[Flexradio] SSB signals reversed

[SteveLornaF]

I am a new user of the SDR-1000 and can only recieve signals on their  
opposite sideband (i.e. 80M on USB not LSB), can anyone help to correct the  
problem?
 
Using the latest 1.61 release software.
 
Thanks
 
Steve - G1LMN
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Re: [Flexradio] vCOM Dowwnload Not Working

[Philip Covington]

On 5/20/06, John Denson <[EMAIL PROTECTED]> wrote:
> Phil Covington - I notice that the vCOM "Download" button on your web site
> is inoperative.
>
> John Denson
>

It works and I haven't changed anything for months.   You might want
to try again.

Phil C

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[Flexradio] vCOM Dowwnload Not Working

[John Denson]

Phil Covington - I notice that the vCOM "Download" button on your web site 
is inoperative.

John Denson

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[Flexradio] The Flex Suite @ Dayton is now on TeamSpeak

[Tim Ellison]

Eric was able to get "free" wireless for the TeamSpeak connection.  How
resourceful!

-Tim
---
Tim Ellison  
Integrated Technical Services   
Apex, NC USA
919.674.0044 Ext. 25 / 919.674.0045 (FAX)
919.215.6375 - cell
>>> PGP public key available at all public KeyServers <<<
Skype: kg4rzy




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Re: [Flexradio] 16 versus 24 bit audio

[Larry Gadallah]

Message: 39
> Date: Sat, 20 May 2006 16:09:52 -
> From: "Peter Martinez" <[EMAIL PROTECTED]>
> Subject: Re: [Flexradio] 16 versus 24 bit audio
> To: "Flex Reflector" 
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
> reply-type=response
>
> >From G3PLX:
>
> Sami is right. You CAN recover a signal from noise by narrowing the
> bandwidth, so if my signal was level with the noise in 24kHz bandwidth, I
> could filter it to a narrower bandwidth (in software) and improve it's
> SNR.



I've been wondering about this lately; Isn't this getting close to what the
definition of "process gain" is? Incidentally, Analog Devices has some
excellent white papers that go into the details of the tradeoffs between
sensitivity, dynamic range, resolution, etc. Definitely recommended reading.


But that's true regardless of the number of bits in the raw data. So long as
> we have at least half an lsb of additive noise to dither away the
> quantisation problem, we can always gain SNR by reducing the downstream
> software bandwidth. Any more added noise at the front end than this only
> makes things - well - er - noisier.  The noise power calculation I did
> earlier shows that there is ALREADY about the right amount of noise
> inherent
> in the physics to get the dither optimum for a 24-bit ADC at 0dBm.  If
> there
> are cards with more noise than this, it's probably because the designer
> couldn't get it any lower for the price, not because he chose to add more
> noise for some subtle reason.
>
> To take Jim's points, I have only so far measured the rms noise with no
> input, and not yet looked to see if there are any clues in it's spectrum -
> these things would take a lot more time. Clock jitter wouldn't explain
> what
> I see (an output with no input) although it would certainly cause noise in
> the presence of a large pure tone. I haven't tried that yet either.



It would be interesting to see the spectral characteristics of the noise.

Let me close this topic before Phil accuses me of cruelty to dead horses.
> Before I aquired a 24-bit card, I honestly believed that 24-bit cards
> would
> be 8 bits better than 16 bit cards. When I did get one recently, I was
> surprised to find this wasn't the case. Jim is right. 24 bit cards may
> only
> be slightly better than 16-bit cards. I have learned something this week.
>

Seems to me that the marketing folks take the number of bits in the ADC and
run with it. A 16 bit card is only 14 bits, and a 24 bit card is only 18
bits, so the 24 bit card is 4 bits better than the 16 bit card.

Cheers,
-- 
Larry Gadallah, VE6VQ/W7
lgadallah AT gmail DOT com
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Re: [Flexradio] 16 versus 24 bit audio

[Ahti Aintila]

Thanks to everybody for the most interesting and educating discussion.
Nevertheless, I am not happy until the JANUS version with AK5394A is
in my hands and I have modified the QSD and the following amplifier.

Best regards and special thanks to the HPSDR group for the good work
done so far,
Ahti OH2RZ


On 20/05/06, Philip Covington <[EMAIL PROTECTED]> wrote:
> On 5/20/06, Peter Martinez <[EMAIL PROTECTED]> wrote:
> > >From G3PLX:
>
> > Let me close this topic before Phil accuses me of cruelty to dead horses.
> > Before I aquired a 24-bit card, I honestly believed that 24-bit cards would
> > be 8 bits better than 16 bit cards. When I did get one recently, I was
> > surprised to find this wasn't the case. Jim is right. 24 bit cards may only
> > be slightly better than 16-bit cards. I have learned something this week.
> >
> > 73
> > Peter
>
> Hi Peter,
>
> Well, I may have been a little too strong in making that statement
> (dead horse)...it was early morning here...no coffee yet consumed...
> etc... ;-)
>
> The FlexRadio Forum has some interesting discussions in the past about
> different sound cards and what to expect.  It is pretty much true that
> some 24 bit cards are marginally better than some 16  bit cards.  It
> also would be accurate to say that some 24 bit cards are worse than
> some of the better 16 bit cards.
>
> 73 de Phil N8VB
>
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Re: [Flexradio] 16 versus 24 bit audio

[Philip Covington]

On 5/20/06, Peter Martinez <[EMAIL PROTECTED]> wrote:
> >From G3PLX:

> Let me close this topic before Phil accuses me of cruelty to dead horses.
> Before I aquired a 24-bit card, I honestly believed that 24-bit cards would
> be 8 bits better than 16 bit cards. When I did get one recently, I was
> surprised to find this wasn't the case. Jim is right. 24 bit cards may only
> be slightly better than 16-bit cards. I have learned something this week.
>
> 73
> Peter

Hi Peter,

Well, I may have been a little too strong in making that statement
(dead horse)...it was early morning here...no coffee yet consumed...
etc... ;-)

The FlexRadio Forum has some interesting discussions in the past about
different sound cards and what to expect.  It is pretty much true that
some 24 bit cards are marginally better than some 16  bit cards.  It
also would be accurate to say that some 24 bit cards are worse than
some of the better 16 bit cards.

73 de Phil N8VB

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Re: [Flexradio] Dayton

[Mike Naruta]

Welcome Cecil, it's a great radio and an
amazing group.  It puts the fun back into
ham radio.


Mike - AA8K

cbayona wrote:
> My plans were that shortly after Dayton I would purchase a SDR-1000, but
> since I could not purchase the signal generator, and they had SDR-1000's
> in stock, I went ahead and purchased one.
> 
> This was late yesterday as Dayton was closing so I dind't have a chance
> to purchase a small power supply and cables so I can play with it while
> on vacation. That will be taken care of shortly when I head for the bone
> yard.
> 
> The weather yesterday was cloudy in the morning and sunny after 10:00AM,
> today is sunny also but it might rain in the afternoon.
> 
> See ya in the funny papers.

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Re: [Flexradio] 16 versus 24 bit audio

[Peter Martinez]

>From G3PLX:

Sami is right. You CAN recover a signal from noise by narrowing the 
bandwidth, so if my signal was level with the noise in 24kHz bandwidth, I 
could filter it to a narrower bandwidth (in software) and improve it's SNR.

But that's true regardless of the number of bits in the raw data. So long as 
we have at least half an lsb of additive noise to dither away the 
quantisation problem, we can always gain SNR by reducing the downstream 
software bandwidth. Any more added noise at the front end than this only 
makes things - well - er - noisier.  The noise power calculation I did 
earlier shows that there is ALREADY about the right amount of noise inherent 
in the physics to get the dither optimum for a 24-bit ADC at 0dBm.  If there 
are cards with more noise than this, it's probably because the designer 
couldn't get it any lower for the price, not because he chose to add more 
noise for some subtle reason.

To take Jim's points, I have only so far measured the rms noise with no 
input, and not yet looked to see if there are any clues in it's spectrum - 
these things would take a lot more time. Clock jitter wouldn't explain what 
I see (an output with no input) although it would certainly cause noise in 
the presence of a large pure tone. I haven't tried that yet either.

Let me close this topic before Phil accuses me of cruelty to dead horses. 
Before I aquired a 24-bit card, I honestly believed that 24-bit cards would 
be 8 bits better than 16 bit cards. When I did get one recently, I was 
surprised to find this wasn't the case. Jim is right. 24 bit cards may only 
be slightly better than 16-bit cards. I have learned something this week.

73
Peter
 


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Re: [Flexradio] 16 versus 24 bit audio

[Jim Lux]

At 04:35 AM 5/20/2006, Peter Martinez wrote:
>from G3PLX:
>
>Hello Sami. Thanks for joining the discussion - I feared I was on my own for
>a while!  I hope some others will contribute too, so I don't get accused of
>hogging the bandwidth (or perhaps generating too much noise!).
>
>You said:
> >If your low 8 bits are truly random with no input, that's a good
> >thing! It follows that when you inject a non-random signal, it will
> >stand out clearly from white noise.
>
>Let me be simple-minded and respond:-
>
>But if my low 8 bits are truly random and I inject an 8-bit sinewave, isn't
>it then level with the noise? If I could reduce my 8 bits of added noise to,
>say, 4 bits, wouldn't my sinewave then be 24dB above the noise?  I don't see
>how adding that much noise can ever be a good thing to do.  I am not trying
>to trap you, I really would like to understand what's going on here. This is
>all new stuff to me.
>
>I had a private email from a broadcast engineer who confirmed what I was
>saying about dither noise. He said they added 0.5 bits-worth of white noise
>to the analogue signal before digitisation, and subjectively that was the
>best result.
>
>But that's half a bit of dither, not 8 bits.


One might add a few more bits of noise if one wasn't very sure of one's A/D 
linearity performance, and if it didn't impair the overall performance.  If 
you're oversampling, for instance, and going to reduce a 96 ksps down to, 
say, 24 ksps, the power from the added noise would be reduced by the 
"averaging" you're doing with the sample rate reduction.  Considering that 
most people will be filtering the output of the A/D to, say, 20Hz-20Khz, 
this might not be a bad design tradeoff.




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Re: [Flexradio] Dayton

[Larry Taft]

Cecil,
> so I can play with it while on vacation. 

OXYMORON  OXYMORON OXYMORON

SDR \= vacation

You will spend ALL your time messing with the weird, wild, wonderous 
wonderful world of SDR-1000 and have no time for "vacation".

Welcome to the crowd.

73, Larry K2LT

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Re: [Flexradio] 16 versus 24 bit audio

[Jim Lux]

At 02:07 AM 5/20/2006, Peter Martinez wrote:
> >From G3PLX:
>
>
>I just did some proper measurements on the Firebox. On runs of 1200 samples
>at 48kHz, the r.m.s. noise level on the line input is about 400. It's about
>450 on the Mike input.  I did this by squaring each sample, summing these,
>dividing by 1200, and taking the square root. At the same time I calculated
>the DC level, just to make sure that was a lot smaller and didn't confuse
>things.  This is about 9 bits of noise, or more if you suppose the peaks are
>higher.
>
>On the built-in 16-bit card in this laptop the DC level is 15 and I am not
>sure how to allow for this in the r.m.s calculation, but only the lsb
>toggles randomly. The MP3+ is the same, unless I select the mike input when
>the rms noise level is about 10.  These figures are what I would have
>expected intuitively. One would expect the analogue noise-level to be
>roughly the same as the quantisation noise, in the same way that one would
>optimise any receiver so that all sources of noise contribute more-or-less
>the same in order to maximise the dynamic range.

Calculate the average of all the samples, then subtract that from the 
samples, then square, sum, and root.  Or, if you have a statistics package, 
calculate the standard deviation of the input samples.  That will be the 
rms noise voltage.


Of more interest is actually running a clean sine wave in, and then looking 
at the rms noise power as a function of the digitized value.  Analog input 
noise shows up as noise that is independent of where in the cycle it's 
sampled.  Clock jitter shows up as increased noise near the zero crossings 
(where the derivative of the input is highest).



James Lux, P.E.
Spacecraft Radio Frequency Subsystems Group
Flight Communications Systems Section
Jet Propulsion Laboratory, Mail Stop 161-213
4800 Oak Grove Drive
Pasadena CA 91109
tel: (818)354-2075
fax: (818)393-6875



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Re: [Flexradio] Fw: 16 versus 24 bit audio

[Jim Lux]

At 02:31 PM 5/19/2006, Peter Martinez wrote:
> >From G3PLX:
>
>That's a picture of the broadband noise EMITTED from the audio OUTPUT side
>of the soundcard in question. My query is about the digital noise registered
>by the input ADC. The tests I have done so far say that there is 8 bits of
>background noise (1 bit = 6dB) coming from the 24-bit ADC in the Firebox
>with the input shorted. That says there's only 16-bits of useful signal in
>there. Why should I pay for 24 bits when 8 of them are below the noise?
>
>I say again: Is there anybody out there writing their own SDR software that
>has actually masked-off the bottom 8 bits or replaced it with random
>numbers?

I've done a fair amount of A/D testing for a variety of RF applications, 
and have a few comments:
1) Have you tried making the noise measurement with a very clean sinewave 
input at various levels? (this is somewhat non trivial to generate)
2) Is the random low order noise white? Or does it have some spectral shaping?
3) It's possible that the limit of performance is set by something other 
than preamp noise before the A/D.  For instance, if the clock jitter is 
only good to 16 bits, then there's not much point having a 24 bit noise 
level in the analog side.
4) It might be that 24 bit cards may not give you 24 bit performance, but 
might give you 18 bit or 20 bit performance, so you're better off than a 16 
bit card giving you 14 bit performance
5) The other thing you get with things like the Firebox is multiple ins and 
outs.

Jim



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Re: [Flexradio] Henry 3K Classic & SDR-1000

[Larry Taft]

Brian,

How is the diode connected in the iso box?  I've made the mistake of 
putting it across the key line when it really has to be across the coil 
of the relay to suppress the back EMF of the coil.  Even the little 12 
volt coils produce a wallop when the circuit is opened.

If you have a 10X high impedance probe and a scope you can see the back 
voltage is a big number without the diode even with small coils.

I'm familiar with RF loose in the shack.  I burned out the input to my 
Tek 2712 spectrum analyzer a few years ago.  Cost me $2200 to get it 
fixed as I need it for my broadcast measuring business.  Has to have 
traceable calibration to be legal for the measurements so I had to pay 
the shot to Tek for the repairs.

73, Larry  K2LT

Brian Sherrod wrote:
> On 5/20/06, Larry Taft <[EMAIL PROTECTED]> wrote:
>> Brian,
>>
>> What is in the isolation box?  I'm always looking for ways to do the
>> control keying and changeover functions.
>
> Nothing more than a tiny 12v relay I was keying by grounding from pin
> 7 & 15 with diode protection.
>
>
>> As far as repair of the ULN2003A goes, I recommend sending the SDR back
>> to Flex as it is all surface mount stuff on the board.  They are
>> skilled, reasonably priced and fast.  Three out of three is good!  Just
>> had my SDR-1000 in for upgrades and repair.
>
> Okay Larry,  thanks for that info!
>
> 73 Brian / w5ami
>
>

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[Flexradio] Dayton

[cbayona]

Things are pretty interesting at Dayton, I met some of the Flex gang. I
was looking for a HP signal generator but the prices are outrageous so I
skipped buying one.

My plans were that shortly after Dayton I would purchase a SDR-1000, but
since I could not purchase the signal generator, and they had SDR-1000's
in stock, I went ahead and purchased one.

This was late yesterday as Dayton was closing so I dind't have a chance
to purchase a small power supply and cables so I can play with it while
on vacation. That will be taken care of shortly when I head for the bone
yard.

The weather yesterday was cloudy in the morning and sunny after 10:00AM,
today is sunny also but it might rain in the afternoon.

See ya in the funny papers.
-- 

Cecil
KD5NWA
www.qrpradio.com


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Re: [Flexradio] Henry 3K Classic & SDR-1000

[Brian Sherrod]

On 5/20/06, Larry Taft <[EMAIL PROTECTED]> wrote:
> Brian,
>
> What is in the isolation box?  I'm always looking for ways to do the
> control keying and changeover functions.

Nothing more than a tiny 12v relay I was keying by grounding from pin
7 & 15 with diode protection.


> As far as repair of the ULN2003A goes, I recommend sending the SDR back
> to Flex as it is all surface mount stuff on the board.  They are
> skilled, reasonably priced and fast.  Three out of three is good!  Just
> had my SDR-1000 in for upgrades and repair.

Okay Larry,  thanks for that info!

73 Brian / w5ami

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Re: [Flexradio] Henry 3K Classic & SDR-1000

[Larry Taft]

Brian,

What is in the isolation box?  I'm always looking for ways to do the 
control keying and changeover functions.

As far as repair of the ULN2003A goes, I recommend sending the SDR back 
to Flex as it is all surface mount stuff on the board.  They are 
skilled, reasonably priced and fast.  Three out of three is good!  Just 
had my SDR-1000 in for upgrades and repair. 

73, Larry  K2LT

Brian Sherrod wrote:
> Thanks for all the replies on this.  Larry, thanks for making the
> measurements on yours.  That confirms that mine has apparently been
> fried, as it measures about 25 ohms keyed or not, and I do have the X2
> sequencing checked in setup.  I suspect RF must have gotten into my
> isolation box, as I was not connected directly to the Henry key jack
> from the SDR. Initially I had a boat load of RF in the shack until I
> found the problem.  By then it was too late, and it quit keying.

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Re: [Flexradio] Henry 3K Classic & SDR-1000

[Brian Sherrod]

Thanks for all the replies on this.  Larry, thanks for making the
measurements on yours.  That confirms that mine has apparently been
fried, as it measures about 25 ohms keyed or not, and I do have the X2
sequencing checked in setup.  I suspect RF must have gotten into my
isolation box, as I was not connected directly to the Henry key jack
from the SDR. Initially I had a boat load of RF in the shack until I
found the problem.  By then it was too late, and it quit keying.

Oh well, in the meantime I will operate with a switch.  Anyone know
what I need to do to make the repair to the ULN2003A circuit?

Thanks again for the help!

Brian

On 5/19/06, Larry Taft <[EMAIL PROTECTED]> wrote:
> Brian,
>
> I just made some measurements on my SDR-1000 with the power OFF.
> X2-pin7, reads infinite resistance using a digital multimeter.  In the
> Diode test position with plus to pin 7 and minus to ground it shows open
> circuit.  With the leads reversed minus to pin 7 the reading is 540
> millivolts.  This is consistent with the ULN2003A circuit which has a
> transistor open collector from  pin 7  and the emitter to ground.  Also
> a diode cathode pin 7 with anode to ground.  You should get the same
> readings measuring pins 1 through 6  to ground as these are all the same
> kind of circuit.
>
> Unfortunately pins 1 through 6 don't have the  external amplifier
> sequencing delay  so tring to use one of these will result in "hot
> switching" the amp relay.  For now I'd use a foot switch on the amp to
> turn it on just before you PTT the SDR.  One man band sort of thing but
> it will work as I have done it here with my Drake L-4 before I got the
> FET isolated keying module from N0XAS.  I put a back diode in my L-4 as
> I use it with several other radios.
>
> I also looked in the manual for my Henry 2K and it shows there is NO
> back diode for the changeover relay.  There aren't any voltage values
> for the relay so I'm guessing its a 24 volt coil that draws 50 mills or
> so when operated.  Measure the open circuit voltage of the Henry at the
> RCA jack and then use the milliamp meter to check the coil current.
> When the relay keing circuit is opened to turn the relay off you can get
> over 400 volt spike on the control line.  This can cause failure of the
> X2-7 transistor.
>
> Haven't run the Henry in years.  Its gained way too much weight just
> sitting in my office.
>
> 73, Larry  K2LT
>
> Brian Sherrod wrote:
> > I just hooked up my SDR to my Henry 3K Classic.  I used pin 7 and pin
> > 15 (grnd) on the X2 connector to cause the relay to go to ground,
> > however something is wrong.  If I use my ohmmeter between 7 and 15
> > without being keyed, I am seeing about 40 ohms, and when I key the
> > SDR, it does not change.
> >
> > I can transmit on the Henry, but the ant relay does not switch back so
> > that I can receive without pressing the standby switch on the amp.
> >
> > The amp only requires grounding the center pin on the rca jack, which
> > is what I thought a connection of pin 7 & 15 would do on keyup of the
> > SDR.
> >
> > Any suggestions appreciated.
> >
> > Brian / w5ami
> >

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Re: [Flexradio] 16 versus 24 bit audio

[Alberto I2PHD]

Peter Martinez wrote:
>  snip 
> Where is all the noise coming from?  Paul's observation that the noise is 
> 100dB down on full-scale could just mean that there is that much noise 
> coming from the SDR1000 RF hardware, but it would be easy to test this by 
> unplugging it and short-circuiting the soundcard input. If the noise stays 
> at the -100dB mark then it says the soundcard is that noisy, which is 
> exactly what I find here with the Firebox. Maybe Paul has already done this. 
> Note that the software measuring the noise needs to be set to the full 
> soundcard bandwidth for this to make sense - it will look a lot cleaner if 
> you select a narrow filter. 
 > - snip -

Yes, this last sentence is very important. When you evaluate the noise level on 
a spectrum display, don't forget that 
you are evaluating it in a bandwidth equal to the FFT bin size. And you can see 
it easily for yourself. If the program 
allows for this, change the bin size, e.g. make it half, and the base level of 
noise will drop by 3 dB...

73  Alberto  I2PHD


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Re: [Flexradio] 16 versus 24 bit audio

[Philip Covington]

On 5/20/06, Philip Covington <[EMAIL PROTECTED]> wrote:
> On 5/20/06, Peter Martinez <[EMAIL PROTECTED]> wrote:
> > from G3PLX:
> >
> > Hello Sami. Thanks for joining the discussion - I feared I was on my own for
> > a while!  I hope some others will contribute too, so I don't get accused of
> > hogging the bandwidth (or perhaps generating too much noise!).
> >
> > You said:
> > >If your low 8 bits are truly random with no input, that's a good
> > >thing! It follows that when you inject a non-random signal, it will
> > >stand out clearly from white noise.
> >
> > Let me be simple-minded and respond:-
> >
> > But if my low 8 bits are truly random and I inject an 8-bit sinewave, isn't
> > it then level with the noise? If I could reduce my 8 bits of added noise to,
> > say, 4 bits, wouldn't my sinewave then be 24dB above the noise?  I don't see
> > how adding that much noise can ever be a good thing to do.  I am not trying
> > to trap you, I really would like to understand what's going on here. This is
> > all new stuff to me.
> >
> > I had a private email from a broadcast engineer who confirmed what I was
> > saying about dither noise. He said they added 0.5 bits-worth of white noise
> > to the analogue signal before digitisation, and subjectively that was the
> > best result.
> >
> > But that's half a bit of dither, not 8 bits.
> >
> > 73
> > Peter
>
> Really, its like we are continuing to beat a dead horse here... a 24
> bit card is not going to give you the theoretical 24 bits of
> resolution, just like a 16 bit card does not necessarily give  you its
> theoretical 16 bits of resolution.  Now some 16 bit cards can come
> close, but most 24 bit cards under $1000 are really going to give you
> << 18 ENOB.   Go back and do a search of the Flex Forum... there are
> examples of so called 24 bit cards that are much worse than some of
> the better 16 bit cards.
>
> When they say 16 bits or 24 bits, the manufacturers are refering to
> the ADCs rated resolution stated by its manufacturer.  The design of
> the ADC front end, board layout, power supply filtering, etc...
> determines whether the board will even come close to the ADC rated
> specs (which are sometimes very optimistic in the first place).
>
> The Delta 44 became the recommended card for the SDR-1000 for a long
> time because in its price range, its performance was good, it has two
> ins and two outs, and ASIO drivers were available for it.
>
> Phil Harmon has just completed testing of the AKC ADC in the JANUS
> ADC/DAC board.  He is seeing great numbers... this is the ADC used in
> one of the < $1000 cards.  We are not going to see 24 bits, but it
> will be much better than the Delta 44 or the Firebox.
>
> 73 de Phil N8VB

P.S.  I was so jaded by all this ADC "bit" stuff that when discussions
were going on in the old Xylo group about which ADC to use, I lobbied
hard for the TI PCM4202 since it was easy to get compared to the AKM
AK5394A part and I was convinced that the performance would not be all
that much better.  Well, I was definitely wrong about that one... this
came out in testing...  A few very smart individuals were suggesting
all along that we look at the AKM part.  I am glad that Phil VK6APH
and Bill KD5TFD continued to test all possibilities...

73 de Phil N8VB

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Re: [Flexradio] 16 versus 24 bit audio

[Philip Covington]

On 5/20/06, Peter Martinez <[EMAIL PROTECTED]> wrote:
> from G3PLX:
>
> Hello Sami. Thanks for joining the discussion - I feared I was on my own for
> a while!  I hope some others will contribute too, so I don't get accused of
> hogging the bandwidth (or perhaps generating too much noise!).
>
> You said:
> >If your low 8 bits are truly random with no input, that's a good
> >thing! It follows that when you inject a non-random signal, it will
> >stand out clearly from white noise.
>
> Let me be simple-minded and respond:-
>
> But if my low 8 bits are truly random and I inject an 8-bit sinewave, isn't
> it then level with the noise? If I could reduce my 8 bits of added noise to,
> say, 4 bits, wouldn't my sinewave then be 24dB above the noise?  I don't see
> how adding that much noise can ever be a good thing to do.  I am not trying
> to trap you, I really would like to understand what's going on here. This is
> all new stuff to me.
>
> I had a private email from a broadcast engineer who confirmed what I was
> saying about dither noise. He said they added 0.5 bits-worth of white noise
> to the analogue signal before digitisation, and subjectively that was the
> best result.
>
> But that's half a bit of dither, not 8 bits.
>
> 73
> Peter

Really, its like we are continuing to beat a dead horse here... a 24
bit card is not going to give you the theoretical 24 bits of
resolution, just like a 16 bit card does not necessarily give  you its
theoretical 16 bits of resolution.  Now some 16 bit cards can come
close, but most 24 bit cards under $1000 are really going to give you
<< 18 ENOB.   Go back and do a search of the Flex Forum... there are
examples of so called 24 bit cards that are much worse than some of
the better 16 bit cards.

When they say 16 bits or 24 bits, the manufacturers are refering to
the ADCs rated resolution stated by its manufacturer.  The design of
the ADC front end, board layout, power supply filtering, etc...
determines whether the board will even come close to the ADC rated
specs (which are sometimes very optimistic in the first place).

The Delta 44 became the recommended card for the SDR-1000 for a long
time because in its price range, its performance was good, it has two
ins and two outs, and ASIO drivers were available for it.

Phil Harmon has just completed testing of the AKC ADC in the JANUS
ADC/DAC board.  He is seeing great numbers... this is the ADC used in
one of the < $1000 cards.  We are not going to see 24 bits, but it
will be much better than the Delta 44 or the Firebox.

73 de Phil N8VB

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Re: [Flexradio] 16 versus 24 bit audio

[Sami Aintila]

On 5/20/06, Peter Martinez <[EMAIL PROTECTED]> wrote:
>
> You said:
> >If your low 8 bits are truly random with no input, that's a good
> >thing! It follows that when you inject a non-random signal, it will
> >stand out clearly from white noise.
>

I guess my wording was a bit provocative. The point I was trying to
make was that even the noise-ridden, seemingly random low 8 bits may
contain some information.

> But if my low 8 bits are truly random and I inject an 8-bit sinewave, isn't
> it then level with the noise?

When looking at the whole soundcard bandwidth, yes it is. But the
noise is spread out evenly over all frequencies, so you can use a
narrow filter to pick up the signal of interest and get only a small
fraction of the noise. If you had completely discarded the low 8 bits
as "useless", you would have lost all (or at least most) of your
signal.

> I had a private email from a broadcast engineer who confirmed what I was
> saying about dither noise. He said they added 0.5 bits-worth of white noise
> to the analogue signal before digitisation, and subjectively that was the
> best result.
>
> But that's half a bit of dither, not 8 bits.
>

I'm not an expert on dithering (or A/D converters in general), but
probably your 8-9 bits of noise contains also something else and not
just dither.

73, Sami

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Re: [Flexradio] 16 versus 24 bit audio

[Peter Martinez]

from G3PLX:

Hello Sami. Thanks for joining the discussion - I feared I was on my own for 
a while!  I hope some others will contribute too, so I don't get accused of 
hogging the bandwidth (or perhaps generating too much noise!).

You said:
>If your low 8 bits are truly random with no input, that's a good
>thing! It follows that when you inject a non-random signal, it will
>stand out clearly from white noise.

Let me be simple-minded and respond:-

But if my low 8 bits are truly random and I inject an 8-bit sinewave, isn't 
it then level with the noise? If I could reduce my 8 bits of added noise to, 
say, 4 bits, wouldn't my sinewave then be 24dB above the noise?  I don't see 
how adding that much noise can ever be a good thing to do.  I am not trying 
to trap you, I really would like to understand what's going on here. This is 
all new stuff to me.

I had a private email from a broadcast engineer who confirmed what I was 
saying about dither noise. He said they added 0.5 bits-worth of white noise 
to the analogue signal before digitisation, and subjectively that was the 
best result.

But that's half a bit of dither, not 8 bits.

73
Peter


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Re: [Flexradio] 16 versus 24 bit audio

[Sami Aintila]

On 5/20/06, Sami Aintila <[EMAIL PROTECTED]> wrote:
> software uses a 11.25 kHz IF.

Ehm... correction: that's 11.025 kHz.

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Re: [Flexradio] 16 versus 24 bit audio

[Sami Aintila]

Hi Peter,

Let me assure you: 24 bits is better than 16. Exactly how much better
obviously depends on lots of things. And 16 bits would indeed be
plenty if there were 16-bit sound cards that could actually deliver
that resolution.

If your low 8 bits are truly random with no input, that's a good
thing! It follows that when you inject a non-random signal, it will
stand out clearly from white noise. If the ADC resolution is good
enough, this will also work at extremely low input levels i.e. when
the S/N ratio is negative.

Everyone, please read the specs of your sound cards. I don't have the
PreSonus, but I have been using the 24-bit Delta 44 for years, already
before there was SDR-1000. The input dynamic range for Delta 44 is
"only" 99 dB(A). Of course, dynamic range does not necessarily
translate to effective number of bits. Nevertheless, if we calculate
ENOB using this value, we get ENOB = (99 - 1.76) / 6.02 = 16.2 bits.

Then, there's also the dB(A) issue. Using A-weighting for audio
applications may (or may not) make some sense, but for SDR-1000 it
generally does not. For example, it looks like my Delta 44 has been
tweaked to give optimal noise performace between 1-10 kHz. Above 10
kHz it's several dB's worse. Which is a problem because the PowerSDR
software uses a 11.25 kHz IF.

73, Sami OH2BFO

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Re: [Flexradio] 16 versus 24 bit audio

[Paul Fletcher]

Sorry Peter - should have said - these figures are with SDR hardware turned
off and unplugged/inputs shorted out.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Martinez
Sent: 20 May 2006 11:51
To: Flex Reflector
Subject: Re: [Flexradio] 16 versus 24 bit audio

>From G3PLX:

Thanks Paul.  Thank goodness I am not alone!

I just did a simple calculation. The ultimate rock-bottom noise power is 
kTB, where k is Boltzmann's constant (1.38e-23), T is room temperature (300 
degrees K) and B is the bandwidth in which you measure it - let's say 24kHz 
for a 48kHz soundcard, so this works out to 1e-16 watts. Let's say the 
circuit is 600 ohms (just so we can get some idea of the picture), so the 
r.m.s. noise level across a 600 ohm source connected to a soundcard is going

to be the square root of 600e-16, or 0.244uV.

Again, just for the sake of it, let's say the soundcard ADC is designed for 
2.2 volts peak-to-peak (that's 0dBm in 600 ohms). A 24-bit ADC would then 
give 2.2/(2^32) volts quantisation level. That comes out to 0.13uV.  That's 
not far off the circuit noise level. That 'feels' about right to me.  It 
says that an 'ideal' 24-bit soundcard should just be able to 'hear' it's own

input noise. This is how we have always designed the front-ends of things!

I am just throwing numbers in there, and the experts may say that the 
circuit isn't 600 ohm, or I haven't taken into account the noise factor of 
the pre-amp, or the typical ADC full-scale isn't 2.2 volts.  But I am surely

not wrong by a factor of 256?

Where is all the noise coming from?  Paul's observation that the noise is 
100dB down on full-scale could just mean that there is that much noise 
coming from the SDR1000 RF hardware, but it would be easy to test this by 
unplugging it and short-circuiting the soundcard input. If the noise stays 
at the -100dB mark then it says the soundcard is that noisy, which is 
exactly what I find here with the Firebox. Maybe Paul has already done this.

Note that the software measuring the noise needs to be set to the full 
soundcard bandwidth for this to make sense - it will look a lot cleaner if 
you select a narrow filter. I am not familiar with the SDR software so I 
don't know how this works.

73
Peter


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Re: [Flexradio] 16 versus 24 bit audio

[Peter Martinez]

>From G3PLX:

Thanks Paul.  Thank goodness I am not alone!

I just did a simple calculation. The ultimate rock-bottom noise power is 
kTB, where k is Boltzmann's constant (1.38e-23), T is room temperature (300 
degrees K) and B is the bandwidth in which you measure it - let's say 24kHz 
for a 48kHz soundcard, so this works out to 1e-16 watts. Let's say the 
circuit is 600 ohms (just so we can get some idea of the picture), so the 
r.m.s. noise level across a 600 ohm source connected to a soundcard is going 
to be the square root of 600e-16, or 0.244uV.

Again, just for the sake of it, let's say the soundcard ADC is designed for 
2.2 volts peak-to-peak (that's 0dBm in 600 ohms). A 24-bit ADC would then 
give 2.2/(2^32) volts quantisation level. That comes out to 0.13uV.  That's 
not far off the circuit noise level. That 'feels' about right to me.  It 
says that an 'ideal' 24-bit soundcard should just be able to 'hear' it's own 
input noise. This is how we have always designed the front-ends of things!

I am just throwing numbers in there, and the experts may say that the 
circuit isn't 600 ohm, or I haven't taken into account the noise factor of 
the pre-amp, or the typical ADC full-scale isn't 2.2 volts.  But I am surely 
not wrong by a factor of 256?

Where is all the noise coming from?  Paul's observation that the noise is 
100dB down on full-scale could just mean that there is that much noise 
coming from the SDR1000 RF hardware, but it would be easy to test this by 
unplugging it and short-circuiting the soundcard input. If the noise stays 
at the -100dB mark then it says the soundcard is that noisy, which is 
exactly what I find here with the Firebox. Maybe Paul has already done this. 
Note that the software measuring the noise needs to be set to the full 
soundcard bandwidth for this to make sense - it will look a lot cleaner if 
you select a narrow filter. I am not familiar with the SDR software so I 
don't know how this works.

73
Peter


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Re: [Flexradio] 16 versus 24 bit audio

[Paul Fletcher]

Peter,

I asked the same question in a different way a while back. I noticed that
with no inputs to the Firebox (or Delta 44 - I have both) the analogue input
meters on the SDR software were reading in the order of 100dB down on
maximum. Doing some simple maths the theoretical minimum fro a 24bit sample
is 145dB down so where is the other 45dB of noise coming from?

I may be over simplifying things but something doesn't seem right? As for
dithering I read an article recently (maybe from Analog devices - can't
remember) and do remember them talking about 1bit dithering.

What I was trying to ascertain is whether there is a hardware issue or if
it's a software issue (conscious design decision or otherwise).

Apologies that this hasn't helped answer anything, but at least one other on
the group is asking the same question!

Regards,
Paul M1PAF

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Martinez
Sent: 20 May 2006 10:07
To: Flex Reflector
Subject: Re: [Flexradio] 16 versus 24 bit audio

>From G3PLX:

Chris:  I wondered about dithering. If I understand the idea, they introduce

some deliberate noise in the ANALOGUE part of the ADC in order to spread the

spectrum of any quantisation noise. Without it, I can imagine a tiny 
sinewave input of, say, one bit amplitude, looking to the software like a 
one-bit squarewave, with 3rd, 5th, ...  harmonics. Dithering the analogue 
input would add a little noise to the fundamental sinewave but push the 
harmonics down.

But, as you say, surely they would only introduce about one bit's worth of 
dither noise, not eight?  6dB not 48dB?

I just did some proper measurements on the Firebox. On runs of 1200 samples 
at 48kHz, the r.m.s. noise level on the line input is about 400. It's about 
450 on the Mike input.  I did this by squaring each sample, summing these, 
dividing by 1200, and taking the square root. At the same time I calculated 
the DC level, just to make sure that was a lot smaller and didn't confuse 
things.  This is about 9 bits of noise, or more if you suppose the peaks are

higher.

On the built-in 16-bit card in this laptop the DC level is 15 and I am not 
sure how to allow for this in the r.m.s calculation, but only the lsb 
toggles randomly. The MP3+ is the same, unless I select the mike input when 
the rms noise level is about 10.  These figures are what I would have 
expected intuitively. One would expect the analogue noise-level to be 
roughly the same as the quantisation noise, in the same way that one would 
optimise any receiver so that all sources of noise contribute more-or-less 
the same in order to maximise the dynamic range.

Intuitively I would have expected a 24-bit card to do the same, but maybe I 
am being naive. These measurements, and the figure quoted by Alberto, tells 
me that the "24-bit" figure in the spec. is to some extent a marketting 
ploy. I can imagine a scenario where the designer of a new soundcard 
measured that it only gave 18 bits of useful data, so he decides to mask the

data to 18-bits and sell it as an 18-bit card.  The marketting manager of 
the company would surely override him, especially if the competition were 
all actively marketting so-called 24bit cards!

73
Peter


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Re: [Flexradio] 16 versus 24 bit audio

[Peter Martinez]

>From G3PLX:

Chris:  I wondered about dithering. If I understand the idea, they introduce 
some deliberate noise in the ANALOGUE part of the ADC in order to spread the 
spectrum of any quantisation noise. Without it, I can imagine a tiny 
sinewave input of, say, one bit amplitude, looking to the software like a 
one-bit squarewave, with 3rd, 5th, ...  harmonics. Dithering the analogue 
input would add a little noise to the fundamental sinewave but push the 
harmonics down.

But, as you say, surely they would only introduce about one bit's worth of 
dither noise, not eight?  6dB not 48dB?

I just did some proper measurements on the Firebox. On runs of 1200 samples 
at 48kHz, the r.m.s. noise level on the line input is about 400. It's about 
450 on the Mike input.  I did this by squaring each sample, summing these, 
dividing by 1200, and taking the square root. At the same time I calculated 
the DC level, just to make sure that was a lot smaller and didn't confuse 
things.  This is about 9 bits of noise, or more if you suppose the peaks are 
higher.

On the built-in 16-bit card in this laptop the DC level is 15 and I am not 
sure how to allow for this in the r.m.s calculation, but only the lsb 
toggles randomly. The MP3+ is the same, unless I select the mike input when 
the rms noise level is about 10.  These figures are what I would have 
expected intuitively. One would expect the analogue noise-level to be 
roughly the same as the quantisation noise, in the same way that one would 
optimise any receiver so that all sources of noise contribute more-or-less 
the same in order to maximise the dynamic range.

Intuitively I would have expected a 24-bit card to do the same, but maybe I 
am being naive. These measurements, and the figure quoted by Alberto, tells 
me that the "24-bit" figure in the spec. is to some extent a marketting 
ploy. I can imagine a scenario where the designer of a new soundcard 
measured that it only gave 18 bits of useful data, so he decides to mask the 
data to 18-bits and sell it as an 18-bit card.  The marketting manager of 
the company would surely override him, especially if the competition were 
all actively marketting so-called 24bit cards!

73
Peter


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Re: [Flexradio] 16 versus 24 bit audio

[Christopher T. Day]

Peter,

One subtlety that was mentioned once but not emphasized very much is
that some of what you may be seeing as "noise" is actually deliberate
dithering in the card, which has the effect of pushing quantization
noise out to inaudible frequencies. 8-bits seems a bit much for
dithering to me, but still, a "perfect" non-dithering card, i.e., one
that outputs a pure string of '0's for a shorted input, is arguably
_worse_ than a dithering card that outputs a controlled spectrum of
noise on shorted input. It can all get rather weird when there are
non-linear, low S/S+N psychoacoustic detectors at the end of the chain,
i.e., ears.


Chris - AE6VK


-Original Message-
From: Peter Martinez [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 19, 2006 11:20 PM
To: Flex Reflector
Subject: Re: [Flexradio] 16 versus 24 bit audio

>From G3PLX:

Alberto:  Thanks!  That's the information I was seeking.  It looks as if
I 
was half-right, if there are, at best, 18 good bits in a 24-bit card. As

Phil says, if 16-bit cards turn out to be worse that 16-bits, then there
is 
some reason to go for a 24-bit card (which is worse than 24bits!).  But
for 
sure, 24 bit cards are not going to be 256-times (48dB, 8-bits) better
than 
16-bit cards.

I will now continue working to convert my own 16-bit SDR code to
24-bits, 
but I think I will add a button on the front panel which zaps the bottom
8 
bits. It will be VERY interesting to see if it makes any difference. To
take 
Frank's point, as far as I can see from both the ASIO and WMME
interfaces, 
there IS an accessible point in the software where this can be done
before 
the audio data becomes floating-point.

Thanks to all who replied.

73
Peter 


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