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Better sound from an SDR (such as a Flex) or other TX that uses a balanced modu




 
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Author Topic: Better sound from an SDR (such as a Flex) or other TX that uses a balanced modu  (Read 4037 times)
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WB2CAU
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« Reply #50 on: March 05, 2017, 01:16:26 PM »


 Maybe a board if there is enough interest.


Add me to the list of ops with interest in purchasing a blank PCB.  

I had boards made for an old PE project from this guy last year.  I was very impressed with the quality.  His prices are very reasonable.  All he needs is a good clear photo print of the board layout.  

http://www.farcircuits.net/

Although I've made numerous projects using perfboard, it's a PIA.  An etched board makes a much nicer, neater project, and greatly reduces the possibility of a wiring error.   The possibility of a wiring error increases with complexity.  This circuit is complex enough where a wiring error is possible with perfboard.  That's why I'd gladly pay for a printed circuit board.
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W2PFY
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« Reply #51 on: March 05, 2017, 05:13:28 PM »

.
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WA2SQQ
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« Reply #52 on: March 06, 2017, 09:14:34 AM »

... would it be the last thing in the line after the compressor limiter or before it?

Great question ... My audio chain is balanced throughout, and I see that this is only "line" out. Not sure that it makes that much difference feeding the radio with unbalanced line out. Also have one more question - why two phase reversal switches, @ input and output?

Already started building mine.
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KD6VXI
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« Reply #53 on: March 06, 2017, 09:23:24 AM »

1.  Add a phase splitter with a tl082 or 72 op amp on the output.

2.  Phase switch on in and out because when your audio signal goes through the Low Pass Filter you can have phase reversal.

--Shane
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WA2SQQ
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« Reply #54 on: March 06, 2017, 10:01:52 AM »

1.  Add a phase splitter with a tl082 or 72 op amp on the output.
--Shane
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I'll have to research that one - thanks.
Since I had several 074's, I decided to use those throughout. The unused sections could probably be used for the above. I don't think there would be any audible difference using "line" vs a balanced output. Thanks
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Steve - K4HX
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« Reply #55 on: March 06, 2017, 11:47:57 AM »

You want the polarity (phase is an incorrect term) on the input so that the limiter is working on the side of the signal with the smallest amplitude. The switch on the output is required to ensure the side of the signal with the largest amplitude is modulating the transmitter in the positive direction. Without these two switches, much of the effectiveness of the processor could be lost.
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WA2SQQ
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« Reply #56 on: March 06, 2017, 11:56:37 AM »

Thanks for the education ...
Very much appreciated
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IN3IEX
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« Reply #57 on: March 07, 2017, 04:48:55 AM »

DC coupling to the modulator, I suppose...
There is a need to check this point.

G.

I picked up a Flex 5000 over the weekend at the Marlboro flea.  The price was right, so it followed me home....

Anyway, I put the thing on the air, hooked to my normal audio system that I use with the class E rigs.  The Flex has incredible frequency response, and is quite clean.  However, there was one problem - no negative peak limiter.  The result of this is the "folding back" AKA wavelets problem when one exceeds 100% modulation.  Such rigs don't really overmodulate - they fold back.   The resultant distortion is particularly annoying in standard AM detectors (fine in a sync detector)....

The flex has a particularly good low frequency response (down to single digits) so a negative peak limiter is very practical to use here, and there won't be any waveform "tilt".  So, I built a negative peak limiter followed by a 6 pole filter with the 3dB corner at 5.7kHz, and it works just great !!!



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steve_qix
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« Reply #58 on: March 12, 2017, 08:18:30 PM »

You want the polarity (phase is an incorrect term) on the input so that the limiter is working on the side of the signal with the smallest amplitude. The switch on the output is required to ensure the side of the signal with the largest amplitude is modulating the transmitter in the positive direction. Without these two switches, much of the effectiveness of the processor could be lost.

Steve is correct.  The NPL shown works properly for the Flex and with class E rigs because they are set up such that positive going energy generates a positive peak.  This may not be the case with all transmitters, in which case a phase selector on the output would be required.  An easy solution would be to use a TL074 instead of a TL072 in the input, and use one of the other op amps in the TL074 as a unity gain inverting amplifier.   Then add a switch to select between the inverted and non-inverted output.
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« Reply #59 on: March 13, 2017, 09:20:26 PM »

Sorry guys, I don't know what I was thinking.  The negative peak limiter ALREADY has an output phase switch included (I guess I anticipated this need, and put it into the design).  So ignore everything I said circuit-wise in the previous post  Roll Eyes

The circuit is all-ok as is.

Regards,  Steve
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« Reply #60 on: June 18, 2017, 09:53:50 PM »


The latest release of PowerSdr for the Flex Legacy series now has the ability to flip the audio phase. The latest from Darrin Ke9ns is V2.8.40, and the phase flip feature is described below:

06/16/17 v2.8.0.40 Add Phase invert function (setup->Transmit->Phase invert) for your mic audio. To see the difference set the Display mode to Panascope, and toggle the Phase Invert while you transmit

This is good news!

Jim
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N1BCG
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« Reply #61 on: June 19, 2017, 04:38:36 AM »

This circuit could be made even more useful with an adjustable positive peak limiter for those who want to protect mod transformers and other components from excessive voltages.

Same perceived loudness and cleaner sound on most receivers.
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NA3CW
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« Reply #62 on: July 21, 2017, 09:43:52 PM »

Hi, Steve.  I don't remember if we've ever QSOd but I enjoy the PM-AM net on Sunday evenings on 3837. 

Thanks much for posting your limiter circuit.  I have a much modified FT-920.  I drive the balanced modulator directly from my own input xfmr-coupled op amp buffer and an external audio rack. 
A question was raised a few posts above that I was wondering as well: Your circuit is all DC coupled and in your full modulator you run all DC coupled.  Does your limiter need to be DC coupled to the balanced modulator to prevent flux balancing if input transformers and/or capacitors are used in the audio chain?  I can do that but I'd need to remove my present input circuit and rework it as DC coupled to the balanced modulator.  If DC coupling is not required, I can (happily) leave it alone.
Tnx es 73,
Chuck
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« Reply #63 on: July 24, 2017, 02:21:29 PM »

With any asymmetrical limiting, DC coupling is definitely better.

There is a small DC shift that occurs with this type of circuit.  Any AC coupled elements will tend to try to "integrate" this DC shift over time.  If things like coupling capacitors, etc. are large relative to their terminating resistance, the time constant will be long and unless someone put a constant (negative peak clipped) signal into the system, in which case the negative peak clipped waveform will shift lower, all will work quite well.

The other thing that occurs with AC coupling if the audio rolloff is in the usable audio range is waveform "tilt".  This comes about because any low frequency roll-off will tend to act like a differentiator, and the leading edge of the clipped waveform will have a higher amplitude than the trailing edge.

That is why with plate modulated or other transformer coupled modulators, applying negative peak limiting after the modulation transformer/capacitor choke combination (if so wired) works much better than trying to put an asymmetrically clipped signal through the iron.

So, the bottom line is this:  If your transmitter has VERY good low frequency response (flat to lower than 10 or 15 Hz), waveform tilt and shift shouldn't be a problem with normal, varying audio waveforms.

Hope this helps  Smiley
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« Reply #64 on: July 24, 2017, 02:48:48 PM »

With any asymmetrical limiting, DC coupling is definitely better.

[...]
So, the bottom line is this:  If your transmitter has VERY good low frequency response (flat to lower than 10 or 15 Hz), waveform tilt and shift shouldn't be a problem with normal, varying audio waveforms.

Hope this helps  :)


All very true.

Broadcast audio processing pioneer Robert Orban (wonder if he's related to Viktor Orban) once wrote an article in which he suggested that, in order to accurately reproduce an audio processor's tightly-peak-limited waveform, an AM transmitter should have flat response down to 0.7 Hz -- and no, that's not a typo.

73,

Kevin, WB4AIO.
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NA3CW
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« Reply #65 on: July 24, 2017, 03:48:12 PM »

Yes, Steve, it does help. Thank you for the quick reply.  Your response wraps much better wording around what I imagined to be the situation.  The balanced modulator in the FT-920 is a run-of-the-mill IC balanced mixer.  The carrier level is determined by a DC imbalancing current.  At present the audio is AC coupled and rolls off at about 100 Hz so it's not an ideal signal path for asymmetrical waveforms.   

If I brought out a buffered DC-coupled signal pin to the outside world, I could create a full audio chain with frequency contouring, asymmetrical clipping and filtering, carrier control, etc, all in one package and have full control over the results.  I might even throw in a phase rotator.  :-)  Given that the FT-920 is not particularly amenable to experimentation, with all its surface mount components on the backside of the motherboard and a zillion hardwired connections around the edges, I may hold this project off until retirement.  My Yaesu is already quite non-standard with relay controlled filter bypassing, selectable audio sources, modified ALC, etc.   In the meantime I have some interesting design ideas to work out. 

Thanks again and 73,
Chuck
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KD6VXI
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« Reply #66 on: July 24, 2017, 04:40:43 PM »

I ran into problems when I tried to AC couple a NPL into my otherwise DC coupled PWM.

My carrier would shift ALL. over the place!  Very unnerving!  Looked like controlled carrier gone wild!

Removed the DC carrier level control on the pwm and the associated capacitor on the audio  input and let my NPL and assymetry board (mine stretches positive peaks, the npl is a baseline prohibiter in my case) inject a DC level instead.

All is now happy in assymetry land.  I can stretch to 420 pct pos peaks and don't suffer from the carrier following the low frequencies.

Another way of explaining what Steve said!  ☺️

--Shane
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« Reply #67 on: July 24, 2017, 07:10:01 PM »

Two thoughts...

A board that fits into the D-104 base might be neat.

I know, or I think I know, that QIX doesn't mind using quad opamp packages.
But, if anyone is going to go and make a run of PCBs, my opinion is that duals
are far preferable since the quad DIP package is totally obsolete.

There are a whole lot of duals that are very nice for audio applications.
Even surface mount devices that can be put onto DIP adapters, if one wanted to
do so.

Phase flip might be implemented with a simple "see-saw" phase splitter, and a switch
to pick which side of the phase splitter you want as the output? That would be a single
transistor or JFET... cap couple of course.
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« Reply #68 on: July 24, 2017, 07:17:03 PM »

In a past life I was Chief Engineer of KTWR, Guam.  While they have a different set of transmitters now, at that time we had four Harris SW100A's and one HCJB HC-100, all 100kw HF AM broadcast transmitters.  The SW100A's had 80kHz PDM tube modulators.   I designed and installed Dynamic Carrier Control boards for them.  Saved about $5000 per month in power bills.  When there was no audio they'd settle back to 50kw.  As audio arrived, they'd perk back up to 100kw to accommodate the modulation.  We didn't save money when music was playing but we did well when there was a speaker who had long, thoughtful pauses.  Wink

Since I had DC control of the modulators I could mix in carrier level, linearizing feedback, processed audio, and control scaling, into the same circuit.  As Shane related, that's about the only real way to pull off this asymmetrical processing, I think.  It should be straightforward to do a simpler version of this for my FT-920 balanced modulator.

Thanks to all for the great comments!

Chuck
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« Reply #69 on: July 25, 2017, 09:31:34 AM »

Just about done building it, so now I have a question. Going to be feeding a Flex 6500. My PR-40 feeds a Behringer DEQ2496 which currently feeds the 6500 via a balanced audio connection. Not really doing much processing with the 2496, but it was the only thing I had that could serve as  Mic preamp. Since the limiter does not use balanced audio,  I'll use the "LINE" input on the 6500. I don't expect any difference in quality.

Am I correct in thinking that the limiter needs to be the last device before audio hits the 6500?
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« Reply #70 on: July 26, 2017, 12:48:01 PM »

Just about done building it, so now I have a question. Going to be feeding a Flex 6500. My PR-40 feeds a Behringer DEQ2496 which currently feeds the 6500 via a balanced audio connection. Not really doing much processing with the 2496, but it was the only thing I had that could serve as  Mic preamp. Since the limiter does not use balanced audio,  I'll use the "LINE" input on the 6500. I don't expect any difference in quality.

Am I correct in thinking that the limiter needs to be the last device before audio hits the 6500?

Here is my view.

An equalizer should only be used if there is some deficiency in the audio chain, or you're trying to generate an NRSC-type pre-emphasis curve.

Assuming you have plenty of mic gain, the limiter is usually all that is need for amateur radio AM and is placed just before the transmitter audio input.

I was not aware the Behringer DEQ2496 had a mic preamp, but by golly it does:

https://www.soundpro.com/catalog/documents/DEQ2496_brochure.pdf


Phil - AC0OB

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« Reply #71 on: July 26, 2017, 02:58:14 PM »

So in my application the 2496 is a huge overkill, but it serves as a very nice preamp. I usually use the PR-40 which really does not need too much EQing. I have a few other mics that I use from time to time, which can benefit from some "enhancing", so the 2496 does that.

Once I get the limiter done I need to order a mod-monitor, something I've been meaning to do for some time. For me the recent Flex software update was free, so the $200 I saved can be applied towards the mod monitor. Thanks for the input.
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« Reply #72 on: July 26, 2017, 03:47:50 PM »

In my case, I ended up using a Behringer multiband compressor, DSP9024, with its peak limiter disabled, feeding into the line input of my (older SDR-1000) Flex. I then use the leveler/peak limiter function of the Flex to do the final peak control. It works well, better than external peak limiters I have tried.

I have the leveler adjusted so it acts slowly (2ms attack, 5000 ms recovery, 500 ms hang) and never changes the gain more than 4 dB -- this just assures that I am in the "sweet spot" for the final limiter. I am not familiar with SmartSDR, but with PowerSDR 2.5.3, I have choice of "CPDR" (really a clean peak limiter misnamed as a compander) and "DX" (a more aggressive limiter that allows some clipping). Both work well at low settings. With DX I never exceed a setting of zero, and it still is as aggressive as a old-time Top 40 broadcast AM peak limiter. A CPDR setting of 2 is plenty.

In my opinion, the internal limiters work better than the Behringer's limiter function or the Aphex Dominator I was formerly using.

73,

Kevin.
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« Reply #73 on: July 26, 2017, 08:11:52 PM »

Yes, the negative peak limiter (and its filter) should be the very last thing in the audio chain.

Look forward to hearing your report (and maybe you directly!).

Regards,

Steve
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