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--- In softrock40@..., w keith griffith <kgriffit@w...>
wrote:

I have,,, 'SomeWhere' a really good description of the Weaver ssb
method.

There's a decent description of the Weaver method in the DSP section
of all the recent ARRL Handbooks.

How, if at all, does that method compare with the
softrock?

The SoftRock itself does no demodulation. All it does is sample the
incoming signal in such a way as to mix down to 11250Hz. The
downconverted signal is then dumped to the computer's soundcard.

In the subsequent software DSP on the digitized signal, the SSB
"demodulation" consists entirely of a bandpass filtering stage and
another mixing stage to spin the signal centered at 11025Hz down to
0Hz.

CW is the same, except the mixing stage spins the signal down to the
CW offset frequency (700Hz or whatever) rather than 0Hz.

73
Frank
AB2KT

Thanks,,, very fun following the list. I'm not likely to get involved with this type project at the hardware level, my current interests are mostly tube gear, but current interests do change over time ;-)

The SoftRock itself doesn't convert all input signal to 11,250Hz, it has a fixed crystal oscillator, were the signal ends up depends where in the pass-band the original signal is. Some will be at 1KHz some at 5KHz, some at 19KHz depends on how far they are from the center of the pass-band (7.056MHz). A QRP signal a 7.040 will end up at 16KHz.

On the SDR1000 it's another story, they have a agile oscillator so they try to keep the signal you are tuned to at 11KHz away from the crud at at 0Hz.

At 11:26 AM 9/29/2005, you wrote:

--- In softrock40@..., w keith griffith <kgriffit@w...>
wrote:

I have,,, 'SomeWhere' a really good description of the Weaver ssb
method.

There's a decent description of the Weaver method in the DSP section
of all the recent ARRL Handbooks.

How, if at all, does that method compare with the
softrock?

The SoftRock itself does no demodulation. All it does is sample the
incoming signal in such a way as to mix down to 11250Hz. The
downconverted signal is then dumped to the computer's soundcard.

In the subsequent software DSP on the digitized signal, the SSB
"demodulation" consists entirely of a bandpass filtering stage and
another mixing stage to spin the signal centered at 11025Hz down to
0Hz.

CW is the same, except the mixing stage spins the signal down to the
CW offset frequency (700Hz or whatever) rather than 0Hz.

73
Frank
AB2KT







Yahoo! Groups Links

Cecil Bayona
KD5NWA
www.qrpradio.com

'Never argue with an idiot. They drag you down to their level then beat you with experience.'

On 9/29/05, Frank Brickle <ab2kt@...> wrote:

The SoftRock itself does no demodulation. All it does is sample the
incoming signal in such a way as to mix down to 11250Hz. The
downconverted signal is then dumped to the computer's soundcard.

Frank,

Looking at the picture:



I couldn't understand how the mixing down to the magic number 11250 is
obtained.. Would you mind explaining it a bit more?

What I thought was that the signal at 7.056 gets downconverted to zero
IF and because of the soundcard sampling rate of 48 khz, we are able
to tune +-24khz.. I guess there is some issue in my understanding.
WIll be great if you explain it to me and others.

TIA
--
73 - Ramakrishnan, VU3RDD

Cecil,

I just had a look at the powerSDR-SR40 code. Looking at sdr.c line, I
can see that the signal is indeed at 11025 Hz. The RX oscillator runs
at -11025 hz. Now, does that mean, the spectrum we can indeed tune to
is centered around 7056-11.025 = 7044.9 khz? I am a bit confused here.

On 9/29/05, KD5NWA <KD5NWA@...> wrote:

The SoftRock itself doesn't convert all input signal to 11,250Hz, it
has a fixed crystal oscillator, were the signal ends up depends where
in the pass-band the original signal is. Some will be at 1KHz some at
5KHz, some at 19KHz depends on how far they are from the center of
the pass-band (7.056MHz). A QRP signal a 7.040 will end up at 16KHz.

On the SDR1000 it's another story, they have a agile oscillator so
they try to keep the signal you are tuned to at 11KHz away from the
crud at at 0Hz.

At 11:26 AM 9/29/2005, you wrote:

--- In softrock40@..., w keith griffith <kgriffit@w...>
wrote:

I have,,, 'SomeWhere' a really good description of the Weaver ssb
method.

There's a decent description of the Weaver method in the DSP section
of all the recent ARRL Handbooks.

How, if at all, does that method compare with the
softrock?

The SoftRock itself does no demodulation. All it does is sample the
incoming signal in such a way as to mix down to 11250Hz. The
downconverted signal is then dumped to the computer's soundcard.

In the subsequent software DSP on the digitized signal, the SSB
"demodulation" consists entirely of a bandpass filtering stage and
another mixing stage to spin the signal centered at 11025Hz down to
0Hz.

CW is the same, except the mixing stage spins the signal down to the
CW offset frequency (700Hz or whatever) rather than 0Hz.

73
Frank
AB2KT







Yahoo! Groups Links

Cecil Bayona
KD5NWA
www.qrpradio.com

'Never argue with an idiot. They drag you down to their level then
beat you with experience.'






Yahoo! Groups Links

--
73 - Ramakrishnan, VU3RDD

The previous message that I answered was talking about the hardware (SR-40), you are talking about the software, two different subjects.

The previous message is talking about the signal being at 11KHz then it's read by the sound card, your talking about what happens after the sound card has read the signals.

Sorry for the confusion but we are talking about different things.


At 12:55 PM 9/29/2005, you wrote:

Cecil,

I just had a look at the powerSDR-SR40 code. Looking at sdr.c line, I
can see that the signal is indeed at 11025 Hz. The RX oscillator runs
at -11025 hz. Now, does that mean, the spectrum we can indeed tune to
is centered around 7056-11.025 = 7044.9 khz? I am a bit confused here.

On 9/29/05, KD5NWA <KD5NWA@...> wrote:

The SoftRock itself doesn't convert all input signal to 11,250Hz, it
has a fixed crystal oscillator, were the signal ends up depends where
in the pass-band the original signal is. Some will be at 1KHz some at
5KHz, some at 19KHz depends on how far they are from the center of
the pass-band (7.056MHz). A QRP signal a 7.040 will end up at 16KHz.

On the SDR1000 it's another story, they have a agile oscillator so
they try to keep the signal you are tuned to at 11KHz away from the
crud at at 0Hz.

At 11:26 AM 9/29/2005, you wrote:

--- In softrock40@..., w keith griffith <kgriffit@w...>
wrote:

I have,,, 'SomeWhere' a really good description of the Weaver ssb
method.

There's a decent description of the Weaver method in the DSP section
of all the recent ARRL Handbooks.

How, if at all, does that method compare with the
softrock?

The SoftRock itself does no demodulation. All it does is sample the
incoming signal in such a way as to mix down to 11250Hz. The
downconverted signal is then dumped to the computer's soundcard.

In the subsequent software DSP on the digitized signal, the SSB
"demodulation" consists entirely of a bandpass filtering stage and
another mixing stage to spin the signal centered at 11025Hz down to
0Hz.

CW is the same, except the mixing stage spins the signal down to the
CW offset frequency (700Hz or whatever) rather than 0Hz.

73
Frank
AB2KT







Yahoo! Groups Links

Cecil Bayona
KD5NWA
www.qrpradio.com

'Never argue with an idiot. They drag you down to their level then
beat you with experience.'






Yahoo! Groups Links

--
73 - Ramakrishnan, VU3RDD




Yahoo! Groups Links

Cecil Bayona
KD5NWA
www.qrpradio.com

'Never argue with an idiot. They drag you down to their level then beat you with experience.'

Agree. Why I talked about software is that the only way I can conclude
the center frequency is 11.025khz is by looking at what the software
is trying to do. I do not yet have SR40 hardware beside me to verify
physically. So my reference to software was intentional.

Anyway, that does not address my doubt..

On 9/29/05, KD5NWA <KD5NWA@...> wrote:

The previous message that I answered was talking about the hardware
(SR-40), you are talking about the software, two different subjects.

The previous message is talking about the signal being at 11KHz then
it's read by the sound card, your talking about what happens after
the sound card has read the signals.

Sorry for the confusion but we are talking about different things.


At 12:55 PM 9/29/2005, you wrote:

Cecil,

I just had a look at the powerSDR-SR40 code. Looking at sdr.c line, I
can see that the signal is indeed at 11025 Hz. The RX oscillator runs
at -11025 hz. Now, does that mean, the spectrum we can indeed tune to
is centered around 7056-11.025 = 7044.9 khz? I am a bit confused here.

On 9/29/05, KD5NWA <KD5NWA@...> wrote:

The SoftRock itself doesn't convert all input signal to 11,250Hz, it
has a fixed crystal oscillator, were the signal ends up depends where
in the pass-band the original signal is. Some will be at 1KHz some at
5KHz, some at 19KHz depends on how far they are from the center of
the pass-band (7.056MHz). A QRP signal a 7.040 will end up at 16KHz.

On the SDR1000 it's another story, they have a agile oscillator so
they try to keep the signal you are tuned to at 11KHz away from the
crud at at 0Hz.

At 11:26 AM 9/29/2005, you wrote:

--- In softrock40@..., w keith griffith <kgriffit@w...>
wrote:

I have,,, 'SomeWhere' a really good description of the Weaver ssb
method.

There's a decent description of the Weaver method in the DSP section
of all the recent ARRL Handbooks.

How, if at all, does that method compare with the
softrock?

The SoftRock itself does no demodulation. All it does is sample the
incoming signal in such a way as to mix down to 11250Hz. The
downconverted signal is then dumped to the computer's soundcard.

In the subsequent software DSP on the digitized signal, the SSB
"demodulation" consists entirely of a bandpass filtering stage and
another mixing stage to spin the signal centered at 11025Hz down to
0Hz.

CW is the same, except the mixing stage spins the signal down to the
CW offset frequency (700Hz or whatever) rather than 0Hz.

73
Frank
AB2KT







Yahoo! Groups Links

Cecil Bayona
KD5NWA
www.qrpradio.com

'Never argue with an idiot. They drag you down to their level then
beat you with experience.'






Yahoo! Groups Links

--
73 - Ramakrishnan, VU3RDD




Yahoo! Groups Links

Cecil Bayona
KD5NWA
www.qrpradio.com

'Never argue with an idiot. They drag you down to their level then
beat you with experience.'






Yahoo! Groups Links

--
73 - Ramakrishnan, VU3RDD

--- In softrock40@..., Ramakrishnan Muthukrishnan
<vu3rdd@g...> wrote:

I couldn't understand how the mixing down to the magic number 11250

is

obtained.. Would you mind explaining it a bit more?

Sorry, typo. it's 11025. I got it right in a subsequent paragraph :-)

What I thought was that the signal at 7.056 gets downconverted to

zero

IF and because of the soundcard sampling rate of 48 khz, we are able
to tune +-24khz..

11025 * 2 = 22050.
22050 * 2 = 44100.

11025Hz is the center of the presumed "good" passband on typical
soundcards capable of handling audio CD rates.

You don't want to downconvert to 0 because most soundcards have
bandpass filtering that roll off towards 0 and the Nyquist frequency.
If you downconverted to 0, you'd be filtering off most of the signal
before it got to your signal processing.

Therefore you downconvert such that the signal of interest is in the
"best" part of the soundcard passband.

The subsequent mixing stage (complex oscillator at -11025 or whatever)
then moves the signal down to 0 once it's in digital form. Then the
remainder of the processing takes place.

73
Frank
AB2KT

But on the SR-40 the oscillator is fixed! if you want to listen to 7.058 you have 2KHz audio coming out, the only thing you can do is change the crystal (good luck) or use a DDS instead of the built in fixed crystal.

If you want to listen to a signal on 7.0563, good luck!

Now, if you are changing the crystal, try to get the frequency of interest away from 0 Hz.


At 08:44 AM 9/30/2005, you wrote:

--- In softrock40@..., Ramakrishnan Muthukrishnan
<vu3rdd@g...> wrote:

I couldn't understand how the mixing down to the magic number 11250

is

obtained.. Would you mind explaining it a bit more?

Sorry, typo. it's 11025. I got it right in a subsequent paragraph :-)

What I thought was that the signal at 7.056 gets downconverted to

zero

IF and because of the soundcard sampling rate of 48 khz, we are able
to tune +-24khz..

11025 * 2 = 22050.
22050 * 2 = 44100.

11025Hz is the center of the presumed "good" passband on typical
soundcards capable of handling audio CD rates.

You don't want to downconvert to 0 because most soundcards have
bandpass filtering that roll off towards 0 and the Nyquist frequency.
If you downconverted to 0, you'd be filtering off most of the signal
before it got to your signal processing.

Therefore you downconvert such that the signal of interest is in the
"best" part of the soundcard passband.

The subsequent mixing stage (complex oscillator at -11025 or whatever)
then moves the signal down to 0 once it's in digital form. Then the
remainder of the processing takes place.

73
Frank
AB2KT







Yahoo! Groups Links

Cecil Bayona
KD5NWA
www.qrpradio.com

'Never argue with an idiot. They drag you down to their level then beat you with experience.'

On 9/30/05, Frank Brickle <ab2kt@...> wrote:

--- In softrock40@..., Ramakrishnan Muthukrishnan

11025 * 2 = 22050.
22050 * 2 = 44100.

11025Hz is the center of the presumed "good" passband on typical
soundcards capable of handling audio CD rates.

You don't want to downconvert to 0 because most soundcards have
bandpass filtering that roll off towards 0 and the Nyquist frequency.
If you downconverted to 0, you'd be filtering off most of the signal
before it got to your signal processing.

Therefore you downconvert such that the signal of interest is in the
"best" part of the soundcard passband.

Thanks Frank. That's quite clear.

Is it correct to assume that "if there is no sound card, and if we
have a higher sampling rate ADC, we can see more portion of the band"?
Or rephrasing it this way - What would be the passband of the signal
output from the QSD ?

--
73 - Ramakrishnan, VU3RDD

--- In softrock40@..., Ramakrishnan Muthukrishnan
<vu3rdd@g...> wrote:

Is it correct to assume that "if there is no sound card, and if we
have a higher sampling rate ADC, we can see more portion of the band"?
Or rephrasing it this way - What would be the passband of the signal
output from the QSD ?

These are somewhat independent issues. It is certainly true that the
higher the sampling rate (~ADC freq) the more of the band you could
see, potentially. However the output bandwidth of the QSD really
depends on the specific components in the QSD. What it is in the case
of the SoftRock I do not know.

One of the clever features of the QSD is that the effective Q of the
sampling filters is constant over the tuning range of the QSD. Not
something that affects the SR40 much, of course :-)

73
Frank
AB2KT

On 9/30/05, Frank Brickle <ab2kt@...> wrote:

--- In softrock40@..., Ramakrishnan Muthukrishnan
These are somewhat independent issues. It is certainly true that the
higher the sampling rate (~ADC freq) the more of the band you could
see, potentially. However the output bandwidth of the QSD really
depends on the specific components in the QSD. What it is in the case
of the SoftRock I do not know.

One of the clever features of the QSD is that the effective Q of the
sampling filters is constant over the tuning range of the QSD. Not
something that affects the SR40 much, of course :-)

Thanks a lot, Frank, for patiently explaining it.

--
73 - Ramakrishnan, VU3RDD

This is not what we do on the Softrock40. The 11 kHz IF is used in the SDR-1000 in order to do exactly what Frank says and that is avoid 1/F, DC, junk, noise, etc. at 0. We are attempting to maximize the performance.


On the softrock40, we have one oscillator on the board and it is not steerable. It provides a 48 kHz (theoretically) wide signal. We want to be able to tune ANYWHERE in that band. So we in fact live with the little hump of nasty at 0 frequency and tune the software oscillator from -24 to 24 kHz around the center frequency. We get single sideband demodulation by using a complex tap weight based bandpass filter using a very clever mathematical trick which allows us to do huge, fantastic filters with little more computation than much smaller (and poorer filters) done by the normal convolution.
In the newer Windows release which should be out some time next week (depending no Eric's work schedule), PowerSDR will carry SR40 support in the base Windows code, and you can select SDR-1000, Softrock 40, and DEMO (no hardware required) mode from the setup panel after install or the installation wizard during install.

Bob
N4HY


Frank Brickle wrote:

--
Laziness is the number one inspiration for ingenuity. Guilty as charged!

Bob,

Thanks a lot. I plan to replace the crystal with my NJQRP DDS for the
time being. Thanks again. This thread enlightened and increased my
understanding of basics.

73
Ramakrishnan, VU3RDD

On 10/1/05, Robert McGwier <rwmcgwier@...> wrote:

This is not what we do on the Softrock40. The 11 kHz IF is used in the
SDR-1000 in order to do exactly what Frank says and that is avoid 1/F,
DC, junk, noise, etc. at 0. We are attempting to maximize the performance.


On the softrock40, we have one oscillator on the board and it is not
steerable. It provides a 48 kHz (theoretically) wide signal. We want
to be able to tune ANYWHERE in that band. So we in fact live with the
little hump of nasty at 0 frequency and tune the software oscillator
from -24 to 24 kHz around the center frequency. We get single sideband
demodulation by using a complex tap weight based bandpass filter using a
very clever mathematical trick which allows us to do huge, fantastic
filters with little more computation than much smaller (and poorer
filters) done by the normal convolution.
In the newer Windows release which should be out some time next week
(depending no Eric's work schedule), PowerSDR will carry SR40 support
in the base Windows code, and you can select SDR-1000, Softrock 40,
and DEMO (no hardware required) mode from the setup panel after install
or the installation wizard during install.

Bob
N4HY


Frank Brickle wrote:

--- In softrock40@..., Ramakrishnan Muthukrishnan
<vu3rdd@g...> wrote:

I couldn't understand how the mixing down to the magic number 11250

is

obtained.. Would you mind explaining it a bit more?

Sorry, typo. it's 11025. I got it right in a subsequent paragraph :-)

What I thought was that the signal at 7.056 gets downconverted to

zero

IF and because of the soundcard sampling rate of 48 khz, we are able
to tune +-24khz..

11025 * 2 = 22050.
22050 * 2 = 44100.

11025Hz is the center of the presumed "good" passband on typical
soundcards capable of handling audio CD rates.

You don't want to downconvert to 0 because most soundcards have
bandpass filtering that roll off towards 0 and the Nyquist frequency.
If you downconverted to 0, you'd be filtering off most of the signal
before it got to your signal processing.

Therefore you downconvert such that the signal of interest is in the
"best" part of the soundcard passband.

The subsequent mixing stage (complex oscillator at -11025 or whatever)
then moves the signal down to 0 once it's in digital form. Then the
remainder of the processing takes place.

73
Frank
AB2KT







Yahoo! Groups Links

--
Laziness is the number one inspiration for ingenuity. Guilty as charged!





Yahoo! Groups Links

--
73 - Ramakrishnan, VU3RDD

--- In softrock40@..., Robert McGwier <rwmcgwier@c...>
wrote:

This is not what we do on the Softrock40. The 11 kHz IF is used in

the

SDR-1000 in order to do exactly what Frank says and that is avoid

1/F,

DC, junk, noise, etc. at 0. We are attempting to maximize the

performance.

Hello Bob,

Could you resume in a few sentences the difference between the
SoftRock-40 and the SDR-1000 in this regard?
I could go back to the articles on the SDR-1000 and try to figure it
out, but I think that you can give a better answer.

The softRock-40 has a fixed oscillator and can software-tune +/- 24
kHz around that frequency which means that near the oscillator
frequency we get "junk". With the SDR-1000 what happens when we tune
a particular frequency?

Jean-Claude Abauzit, PJ2BVU

When tuning an SDR 1000, if spur reduction is turned off, the DDS is tuner 11025 khz below the frequency of interest and the software (DttSP) oscillator is set to -11025 khz to recover the frequency of interest. This works well but for the spurs from the DDS.

If spur reduction is turned on (the normal case) the DDS is tuned to the nearest low spur point (as documented by some of the AD docs) and then DttSP oscillator is set to an appropriate value to recover the signal of interest. This generally results in a DttSP oscillator value between -9.5 and -12.5 khz.

On the SoftRock all the tuning is done with the DttSP oscillator and it ranges over +/- 24 khz - giving a tuning range of 48 khz.

Regards,

Bill (kd5tfd)