and 70 Hz is absolutely amazing.
It really does offer a whole new dimension to the listening experience.

Paul, W9AC

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[David Woolley (E.L)]

Paul Christensen wrote:
> Bruce, as Barry points out, the K3's audio response really needs the
> ability to extend down significantly lower than the current ~ 250 Hz

> we're not concerned with the broadcast of music in our service, then the
> male voice (lowest relevant fundamental of 70 Hz) and certain digital

Billions of people are quite happy with the 300kHz cutoff used by the
public telephone system (a total of 3.1kHz between 300 and 3.4kHz).  The
critical frequency for speech communications is the lowest formant
frequency; it is the resonances in the vocal cavity that convey speech,
not the fundamental.

The lowest formant in normal speech (formant 1 for "u") spans about 200
to 400Hz, so 250Hz probably is a reasonable compromise; it will include
the peak and only slightly distort the lower edge.


> modes in SSB mode (e.g., PSK31) may be used to establish the lower
> response threshold.

When you use digital modes with the receiver configured for SSB, you are
really using the audio path as a final IF of about 1kHz.  As long as the
  audio passband is flat across the actual digital signal, it doesn't
matter what its centre frequency (final final IF frequency) is.

That's basically the strategy used by telephone modems (except 56kbs
ones, which play tricks with the digitisation of the signal).  The
faster ones use a carrier of about 1800Hz (nominal channel centre is
1850 Hz).


--
David Woolley
"The Elecraft list is a forum for the discussion of topics related to
Elecraft products and more general topics related ham radio"
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[P.B. Christensen]

> Billions of people are quite happy with the 300kHz cutoff used by the
> public telephone system (a total of 3.1kHz between 300 and 3.4kHz).

Presumably, you mean 300 Hz, and not 300 kHz.

> The critical frequency for speech communications is the lowest formant
> frequency; it is the resonances in the vocal cavity that convey speech,
> not the fundamental.

You're referring to audio passband in which minimum bandwidth is used as a
function of maximum articulation.  Bell Labs and other acoustical
researchers in the late 19th and early 20th centuries found that reasonable
bandwidth for voice communication was in the area you describe.  The concept
was further carried on with the advent of HF SSB transmissions in the late
'40s, although AT&T Long Lines was already economizing SSB bandwidth well
before then.

As a practical matter, analog circuits in which crystal filters are used
during SSB generation (e.g., Collins) precluded audio transmission below 300
Hz.  But we're not talking about land-line voice circuits here and
technology has marched-on well past the usage of  SSB-generation techniques
that rely on crystal filtering for economy.  Through DSP, we easily have the
ability to extend low frequency content down well below 300 Hz and a
significant list of other tranceiver manufacturers are accommodating this
demand in the amateur radio community.

> The lowest formant in normal speech (formant 1 for "u") spans about 200 to
> 400Hz, so 250Hz probably is a reasonable compromise; it will include the
> peak and only slightly distort the lower edge.

It may be a compromise, but a considerable amount of voice detail occurs
below 250 Hz, even though it offer little or nothing to intelligibility and
articulation.  Using a microphone, a sound card, and FFT software (e.g.,
SpectraPlus) observe the extent of low frequency content of you own voice.
If you believe that relevant energy is limited to 250 Hz, you need to
research this in better detail.  We already have reasonably good
articulation in the bandwidth between 300 Hz and ~ 2.8 kHz but little added
bandwidth is required to add nearly two acoustical octaves to the listening
experience.

Incidentally, an argument can be made that to maximize intelligibility, the
upper passband should actually be closer to 3.5 kHz or even 4.0 kHz (see a
graph of the now-classic Fletcher-Munson family of loudness curves and
re-determined in later years by modern researchers).  Beyond that limit,
diminishing returns on intelligibility occur.  The reason the ear is most
sensitive in this area is that the ear canal forms a closed pipe against the
tympanic membrane.  The resulting fundamental frequency and harmonics can
then be calculated with precision by the width of the ear canal and its
length.  Of further interest is that maximum energy in the human voice is
also matched-up against the curves.  So, either millions of years of
evolution or God made this association possible.  Take your pick.

The key point is that compromises must be made againt the current
state-of-the-art.  It is my belief that the benefit of adding a Rx/Tx range
down to ~ 70 Hz greatly outweighs the relatively little extra bandwidth
needed (less than 200 Hz of B/W) to make a profound difference in the
listening experience.

Paul, W9AC

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[Doug Alspaugh]

Besides considering how nice it sounds we need to be considering how wide our signals are on the band.Rielly (sp) of FCC fame has warned the ham community about being careful with the width of our signals when we are trying to transmit "Hifi" sounding transmissions.Plus it's just the neighborly thing to try not to interfere with others.


73 Doug K3DUG
K3 #????


 -------------- Original message ----------------------
From: "Paul Christensen" <[hidden email]>

> > Billions of people are quite happy with the 300kHz cutoff used by the
> > public telephone system (a total of 3.1kHz between 300 and 3.4kHz).
>
> Presumably, you mean 300 Hz, and not 300 kHz.
>
> > The critical frequency for speech communications is the lowest formant
> > frequency; it is the resonances in the vocal cavity that convey speech,
> > not the fundamental.
>
> You're referring to audio passband in which minimum bandwidth is used as a
> function of maximum articulation.  Bell Labs and other acoustical
> researchers in the late 19th and early 20th centuries found that reasonable
> bandwidth for voice communication was in the area you describe.  The concept
> was further carried on with the advent of HF SSB transmissions in the late
> '40s, although AT&T Long Lines was already economizing SSB bandwidth well
> before then.
>
> As a practical matter, analog circuits in which crystal filters are used
> during SSB generation (e.g., Collins) precluded audio transmission below 300
> Hz.  But we're not talking about land-line voice circuits here and
> technology has marched-on well past the usage of  SSB-generation techniques
> that rely on crystal filtering for economy.  Through DSP, we easily have the
> ability to extend low frequency content down well below 300 Hz and a
> significant list of other tranceiver manufacturers are accommodating this
> demand in the amateur radio community.
>
> > The lowest formant in normal speech (formant 1 for "u") spans about 200 to
> > 400Hz, so 250Hz probably is a reasonable compromise; it will include the
> > peak and only slightly distort the lower edge.
>
> It may be a compromise, but a considerable amount of voice detail occurs
> below 250 Hz, even though it offer little or nothing to intelligibility and
> articulation.  Using a microphone, a sound card, and FFT software (e.g.,
> SpectraPlus) observe the extent of low frequency content of you own voice.
> If you believe that relevant energy is limited to 250 Hz, you need to
> research this in better detail.  We already have reasonably good
> articulation in the bandwidth between 300 Hz and ~ 2.8 kHz but little added
> bandwidth is required to add nearly two acoustical octaves to the listening
> experience.
>
> Incidentally, an argument can be made that to maximize intelligibility, the
> upper passband should actually be closer to 3.5 kHz or even 4.0 kHz (see a
> graph of the now-classic Fletcher-Munson family of loudness curves and
> re-determined in later years by modern researchers).  Beyond that limit,
> diminishing returns on intelligibility occur.  The reason the ear is most
> sensitive in this area is that the ear canal forms a closed pipe against the
> tympanic membrane.  The resulting fundamental frequency and harmonics can
> then be calculated with precision by the width of the ear canal and its
> length.  Of further interest is that maximum energy in the human voice is
> also matched-up against the curves.  So, either millions of years of
> evolution or God made this association possible.  Take your pick.
>
> The key point is that compromises must be made againt the current
> state-of-the-art.  It is my belief that the benefit of adding a Rx/Tx range
> down to ~ 70 Hz greatly outweighs the relatively little extra bandwidth
> needed (less than 200 Hz of B/W) to make a profound difference in the
> listening experience.
>
> Paul, W9AC
>
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[Barry N1EU]

dalspaugh wrote

Besides considering how nice it sounds we need to be considering how wide our signals are on the band.

The signal is 2.7Khz wide regardless of where Fc is.  We're not talking about wider signals, we're talking about enabling a downward shift of the passband.  Besides, this thread was about receive, not transmit.

73,
Barry N1EU

[Jay Bromley]

Hi Guys,

I know I am a couple days late on this, but I have been wanting to comment
on this audio thing.

Billions of folks may be content with restricted audio, but I wouldn't be.
Let me first state I do a ton of listening or monitoring on all modes.
Right now I am on 7.020 using a wide filter set for 3.6 KHz listening to CW
stations coming and going as I work in the shack.  A also have a few AM
programs on the weekend I like to listen to.  With my outboard audio amp and
bookshelf speaker system it reminds me of the old days when REAL
communication receivers had room filling volume.

Let me move away from phone operation and talk digital, I do a little of
that as well.  Using programs such as HRD's DM780 and MixW if you have a
receiver rolling off the low or high end response you will loose the ability
to monitor more effectively on modes such as PSK31.  Now for you guys that
can only think narrow as well as listen that way, let me explain what I do
in my shack.  When I am monitoring the bands on PSK31, I use the widest
filter possible.  At the extremes of the rig's bandpass their maybe a
problem on transmitting a very low or high tone.  However both programs
mentioned above have a centering function that grabs the signal and puts it
at 1500 hz.  If you are using CAT with  your rig, this is all automatic, it
centers the signal right on 1500 Hz and adjust the VFO accordingly.  Say I
find a brand new country and find him at 100 Hz using DM780, I hit "Center"
and it moves the station of interests to 1500 Hz.  Then I can point and
click for a narrower receive filter.  I am now ready to transmit.

If you have ever gone fishing, you catch more fish with a large net than you
do with a small one.  You can always throw back in the small ones or
unwanted ones.  If your receiver is restricted, you will and can miss
stations that might interest you!  Operators that use DM780 with
SuperBrowser know what I am talking about.  Just for fun I decide to see how
many stations I would missed with a rolled off response on the low end of
300 hz, typical "communication grade" receiver of yester year.  I fired up
non communication grade receiver and set the cursor at 70.1 hz, I still got
good print on PSK31!  So using a channel width of 30 to 50 hz on PSK31 means
I could  miss up to 7 to 4 station respectively.   This doesn't even count
for the higher end response!  For those that would like to see, I have jpg's
of stations received at 70.1 Hz and 3604.3 Hz that printed cleanly I can
send to those that are interested.  I have other receivers that could do
better on the high end, but that is not the point I am making here.

Over the years I have a ton of receivers, to be honest I would not know how
to classify a "communications grade" receiver.  When I hear "communications
grade" I think heavy, military like ruggedness, and versatility!   I never
think 300-300 KHz, but that is just me.

So end the end, this cool aide drinker would like a choice of favors, the
more versatility, the better for me.  I hope in the end the K3 has the
ability to go wide on IF/audio frequency response and then let me choose how
narrow I need to go on all modes.  There a lot of reasons to have a nice
flat, wide, low distortion bandpass, not just for good sound, but that would
be great also!

I would be curious to what the K3 ultimate bandpass would be and would that
also be completely adjustable?  Yes I did glance at the various pdf.s, but
didn't see a spec for this.

Thanks for letting me add my 2 cents worth.

73 de w5jay/jay..

> Paul Christensen wrote:
>> Bruce, as Barry points out, the K3's audio response really needs the
>> ability to extend down significantly lower than the current ~ 250 Hz
>
>> we're not concerned with the broadcast of music in our service, then the
>> male voice (lowest relevant fundamental of 70 Hz) and certain digital
>
> Billions of people are quite happy with the 300kHz cutoff used by the
> public telephone system (a total of 3.1kHz between 300 and 3.4kHz).  The
> critical frequency for speech communications is the lowest formant
> frequency; it is the resonances in the vocal cavity that convey speech,
> not the fundamental.
>
> The lowest formant in normal speech (formant 1 for "u") spans about 200 to
> 400Hz, so 250Hz probably is a reasonable compromise; it will include the
> peak and only slightly distort the lower edge.
>
>
>> modes in SSB mode (e.g., PSK31) may be used to establish the lower
>> response threshold.
>
> When you use digital modes with the receiver configured for SSB, you are
> really using the audio path as a final IF of about 1kHz.  As long as the
> audio passband is flat across the actual digital signal, it doesn't matter
> what its centre frequency (final final IF frequency) is.
>
> That's basically the strategy used by telephone modems (except 56kbs ones,
> which play tricks with the digitisation of the signal).  The faster ones
> use a carrier of about 1800Hz (nominal channel centre is 1850 Hz).
>
>
> --
> David Woolley

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