Improving audio fidelity for HF single sideband Microphone content has been split to another page |
<<<<<<<< UNDERGOING RECONSTRUCTION >>>>>>>>> Although somewhat different animals, AM-FM audio processing techniques can be applied to ssb transmission to achieve some improvement in the audio quality. The first step is to ask what exactly we are trying to achieve? Maximum loudness equates to the highest rms value audio waveform, usually produced by a combination of dynamic compression and clipping. Maximum fidelity minimizes distortion and maximizes frequency response. There is significant controversy about unnecessarily wide occupied voice channel bandwidth being " more than your fair share." You have to ask yourself whether or not plus or minus 15 KHz sidebands (in the case of am) yielding 50 Hz to 15 kHz audio frequency response is necessary, advisable, or responsible. I can't answer that but I can tell you that a100 Hz to 7kHz voice channel sounds very natural. In the case of ssb, adding just a bit more to the 2.4 or 2.7 kHz normal , bandwidth say 500 Hz, adds a lot to the perceived quality. Back to our basic question: Am I after maximum intelligibility, or the most natural sound? You can't maximize both, but you can get close - my personal choice is somewhere in between. |
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In the pursuit of optimized ssb, bandwidth-extended ssb, or full carrier AM audio quality, I think most hams start out looking in the wrong place. Here are the key goals -
The most important factors for low distortion ssb and essb are (essentially) choice of rig, and optimal drive level adjustment - which I personally can't do without seeing the rf envelope on an oscilloscope. While surfing some "hi-fi" ssb and am web sites, I see some hams have virtual recording studios grafted onto their hamshack, including very expensive microphones, but no scope to get maximum peak output without clipping (or negatively over modulating in the case of am). I can only presume that these folks are unfamiliar with its use. The scope is a very useful tool in adjusting for best quality. Two-tone testing with a spectrum analyzer measures the non-linearity of the ssb generation and amplification, which causes audio distortion, but isn't practical for most hams. I can tell you that the Kenwood 870, with its dsp generation, is among the best I've seen in this respect. With a bit more audio awareness in recent years, several other recently designed rigs have also improved this aspect. The other major
quality determining element (perhaps the one which most people
perceive as high quality) - limited primarily by the rig is the
audio frequency
response. Traditional ssb
transmitters generate double sideband using a balanced
modulator, then pass only usb or lsb through a crystal or dsp filter, the
width of which determines the maximum audio bandwidth (frequency
response). Typically this is quantified at the -3 dB points,
for example at 300 hz and 2700 hz may be 3 dB below the center response.
Although certainly adequate for basic voice intelligibility, it's far
from natural
sounding. A 2100 Hz bandwidth (300-2400) sounds even more unnatural.
The 870, using dsp generation with front-panel variable filtering, can
pass lower than 200 hz and greater than 3000 hz at the -3dB points, which,
assuming a receiver is used which has appropriately wide IF bandwidth and
audio response, makes a noticeable improvement.
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| Transmitter
frequency response can be measured with an audio tone generator (with
an appropriate resistor pad reducing it to mic level) slowly
sweeping the 50hz - 5khz audio range at about 30% transmitter
output power, while noting the wattmeter. When 15% power is
shown, the -3 dB point is indicated. Once the transmitter and microphone frequency response is known and charted, an equalizer can be set to correct ( or pre-correct ) many shortcomings. Initially, maximally flat response should be the goal, beyond that personal preferences can modify that. Maximum intelligibility will always be had with boosted presence. (2000 - 3000 Hz - up to about 5-6 kHz for am), but some of the "hi-fi" guys seem to like bass boost - perhaps this is to offset some perceived shortcoming? |
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The equal-loudness curves shown at left are derived from the original Fletcher and Munson human hearing research. The 2 kHz to 4 kHz droops essentially define the presence band which add intelligibility to speech. | ||
| One benefit of ssb, aside from its enhanced modulation efficiency when compared to am, is to communicate using the smallest practical bandwidth - the receiver hears less noise that way, and highest signal-to-noise transmission is achieved through the noisy medium. There are major advantages to be gained by reducing the dynamic range of the transmitter audio through compression and clipping, which we'll discuss later. But when band conditions permit good signal to noise, its remarkable what a seemingly minor frequency response increase can do for the "naturalness" of the audio. Speaking of conditions, remember that efficient antennas and more transmit power go a long way toward improving the signal-to-noise. QRO from 100 to 1,000 watts increases your received signal-to-noise ratio by exactly 10 dB, every time. | |||
| There are "hi-fi" fad-type amateur processing and eq products available, but I suggest standard pro or semi-pro equipment, which can be had for only slightly more money. | |||
| Professional microphone processors can cost more than your hamshack but there are several that are economical and do a wonderful job. Typically these units consists of a preamp to bring mic level (-60 dBm) up to line level (~ 0dBm), and then add certain enhancements. Dynamic range compression is found on most units as is some form of bass/treble (equalization) control. Another common feature is downward expansion gate which is sort of a subtle vox - when speech pauses, it reduces gain slightly to lessen apparent background noise. De-esssing is also a common process (This dynamically reduces sibilant treble when excessive sss are present - but most of the sss are above 3 kHz anyway.) The dbx 286A does this job well and includes input and output RF filtering at about $175. | |||
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| A third octave equalizer like this Dod unit for $99 has 31 bands of individual adjustment which are useful for eliminating the frequencies which we cannot transmit, and therefore do not wish to be acted upon by dynamics processing. The above dbx unit conveniently provides for side chain eq insertion. I set it to cut everything below about 100 hz, and everything above 4000 hz. If my transmitter had -3 dB points at 300 and 2600, I would boost those frequencies some to compensate for the filter roll off, which would widen it some. | |||
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I alternate between the 870's internal audio processing which is actually very good (but nowhere close to flat frequency response, as they tailor the response to split the voice formants, processing them separately), and using just the agc with no clipping. The AP makes for louder, denser audio, but the frequency response is somewhat peculiar. While normally any clipping of the waveform is taboo in pro audio circles, the reality of HF communications is that it can gain you better apparent signal to noise through higher average transmitted power, at the tradeoff of slight peak distortion. It is however an outright distortion generator, so must be used sparingly to preserve good quality. With AM modulation, broadcasters generally use asymmetrical clipping, set to brick wall clip downward (negative) modulation peaks at 95% to avoid splatter, and clipping the positive (louder!) excursions at 125%. Most ham AM transmitters are pretty poor positive modulators. FM broadcasters, even those with formats that don't allow aggressive processing, (not many) use some amount of clipping to prevent overmodulation. At the time of
this revision in May of 2009, I'm currently building a
dedicated broadcast quality audio pre-processor adapted for ssb
speech, using phase scrambling and EQ contouring. Component values
may be adapted to favor AM. some schematics are below - stay tuned
for more details published here. |
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The overall block diagram of a typical high fidelity processor is shown at left. If high ambient RF is not a concern, a really good preamp can be put together from a dedicated $8 Analog Devices chip. A very similar unit from THAT corp is here. In my case, with RF present, I prefer transformer isolation, using high quality mic transformers like the UTC ouncer 0-8 series with shield and shock isolation. This traditional approach works very well, but good transformers can be expensive. Jensen is another good brand. | ||
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| Just after the mic preamp, it's a good idea to band limit the audio to near what will eventually be transmitted. This prevents gain-reduction caused by spectral energy which won't make it all the way through the filter chain. It also minimizes the frequencies available to produce intermodulation. The low corner may be set anywhere between 50 to 300 Hertz, and the high roll off set at 3200 to 7000, depending upon your mode and your goals. The second-order filters shown are cascaded in as many sections as desired - I recommend 2 high-pass, and 3 low-pass. Fewer stages of higher order circuits can also work, but can be fussy to adjust. The pots in the circuits shown control the passband flatness / ultimate slope tradeoff. |  |
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An ALL-PASS? Speech as well as certain music is often very asymmetrical - that is to say the positive portion of the waveform can be much different than the portion below the zero line. Dynamic processing will be more exacting and efficient with improved symmetry - that's why virtually all modern broadcast audio processors use some means of phase-rotation. The technique imparts variable phase shift across the audio passband, particularly in the bass-midrange region - about 500 Hertz. With many types of natural-source audio, symmetry is in fact improved. For full-carrier AM, a negative clipper may ultimately follow to limit negative modulation to about 95%, while letting positive peaks supermodulate. This method was the basis of Leonard Kahn's famous Symmetra-Peak AM pre-processor, introduced for broadcast in the 50's. The block diagram for his patent is shown at right, as well as a depiction of the effect on the audio waveform. For SSB, where both positive and negative audio peaks create RF equally, symmetry contributes to higher rms level. All-Pass and "phase-scrambler" are essentially synonymous. The Op-Amp circuit shows one section. Often two or more are cascaded. |
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No audio discussion would be complete without opinion. The current state of audio equipment merchandising is so sad; in no other segment of electronics can you find more psychoacoustic snake oil successfully sold. I estimate that over half the money spent on "high-end" equipment is entirely wasted on hyped and often misleading features and specs. A perfect example is the retro-fad use of tubes in audio stages. One computer motherboard manufacturer even boasts about their "tube" output amplifier for its built-in sound card output stage. There are those who will swear by their $300 tube "preee" (pre-amplifier) instead of a $3.00 op amp. Having personally done proof-of-performance testing on tube broadcast equipment ( high-dollar professional gear ) in the 70's, and since then, many generations of solid-state equipment, let me bear witness to the fact that, with all other things being equal, modern solid-state devices excel at audio, and tubes can only come close at best, and are awful at worst. One argument heard is that tubes clip less abruptly, therefore producing distortion which is less harsh sounding; (This from the guitar amplifier crowd.) Certainly this is true, but the argument is ridiculous because when properly operated, one should never reach the clipping point in a normal amplifier - if you do this by overdriving a stage, then you're intentionally introducing distortion, the prevention of which is what our best efforts are all about. Tubes are microphonic, noisier, are in a constant state of deterioration, and due to the high impedances involved, necessarily require transformers, which unless very high quality, add low-frequency distortion. And otherwise sane people pay extra for these features! "...but I can hear a difference! " is frequently cited as the entire rationale. I don't mean to suggest that analytical listening has no place, but when subjective tastes trump actual measured distortion and tonal colorations, we're ignoring reality. And they probably are hearing differences - just as the 17 year-old generation, accustomed to hearing low-bit-rate MP3 distortion, think something is wrong when they hear the original non-bit-reduced .wav file. People have expectations, and sometimes folks prefer unnatural sounding audio. So, the ideal of tube audio being superior in any way other than replicating nostalgic coloration is simply nonsense. Similarly there are others who will pay huge amounts for gold-plated, de-oxygenated speaker cables. "A fool is born every minute" - P.T. Barnum. Rule-of-thumb: don't believe audio discussion which avoids technical metrics like signal-to-noise ratio, frequency response, harmonic and intermodulation distortion, but is instead liberally spiced with hyperbole. Meaningless, often misleading terms are typically used by salespeople to persuade you to buy. Or by newbies who are ignorant of the actual technical methods. I'll state that outside of certain high-power RF and a few other very specialized applications, the only good argument for tube equipment is for nostalgia - and for precisely that reason, I own a Johnson Viking Ranger, a Collins 75A3, and am currently restoring a 1937 National NC-101X. I also take pictures of nice looking RF tubes |
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