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Friday, December 2, 2022

But why? Why Can't I Listen to DSB (or AM) on my Direct Conversion Receiver?

I've said this before:  I just seems so unfair.  We just should be able to listen to DSB signals with our beautifully simple homebrew Direct Conversion receivers. I mean, building a DSB transmitter is a natural follow-on to DC receiver construction.  And we are using AM shortwave broadcast stations (Radio Marti --I'm looking at you)  to test our DC receivers for AM breakthrough.  But when we tune these stations in, they sound, well, awful.  So unfair!  Why?   Unfortunately it has to do with laws.  Laws of physics and mathematics.  Blame Fourier, not me.  

Over the years there has been a lot of handwaving about this problem.  From Doug DeMaw, for example: 

In his "W1FB's Design Notebook," Doug wrote (p 171):  "It is important to be aware that two DSSC (DSB) transmitters and two DC receivers in a single communication channel are unsatisfactory.  Either one is suitable, however, when used with a station that is equipped for SSB transmissions or reception. The lack of compatibility between two DSSC (DSB) transmitters and two DC receivers results from the transmitter producing both USB and LSB energy while the DC receiver responds to or copies both sidebands at the same time."

That's correct, but for me, that explanation didn't really explain the situation.  I mean we listen to AM signals all the time.  They produce two sidebands, and our receivers respond to both sidebands, and the results are entirely satisfactory, right?  Why can't we do this with our Direct Conversion receivers?  I struggled with this question before: https://soldersmoke.blogspot.com/2015/07/peter-parker-reviews-dsb-kit-and.html  You can see in that post that I was not quite sure I had the answer completely correct. 

It took some discussion with a fellow Vienna Wireless Society member, and some Googling and Noodling for me to figure it out.  But I think I've got it: 

Imagine a station transmitting a DSB signal at 7100 kHz with a 1 kHz tone at the AF input.  There will be signals at 7101 kHz and at 7099 kHz.  Assume the carrier is completely suppressed. 

We come along with our DC RX and try to tune in the signal. 

Remember that they heart of the DC RX is a product detector, a mixer with the VFO (or PTO) running as close as we can get it to the suppressed carrier frequency (which we can't hear).  

Lets assume that we can somehow get our VFO or PTO exactly on 7100 kHz.  The incoming signals will mix with the VFO/PTO signal.  We are looking for audio, so we will focus on the difference results and ignore the sum results of the mixing.  

The difference between 7101 and 7000 is 1 kHz.  Great! And the difference between 7099 and 7000 is 1 kHz also.  Great again, right?  We are getting the desired 1 kHz signal out of our product detector, right?  So what's the problem?  

Here it is: SIDEBAND INVERSION.  Factoring in this part of the problem helps us see the cause of the distortion that plagues DSB-DC communication more clearly. 

Remember the Hallas Rule:   Whenever you subtract the modulated signal FROM the unmodulated signal, the sidebands invert.  So, in this case, we are subtracting that 7099 "lower sideband" signal FROM the 7100 VFO/PTO signal.  So it will invert.  It will become an upper sideband signal at 1 kHz.  We will have two identical 1 kHz signals at the output.  Perfect right?   Not so fast. Not so PERFECT really.  

The perfect outcome described above assumes that our VFO/PTO signal is EXACTLY on 7100 kHz.  And exactly in phase with the suppressed carrier of the transmitter.  But if it is even SLIGHTLY off, you will end up with two different output frequencies, signals that will move in and out of alignment, causing a wobbling kind of rapid fade-in, fade-out distortion.  You can HEAR this happening in this video by Peter Parker VK3YE, starting at 6:28:

And you can see it in this LTSpice simulation.  

This LTSpice model just shows two diode ring mixers.  The transmitter is on the top, the receiver is on the bottom. The transmitter has RF at 7100 kHz at L1 and audio at 1 kHz at R1.   The receiver has the VFO at 7100.001 L7,  DSB from the transmitter at L12 with audio appearing at R4.  It is instructive to watch the output as you move the VFO frequency.  If you move the VFO freq away from the transmit carrier osc frequency you will see the distortion.  Here is the netlist for the LTSpice simulation: 

On paper, using simple mixer arithmetic, you can tell that it will be there. With the VFO/PTO just 1 Hz (that's ONE cycle per second) off, you will end up with outputs at 1.001 kHz and at .999 kHz.  Yuck.  That won't sound good. These two different frequencies will be moving in and out of alignment -- you will hear them kind of thumping against each other.    And that is with a mere deviation of 1 Hz in the VFO/PTO frequency!  We are scornful when the SDR guys claim to be able to detect us being "40 Hz off."  And before you start wondering if it would be possible to get EXACTLY on frequency and in phase, take a look at the frequency readout on my PTO.  

Now consider what would happen if the incoming signal were SSB, lets say just a tone at 7101 kHz.  We'd put our VFO at around 7100 kHz and we'd hear the signal just fine.  If we were off a bit we'd hear it a bit higher or lower in tone but there would be no second audio frequency coming in to cause distortion.  You can hear this in the VK3YE video:  When Peter switches to SINGLE Sideband receiver, the DSB signals sound fine. Because he is receiving only one of the sidebands. 

The same thing happens when we try to tune in an AM station using a Direct Conversion receiver:  Radio Marti sounds awful on my DC RX, but SSB stations sound great. 

My Drake 2-B allows another opportunity to explore the problem.  I can set the bandwidth at 3.6 kHz on the 2-B, and set the passband so that I will be getting BOTH the upper and the lower sidebands of an AM signal. With the Product Detector and the BFO on,  even with the carrier at zero beat  AM sounds terrible.  It sounds distorted.  But -- with the Product Detector and BFO still on --  if I set the 2-B's  passband to only allow ONE of the sidebands through,  I can zero beat the carrier by ear, and the audio sounds fine. 

There are solutions to this problem:  If you REALLY want to listen to DSB with a DC receiver, build yourself a synchronous detector that gets the your receivers VFO EXACTLY on frequency and in phase with the transmitter's oscillator.  But the synchronizing circuitry will be far more complex than the rest of the DC receiver. 

For AM, you could just use a different kind of detector.  That will be the subject of an upcoming blog post. 

Please let me know if you think I've gotten any of this wrong.  I'm not an expert -- I'm just a ham trying to understand the circuitry. 


  1. Bill, I too pondered this some time ago (my first "rig" was DC & DSB) and I came to the same conclusion as you ... I agree, all should work if the VFO is exactly in the middle of the sidebands. But wait, there's more. I think you also need a perfect mixer.

  2. Nev: I'm glad to hear that I am not the only one thinking about this. I'll have to give mixer perfection some more thought! 73 Bill

  3. A synchronous detector would have a rather hard time with DSB stations because of the missing/suppressed carrier. The carrier would have to be extracted somehow from the sidebands.

    You can listen to another DSB station or have a perfectly fine 2-way DSB-QSO by inverting the audio at RX (or TX) and moving the VFO up or down 3kHz for a standard 300...2700Hz voice channel.

  4. Yes, synchronizing the VFO frequency and(!) phase to the carrier it will make it work. But, you probably ask, how can you sync to a carrier if it is suppressed, not transmitted? Luckily there is a solution: The Costas loop.

  5. DSBsc was never a ham thing until after SSB arrived.

    GE was interested, hoped to move in with it instead of Collins and SSB. There are advantages, but on ly if you have a synchronous detector. So GE had one in the late forties. And in 1957 or 58, Webb described one in CQ. He worked for GE. But it was at least as complicated as an S38. Webb also had a couple of articles on DSB in general that year.

    It changed in 1970 when ICs made it feasible. But a lot was about AM. Signetics had a line of analog PLLs, which made AM easy, but needed the carrier. There were some schemes in Ham Radio magazine that did use the two sidebands to place the BFO right between.

    But it wss Sony's 2010 in the esrly eighties that made sybchronous detectors well known. I have a suspicion it wasn't just about synchronous detection, but it's the same hardware needed for a sideband slicer, like in the fifties. Get decent SSB reception without a narrower filter.

    So DSBsc was mostly a "cheap way to get on SSB". Lots of mods of AM rig outputs. It allrelied on an SSB receiver at the other end, which converted DSB to SSB. They'd never know. And thus no reception problems, except for AM era receivers without the good selectivity.

    If you're going to invert audio, you need to look at a better transmitter. Simple phasing works because you can filter the audio before the phasig network.

    Or SDR.

    The problem with ham radio is it focuses too.much on simplicity.Doug DeMaw did it endlessly, trying to lure people in. But too often people stay with simple. Got to have that 3 transistor rig, no matter the limitations.

  6. Bill - just wondering if you derived an I and Q from audio output of the DC Receiver, and then sum/subtract them, would this allow you to get rid of one of the "sidebands" ? Can be done in DSP or analog with op-amps - thoughts?



  7. Dave: Yes anything that would get rid of one of the sidebands would allow you to tune in the other one without the distortion I describe. That's why I had so much success with my DSB transmitter -- all the other stations were listening with SSB receivers (transceivers really). You could do what you describe with a filter SSB receiver or with the phasing type receiver you describe. I built one of these phasing receivers back in 2015: https://soldersmoke.blogspot.com/2015/12/new-rig-frankenstein-phasing-receiver.html The problem is that these receivers are a LOT more complicated than an ordinary Direct Conversion receiver. They are even more complicated than a filter type superhet. Bill

  8. Hi Bill - Thanks! So if I understand correctly, not only would you need to send the recovered audio through a constant 90 degree phase shift network to obtain a "Q" version, but you would also have to have two mixers, one fed from the LO and another fed from a 90 deg shifted version of the LO? In that case, I can see how that would add enough complexity to throw the project into another bracket entirely.

  9. There were direct conversion phasing receivers in the late sixties. One is even in the last ARRL SSB manual, using a B&W audio phasing network.

    About 1974 there was a better one in Ham Radio. Using a divider for the RF phasing. The author a year or so.later had a matching transmitter.p

    Gary Breed had a fancy phasing receiver in the late eighties in QST. He even terminated the mixers.

    Then Rick Campbell had a string of high performance phasing rigs in QST , in the nineties or maybe early 2000s.

    An SDR is just a phasing rig with the audio phasing done digitally.


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