• Get involved.
    We want your input!
    Apply for Membership and join the conversations about everything related to broadcasting.

    After we receive your registration, a moderator will review it. After your registration is approved, you will be permitted to post.
    If you use a disposable or false email address, your registration will be rejected.

    After your membership is approved, please take a minute to tell us a little bit about yourself.
    https://www.radiodiscussions.com/forums/introduce-yourself.1088/

    Thanks in advance and have fun!
    RadioDiscussions Administrators

new 8600 software offering SSB stereo

rorban said:
The white paper I wrote contains a calculation proving that a certain input signal (two sinewaves, identical expect for a 90 degree phase shift, applied to the L and R channels) when peak-limited using techniques appropriate for DSB modulation, causes peak modulation of about 138% when using compatible SSB. The calculation agrees with the measurement shown in the white paper. I therefore concluded that the SSB modulator needs more peak limiting than the DSB case to accommodate cases like this. Moreover, I observed experimentally that real-world program material when peak-limited correctly for DSB operation also causes very substantial overmodulation in the SSB case. Our SSB modulator uses filters with no group delay distortion, so it is essentially "ideal." Because my conclusion is backed up by theory and measurements that verify the theory, I would call it way more than an "opinion."

It may turn out to be possible to modify the system and/or improve the extra audio processing so that the DSB and SSB transmissions sound essentially equal on-air. This is still something we are investigating. One possibility is to use DSB from 0 to around 2 kHz (a frequency band where a significant amount of program energy resides) and SSB above 2 kHz. This would extend the baseband spectrum to 40 kHz instead of 38 kHz, as pure SSB would. However, it would ease audio processing requirements for SSB overshoots and would also reduce the 3 dB multiplex power penalty that pure SSB introduces.

Bob Orban

Bob,

As you know, until more work is done, we won't know the finalized outcome. Yes, the overshoots exist. The trick is in how we choose to deal with those. As you say, we may find a suitable method, when benefits are beyond what's possible now.

I wasn't doubting you. I'm doubting another who is basing his opinion on white papers, as compared to further testing of the info presentated in a white paper. There's a difference.

-Frank Foti
 
FFoti1 said:
This is totally wrong! Now your comment is inappropriate. If anything, we went out of our way to credit Bill Gillman for his paper. Also, there are hundreds who've heard my public presentations on this topic, and each time I have given credit to where credit is due. What I have said, is we were the first to offer a public demonstration of the method.

-Frank Foti

As long as you remember that it wasn't YOUR idea, I have no problem. But I feel that line has been blurred a bit. The causal observer, including my DOE, thinks it's your idea when in fact I had to show him the original paper.

Your name carries alot of clout, as does Mr. Orban. I'm sure if Orban and Vorsis and everyone who is implementing this has done so by now... they must have had these ideas for awhile as well. I cannot say for sure, but I would venture that this takes alot more than just flipping a switch based on what I've read.
 
FFoti1 said:
I wasn't doubting you. I'm doubting another who is basing his opinion on white papers, as compared to further testing of the info presentated in a white paper. There's a difference.

-Frank Foti

So what's the purpose of a white paper if not to state facts and encourage opinions? And Bob's white paper is a thorough test of this of what it does with the hardware and software available now. Can you come up with a limiter that can take care of the issues? Maybe, but the challenge is to do it without further destroying audio quality or damaging the subcarrier domain.
 
BabyDJ said:
Honestly, I'm not a big fan of your processors and have been hounded into trying to get an 11 by my management because of SSB. I have a 6, an 8400 and two Vorsis AM processors and wanted my own choices to replace my older gear. I'm actually happy that there are others that have come out with it.

We at least know where you stand! The reasons behind your tone is well understood by everyone now. You don't have to like our products. That is fine.

Your DOE may have assumed that the SSB idea was our idea, but no where did we claim this to be so. Maybe you should take this up with the DOE.

We have credited all who have worked at SSB FM Stereo in the past, and we are simply trying to take their experiments to the next level.

We have stated that the level of today's technology has made this possible to "play" with the idea in a way that is much easier now than in the analog days when others have tried it. There are many different versions of SSB to play with, and we are now at a place to really see where this concept can take us. So we put it on the table, and others have since jumped in the SSB game to play as well.

When this idea was introduced years ago, anyone who is anyone in audio processing has dreamed of messing around with SSB and seeing if there maybe is something in there to allow us (as an industry) to maybe improve on things. This is all we're trying to accomplish here.

I have nothing more to say...for me this topic is closed.

Thank you.

Regards,
-Cornelius
 
BabyDJ said:
As long as you remember that it wasn't YOUR idea, I have no problem. But I feel that line has been blurred a bit. The causal observer, including my DOE, thinks it's your idea when in fact I had to show him the original paper.

Your name carries alot of clout, as does Mr. Orban. I'm sure if Orban and Vorsis and everyone who is implementing this has done so by now... they must have had these ideas for awhile as well. I cannot say for sure, but I would venture that this takes alot more than just flipping a switch based on what I've read.

Then, you never really had a problem with us, as we NEVER claimed, insinuated, or alluded that it was our idea. Any blurring as you say, is on your end. As stated prior, our paper, and subsequent article makes clear mention of William Gillman's paper, and that alone is testament by us towards whom gets credit for the idea in the first place.

Further, I called Bill over the summer to see if he wanted to be involved with this. He told me he didn't, but was aware of the what was going on, and told me he'd be happy to support any further activities regarding the topic.

Considering how our industry has chosen to view our company, and me personally, I took great care to make sure how this topic was postured. Yes, I admit that we were the first to bring it to light recently, but we NEVER took credit, or even tried to do so with respect to the creation of the concept itself.

I respectfully ask that you have a look at your own personal 'distortion-canceling' mechanism you employ with regards to topics of this nature! :)

-Frank Foti
 
BabyDJ said:
So what's the purpose of a white paper if not to state facts and encourage opinions? And Bob's white paper is a thorough test of this of what it does with the hardware and software available now. Can you come up with a limiter that can take care of the issues? Maybe, but the challenge is to do it without further destroying audio quality or damaging the subcarrier domain.

Bob's paper is one perspective, of which there's many. It appears you've chosen only his view, as the be-all, end-all. Hopefully, you'll broaden your horizons a bit and give consideration to more information, as it becomes available.

You are correct. The challenge is to create a SSB stereo generator that does not degrade the sonic integrity of the audio signal. So far, it seems we may have this, and we'll know more about it very soon.

-Frank Foti
 
It's my understanding that, using the current DSB stereo FM standards, L-R energy gets to 'sneak in' under L+R's level (especially important when composite clipping is applied), because its peak level is generally 6 dB lower than that of the Main Channel L+R's.

If I am following the SSB thread correctly, it would appear that by bumping up its peak level by 6 dB, the L-R must then accept an equivalent amount of increased clipping (or other processing) with regard to that which may be generally applied present-day to a composite signal.

At the moment, I don't have the means to test this, but I would surmise that 6dB more L-R clipping than present practices may have a serious downside. Or am I off-base? Composite processing is already really pushing the quality-perception envelope. At the least, I would anticipate SSB will thus have a different sound texture than the DSB stereo we all grew up with.

No matter how you slice it, the SSB concept is fascinating! Glad to see it generating so much discussion!

Kind Regards,
David
 
David Reaves said:
It's my understanding that, using the current DSB stereo FM standards, L-R energy gets to 'sneak in' under L+R's level (especially important when composite clipping is applied), because its peak level is generally 6 dB lower than that of the Main Channel L+R's.

If I am following the SSB thread correctly, it would appear that by bumping up its peak level by 6 dB, the L-R must then accept an equivalent amount of increased clipping (or other processing) with regard to that which may be generally applied present-day to a composite signal.

At the moment, I don't have the means to test this, but I would surmise that 6dB more L-R clipping than present practices may have a serious downside. Or am I off-base? Composite processing is already really pushing the quality-perception envelope. At the least, I would anticipate SSB will thus have a different sound texture than the DSB stereo we all grew up with.

No matter how you slice it, the SSB concept is fascinating! Glad to see it generating so much discussion!

Kind Regards,
David

David,

I used to think about the modulated L-R as you described it, but it's not the case. In DSB mode, both sidebands are down 6dB, as you observe them on a spectrum analyzer, with respect to the L+R signal. But, the composite of the two sidebands adds up, and their amplitude is the same as would be a SSB carrier, with 6dB gain added to it.

If this description doesn't make sense for you, email me, and I'll share some pix of the two signals.

-Frank Foti
 
Indeed, Frank -- I think this whole thread might be based on a misconception.

In a DSB Stereo subcarrier, the LSB and USB do not actually cancel -- they add. If you completely remove a certain frequency range from the USB, you lose 6dB in that frequency range.

For example, my composite clipper back-end, when used in RDS protection mode, yields 17.5 KHz frequency response but with reduced stereo separation above 16 kHz, due to cutting out everything above 54 KHz in the name of RDS protection. This causes a -6dB shelf in the decoded L-R between 16 and 17.5.


If we consider a theoretical, unlikely analog tuner with flat MPX frequency response up to 38 kHz, and then a gentle rolloff above that, SSB could actually improve the stereo separation performance of this tuner -- unless of course they compensated for their MPX frequency response deficiencies by adding gain to the L-R signal.


Best regards,
///Leif
 
David Reaves said:
It's my understanding that, using the current DSB stereo FM standards, L-R energy gets to 'sneak in' under L+R's level (especially important when composite clipping is applied), because its peak level is generally 6 dB lower than that of the Main Channel L+R's.

If I am following the SSB thread correctly, it would appear that by bumping up its peak level by 6 dB, the L-R must then accept an equivalent amount of increased clipping (or other processing) with regard to that which may be generally applied present-day to a composite signal.

At the moment, I don't have the means to test this, but I would surmise that 6dB more L-R clipping than present practices may have a serious downside. Or am I off-base? Composite processing is already really pushing the quality-perception envelope. At the least, I would anticipate SSB will thus have a different sound texture than the DSB stereo we all grew up with.

My white paper discusses the peak modulation of SSB in considerable detail. Assuming that the SSB modulator introduces no group delay distortion, and assuming that the audio processing applied to the DSB and SSB generators is the same, the following conditions will produce the same modulation in the DSB and SSB cases:

1. Pure L+R
2. Pure L-R
3. Single-channel only
4. in-phase material panned anywhere in the stereo soundfield (i.e. what would occur if you took a mono track and panned it from fully left to fully right)

What *does* cause overshoots in the SSB modulation is material that is uncorrelated on the left and right channels. The "two identical sinewaves with 90 degrees phase shift between them" that I discussed in my white paper may be the worst case, producing about 138% modulation (assuming 9% pilot injection).

The best way to get a gut feel for this is to look at the composite waveforms on an oscilloscope and realize that the SSB subchannel does not interleave with the main channel; it adds to it. The SSB modulator takes the L-R, inverts the spectrum, and arithmetically frequency-shifts it upward by 38 kHz. A simple example is 1 kHz left-only modulation, which will produce equal amplitude 1 kHz and 37 kHz sinewaves in the composite. At certain times, the peaks of the 1 kHz and 37 kHz sinewaves will coincide, producing peak modulation that is twice that of the 1 kHz and 37 kHz tones considered separately.

Because of the loss of phase coherence between the L+R and L-R signals caused by the frequency shift and spectral inversion (even in an SSB modulator that adds no excessive group delay distortion because its lowpass filters or Hilbert transformers have linear phase), the worst-case peak modulation is the sum of the magnitudes of the L+R and L-R signals. There is no interleaving, unlike the DSB system. If you look at Fig. 11 in my paper (the vector summation of left and right signals to produce the L+R and L-R signals), it is clear that in the worst case, the sum of the magnitudes of the L+R and L-R signals can cause the peak modulation to increase by a factor of SQRT(2) (141%) compared to the peak modulation produced by the DSB system. When you add the pilot tone into the picture, the worst-case overshoot becomes about 138% because the pilot modulation is the same for both systems.

While I can't yet prove mathematically that the maximum overshoot for arbitrary signals in the left and right channels is also 138%, my intuition suggests that it is. I base my argument on the fact that peak modulation is an instantaneous measurement, so the modulation measurement criterion is simple compared to, for example, r.m.s., which involves an integration over time. If you look at Fig 11 in terms of the vector sum and difference of L and R, there is no combination of L and R that can cause the instantaneous sum of |L+R| + |L-R| to be more than 2 * SQRT(2), assuming that |L| and |R| are both constrained to 1 by the audio processor preceding the stereo encoder and that the SSB generator's filters have constant group delay so they do not cause this constraint to be violated. (The |x| notation means that you take the magnitude, i.e. the length of the vector, ignoring the phase.)

The 90 degree relationship is also known mathematically as a "quadrature" relationship and is related to the mathematical concept of "linear independence." My working hypothesis is that linearly independent left and right signals whose magnitudes are constrained to 1 and whose bandwidths are constrained to less than 19 kHz can produce overshoots as large as 138 % modulation but no larger in the SSB system. This hypothesis could be falsified by anyone who could supply an example of a pair of L and R signals that met this criterion but produced larger overshoots. While I don't want to take the time to do this right now, it would be worthwhile to consider combinations of sinewaves making up the Fourier elements of a squarewave because once can choose combinations whose peak level is lower than the peak level of the sum of the magnitudes of individual sinewaves.

Exercise for the reader: consider the sum of a 1 kHz and 3 kHz sinewave, where the amplitude of the 3 kHz sinewave is 11.11...% of the amplitude of the 1 kHz sinewave and they are phase-locked such that every zero crossing of the 1 kHz sinewave is time-coincident with the zero-crossing of the 3 kHz sinewave.

or expressed as an equation

L = 1.125 [SIN(2 pi 1000 t) + 0.1111 SIN(2 pi 3000 t)]

(The 1.125 factor makes the maximum peak level of L = 1.)

Find an R signal that meets the "magnitude<= 1" and bandwidth constraints that causes the SSB peak modulation to exceed 138%.

Bob Orban
 
Thank you Frank, and thank you Bob for responding to my query with clarity and offering a fuller understanding of the subject.

It reminds me how great it is to be able to interact here with the movers and shakers of the industry in a civil fashion, and how the higher up you go, the more freely the folks are with information (and patience!).

Assuming Bob's data holds, that would indicate the efficacy of a means to sense and react to diminished coherence as part of any stereo/SSB clipping process.

Correct?

Kind Regards,
David
 
David Reaves said:
Thank you Frank, and thank you Bob for responding to my query with clarity and offering a fuller understanding of the subject.

It reminds me how great it is to be able to interact here with the movers and shakers of the industry in a civil fashion, and how the higher up you go, the more freely the folks are with information (and patience!).

Assuming Bob's data holds, that would indicate the efficacy of a means to sense and react to diminished coherence as part of any stereo/SSB clipping process.

Correct?

Perhaps. Our first-pass modulation controller is a finite attack time look-ahead limiter applied to the SSB baseband waveform, followed by our half-cosine interpolation composite limiter to take care of overshoots that pass through the look-ahead limiter. By acting on the composite waveform, this processing (which is spiritually akin to an AM Volumax -- thanks Emil!) automatically takes the overshoots into account and no fancy extra processing in the left/right domain is necessary. We have by no means optimized the simple, first-pass composite processing yet, but I think that the logical place to put the SSB overshoot controller is definitely in the composite path. The good news is that if I'm right about the maximum overshoot that SSB can induce, the necessary peak reduction is always less than 3 dB, and it's not too hard to peak limit by 3 dB without objectionable side effects. However, having no objectionable side effects while preserving the same loudness that one gets in DSB modulation is a difficult problem and we are not there yet.

That being said, my thoughts right now are moving more in the direction of defining the transmission standard so that it reduces the SSB-related overshoots that the modulation controller needs to correct. One promising direction, which we have not yet tried, is to transmit lower frequencies in DSB mode, as I said in an earlier post on this thread. This is one place where you get a free lunch because the first seven audible musical octaves fit into the 0 to 2 kHz range, so transmitting a considerable proportion of the total program energy in DSB mode would only add about 2 kHz to the baseband spectrum.

Bob Orban
 
rorban said:
David Reaves said:
Thank you Frank, and thank you Bob for responding to my query with clarity and offering a fuller understanding of the subject.

It reminds me how great it is to be able to interact here with the movers and shakers of the industry in a civil fashion, and how the higher up you go, the more freely the folks are with information (and patience!).

Assuming Bob's data holds, that would indicate the efficacy of a means to sense and react to diminished coherence as part of any stereo/SSB clipping process.

Correct?

This is an interesting thought!

Between what we're working on, and what you are working on, I think there will be some interesting solutions available for this option of transmission.

-C

Perhaps. Our first-pass modulation controller is a finite attack time look-ahead limiter applied to the SSB baseband waveform, followed by our half-cosine interpolation composite limiter to take care of overshoots that pass through the look-ahead limiter. By acting on the composite waveform, this processing (which is spiritually akin to an AM Volumax -- thanks Emil!) automatically takes the overshoots into account and no fancy extra processing in the left/right domain is necessary. We have by no means optimized the simple, first-pass composite processing yet, but I think that the logical place to put the SSB overshoot controller is definitely in the composite path. The good news is that if I'm right about the maximum overshoot that SSB can induce, the necessary peak reduction is always less than 3 dB, and it's not too hard to peak limit by 3 dB without objectionable side effects. However, having no objectionable side effects while preserving the same loudness that one gets in DSB modulation is a difficult problem and we are not there yet.

That being said, my thoughts right now are moving more in the direction of defining the transmission standard so that it reduces the SSB-related overshoots that the modulation controller needs to correct. One promising direction, which we have not yet tried, is to transmit lower frequencies in DSB mode, as I said in an earlier post on this thread. This is one place where you get a free lunch because the first seven audible musical octaves fit into the 0 to 2 kHz range, so transmitting a considerable proportion of the total program energy in DSB mode would only add about 2 kHz to the baseband spectrum.

Bob Orban
 
Status
This thread has been closed due to inactivity. You can create a new thread to discuss this topic.


Back
Top Bottom