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what is Hi-Fi SSB ?

B

BOOTY MONSTER

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its a new section so why not explain what it is to someone who doesnt know ? i certianly dont know and i figgure i cant be the only one .
teach this parrot some new words to say :shock: :twisted: :LOL: :LOL: :LOL:
 

Hey, this new forum look is far easier to read...I like it.

ESSB, enhanced SSB generally refers to HF operators as it involves heavy modifications of rigs and many times quite a bit of money in added audio gear.

The basic principle is to get a wider and fuller fidelity signal. Done well it can approach AM, though being SSB it won't have all that bandwidth to use.

A nutshell explanation would be.

1) Widen the passband of the transmit. With many radios allowing 2.2Khz that is a narrow window of frequency response. ESSB tends to start at 2.8Khz up through 3KHz. This allows you a wider frequency spectrum of audio to transmit. With 3kHz, you could theoretically pass 300Hz to 3300Hz or shift that higher or lower with carrier offset in the rig.

Lower lows and higher highs than standard SSB. This gives you more fullness and generally a more natural sounding signal.

2) Widen the passband of the receive. Depending on the rig, you may not be able to actually receive and as a result, hear the wider signals that the first stage of modification can enable.

Your receive has to capture the full width of an incoming signal or it won't be able to process the full information that is transmitted.

3) Managing the dynamics of your signal. Using compressors applied properly you will enhance the power output on SSB. Basically trimming down the peaks to meet the average level makes your signal more punchy and efficient.

Broadcast audio, which this whole exercise imitates, is fully optimized through layers of compression and dynamics handling.

4) EQ and managing the tonal characteristics of the audio. When you widen your passband and possibly set your carrier offset to favor more lows, you can introduce muddiness into your signal making it harder to understand. Using EQ properly, you can then clean up the response of your mic and rig to match your own voice properties.

There are a lot of myths about EQ and microphones out there. The bottom line truth is, you need to be set up for YOU. Each operator's voice and manner of speaking will vary widely, so one man's setup won't necessarily work for the next guy.

As you introduce a compressor, it can put emphasis on unwanted spectral ranges, the low midrange in the area of 500Hz or so as one example. This can make your audio sound flat and boxy. An EQ is critical to balance these 'hot spots' in your audio...if applied correctly.

One quick piece of advice from a former sound engineer. An EQ of any quality, should never be used to boost frequencies.

One should always use an EQ to remove unwanted frequencies. This way you don't introduce distortion to the signal. If a signal sounds like it needs more lows, the other way to look at it is, that the signal probably has too much of a mid-range element. If you simply attenuate the mid-range, the lows in effect will come up. As you remove with an EQ, you will then need to boost the gain up a bit after the EQ in the chain, as you are removing signal level.

5) Effects processing. This is a sticky subject. Though not illegal to use some reverb and delay in the amateur service, I personally don't like to hear high levels of it. You definitely don't want the CB type 'echo' sound, but if you understand studio quality reverb and delay setup, you can get a pro-audio quality to your signal.

Think along the lines of the local Z100 pop station where the guys have the booming reverb on their audio. The trouble here is, what you think you are transmitting many times will have much more effect on the receiving side. It is critical to have actual over the air monitoring of your signal to get the levels correct. The ideal here is that the effects wouldn't be immediately noticed unless they were disabled.


The basic chain is

microphone to a studio mixer

Inserted effects processors, reverb/delay.

compressor inline with the mic audio chain after effects

EQ after the compressor...or before...it's a bit of a battle of opinion here.

mixer output into an impedance transformer to match the rig's audio input.

Or alternately, feed the line level out directly into the balanced modulator circuit in the radio itself....this takes some know-how though and you are essentially bypassing all the rig's built in protections to keep your signal clean.

This was only a nutshell explanation if you can believe that...poke around on google for ESSB for more in-depth explanations of the mods involved.

The Kenwood 850/870/940/950 are pretty commonly modified for this use.

Ten-Tec Jupiter radios can pass 3k as stock as can a few newer high-end Yaesu rigs.

The old boat-anchor rigs are also easily modified for this use.

One last bit, if you modify a rig for ESSB by changing filters in it, it will generally be pretty awful for more difficult use like DX chasing or contesting. You are opening the rig up, defeating some of the efficiency and selectivity the filters are actually set up for.
 
I forgot to add this mention.

The KING of ESSB if you want something that just comes out of the box, would be the flex-radio Flex-5000.

http://www.flex-radio.com/

This is a software defined radio. The extensions to it are nearly limitless and out of the box it is a broadcast studio with what it can do.

It requires a solid computer to control it whenever you operate though.

It's not for everyone, but it is the audio freak's radio of choice...it's the grail.

It's actually too good for the experimenter types as there's nothing to do to it once you own it.

I'll eventually own one of these...it's unavoidable for me with a sound production background.
 
That's a very fundamentally sound description of what ESSB or HiFi SSB is all about.

Here's the setup I use:

-Icom 746Pro
-Heil GM mic with HC-5 element, but used on the narrow setting (wide sounds terrible with my voice)
-W2IHY 8 band EQ
-W2IHY EQPlus

It takes a while to play with it to get it to sound right. You have to mess with the settings on the mic, the bass/mid/treble settings on the rig, the mic gain settings on the rig, and also all the EQ, compression, downward expansion, etc. settings on the pre-processing. Oh yeah, and then you need to take notes on what to change if you switch over to AM or FM. It's pretty fun to play with, but definately not "plug and play"
 
BTW, I said that the mic on the wide setting sounds terrible with my voice, but that's just not quite accurate. With the 8 band EQ, I can make it sound really pretty good on wide mode. However, it just sounds too over-the-top to suit me. My "radio voice" is pretty dynamic and well rounded (people say I sound like a TV or Radio guy over the air) and the wide setting makes it sound to "hot" for my tastes. Maybe I just don't quite know how to set it, but everyone seems to like it the way I have it, so....
 
With that 8 band EQ, if you haven't tried this approach, give it a shot.

Level out the EQ to 0 all the way across.

Go band by band, dropping the EQ to at least the half point or farther until you hear the 'trouble' frequency. As you pass each band, reset it to zero before you go to the next one.

Generally there is a specific range that is too hot that makes the rest of the quality sound flat.

You'll find one of the bands that seems to suddenly clear up the audio definition. Once you find it, start adding it back in a bit until you find the threshold. Then take the two neighboring sliders and dip them a bit to split the distance between zero and where the first selected point wound up. You basically wind up with a slight 'V' with the bottom point being the first 'problem' slider you found and the two next to it coming down to meet it.

After you drop an EQ like this, it will drop your output level. Since you also have the EQ Plus box inline, you can then make up for the drive level there and bring it back to the ALC deflection you are looking for.

A good guess for the trouble point is 500Hz or 630Hz area, not sure exactly what the 8band has in that range. That range causes the midrange to sound honky or boxy. If you drop that out a bit depending on your voice of course, it will bring up the bottom and high end, making your signal sound more defined and fuller.

This is basically the 'proper' approach to using any EQ whenever possible. This is mostly because it prevents you from introducing distortion from boosting levels. It's also much quicker to drop out an offending range than basically boosting everything around it to wind up with basically the same overall response curve.

The reason for dropping the two neighboring sliders is to make the response curve more even. It intersects the slope points of all three bands and should give a smoother result.

I'd be curious if you try it, how you find that it worked out. This is the system I generally run through with anyone using Julius' boxes and it seems to really work out well.
 
One last tip as well.

When chaining audio devices, gain staging of each component is critical.

In a nutshell, your drive levels should look like an inclining slope with the 'front' most component at the lowest level of the chain.

If all your devices had even output levels and input sensitivity and all had level controls 1-10 on them, you would as an example have a level of 5 at the first device, 6 at the next, 7 at the next, in a balance to the final input.

If you drive too hard into one stage and throttle it with the next and then boost it again at the next, you are almost certainly introducting distortion on the signal path.
 
If you're thinking of operating a higher bandwidth SSB signal, be careful:

http://www.w8ji.com/transmitter_splatter.htm

Obviously the guy is biased against ESSB, but just be aware that you need to be more concerned with your intermod products and splatter.

The FCC rules states that bandwidth for phone emissions cannot exceed that of standard AM modulation. So you should be well within that if your transmitter is clean.

Also consider more modern modulations like DRM, or WinDRM that was made for ham bands. Being an error corrected digital modulation, it should be do better than standard AM within the same bandwidth.
 
That's a great link actually.

Reminded me that I should have mentioned that, at least in my own eyes, the only bands that should be used for ESSB are 80m and 160m. These days 10m would be o.k. as long as propagation was not up.

ESSB and AM are hog modes and should only be used in appropriate areas of the bands.
 

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