Posted: 4/2/2015 8:03:52 PM EDT
I have a very rough understanding on what this does but would appreciate a good explanation of what it is, how best to use it, when not to use it, etc.
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http://en.wikipedia.org/wiki/Dynamic_range_compression
Use a little all the time. Use a lot for DX/contests. The definition of "too much" is hard to define. It is very rig dependent. The best way to get a handle on this is to get a bunch of on the air signal reports by people who are willing to spend a little time helping you tune things in. You need to get a number of different people involved because, while there is an obviously gross level of "way too much", figuring out where a more subtle level of "too much" is is a very subjective thing. The other way you can judge this is by watching your RF power meter. More compression will show a much higher average power level. There is a fine line between mic gain and compression as well. What kind of rig and microphone are you running? That might help people provide a little more guidance based on their experiences with that particular set up. |
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Listen to AA...I did.
I ended up with a Behringer 302 and I used online reports...a member here...and some of the various playback monitors on the web to get it tuned for 'best' audio. In the end the best combo for me was turning the 'low' down almost all the way, leaving the 'high' at neutral, and setting the comp at 2-3 on the 7200. The online playbacks are the best method to tune these IMHO...I think pic and/or gcw run one from time-to-time, but there are others. $35 mixer $10 mic $10 boom $0 advice from AA and I get nice reports and do well enough in pileups. I mean why settle for 1 knob to fiddle with when you can have 3 and 2 sliders? |
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Listen to AA...I did. I ended up with a Behringer 302 and I used online reports...a member here...and some of the various playback monitors on the web to get it tuned for 'best' audio. In the end the best combo for me was turning the 'low' down almost all the way, leaving the 'high' at neutral, and setting the comp at 2-3 on the 7200. The online playbacks are the best method to tune these IMHO...I think pic and/or gcw run one from time-to-time, but there are others. $35 mixer $10 mic $10 boom $0 advice from AA and I get nice reports and do well enough in pileups. I mean why settle for 1 knob to fiddle with when you can have 3 and 2 sliders? Which mic did you end up with? |
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Speech Compressor increases average transmitted power by reducing peak to average ratio of voice frequencies. In other words it makes weak sounds louder while clipping (reducing) level of loud sounds. If used properly, it helps to improve speech intelligibility and overall power output. Using speech compressor may add as much as one S unit on the receiving end and will make your signal easier to read. Too much compression may cause distortion and bring too much ambient audio noise into your audio.
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http://en.wikipedia.org/wiki/Dynamic_range_compression Use a little all the time. Use a lot for DX/contests. The definition of "too much" is hard to define. It is very rig dependent. The best way to get a handle on this is to get a bunch of on the air signal reports by people who are willing to spend a little time helping you tune things in. You need to get a number of different people involved because, while there is an obviously gross level of "way too much", figuring out where a more subtle level of "too much" is is a very subjective thing. The other way you can judge this is by watching your RF power meter. More compression will show a much higher average power level. There is a fine line between mic gain and compression as well. What kind of rig and microphone are you running? That might help people provide a little more guidance based on their experiences with that particular set up. My home rig is an Icom 7100 and Heil Proset IC. |
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Brundoggie--having owned an IC-7000 myself, I would keep it simple and start by following the instructions in the 7100 manual for setting mic gain and compression. That will get you 90% of the way there if my experience with the 7000 was any indication. The two radios operate almost identically where HF is concerned. However, in a bit more detail...
First and foremost set the TX bandwidth to it's widest setting. Even this is none too wide on the 7100, as it was on the 7000, but it will obtain you a better sound and intelligibility. I used to run my audio through the ACC port on the 7000 which netted me a full 3KHz of audio bandwidth. The ALC meter on the 7000 was notoriously conservative, so you might get away with running a bit more ALC than you might think on the 7100 as well. However ALC will act to limit the peaks of your speech and thereby cause distortion. I'd adjust the mic gain to keep indications at the lower half of the so-called "ALC zone" on the meter, but do have some ALC meter action to ensure you are fully modulating the signal because again that ALC meter is very conservative in the Icom radios and tends to exaggerate just how much ALC is really working. Finally, on compression this is the one thing that you are going to have to get an on-the-air assessment with. The so-called "compression zone" on the compression meter implies that you are adding between 10 and 20 dB of compression, however 10dB is a LOT. I'd stay around 10dB to start, max., and you may find that 3 or 6dB is a better place to operate. |
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The best way to get a handle on this is to get a bunch of on the air signal reports
by people who are willing to spend a little time helping you tune things in. I have a blind ham friend that helps me with on-air adjustments of my audio. "Bring it up a little in the 2 khz - 3 khz region... Good... OK, now bring the low end roll-off up to 200 hz. OK, perfect, don't touch a thing." He can describe just what he's hearing and what to do about it. He's about 400 miles away, so that gives us a good real world test of what it takes to get a decent signal through. The goal is NOT to sound like an FM jazz program announcer, but to have an intelligible signal. |
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I have a blind ham friend that helps me with on-air adjustments of my audio. "Bring it up a little in the 2 khz - 3 khz region... Good... OK, now bring the low end roll-off up to 200 hz. OK, perfect, don't touch a thing." He can describe just what he's hearing and what to do about it. He's about 400 miles away, so that gives us a good real world test of what it takes to get a decent signal through. The goal is NOT to sound like an FM jazz program announcer, but to have an intelligible signal. Quoted:
The best way to get a handle on this is to get a bunch of on the air signal reports
by people who are willing to spend a little time helping you tune things in. I have a blind ham friend that helps me with on-air adjustments of my audio. "Bring it up a little in the 2 khz - 3 khz region... Good... OK, now bring the low end roll-off up to 200 hz. OK, perfect, don't touch a thing." He can describe just what he's hearing and what to do about it. He's about 400 miles away, so that gives us a good real world test of what it takes to get a decent signal through. The goal is NOT to sound like an FM jazz program announcer, but to have an intelligible signal. This^^^^^ I've talked to some operators who were using several thousand dollars worth of audio processing equipment but sounded way below average. One guy was giving everyone advice on their audio but his audio was terrible with way too much base and muffled mid range. I had to open up the front end and set RX width to over 4kHz to be able to hear what he was saying. 
Most modern radios have internal TX audio equalizers and DSP processors. You need someone with a good receiver and good ears to help with tuning your transmitted audio. Self monitoring with earphones does not always work the best. I've been using a combination of an external EQPlus (by W2IHY) processor and FTDX3000's internal TX parametric equalizers to improve TX audio. So far the reports have been very positive. I still need to find someone with good ears to help me tweak the equalizers. I also found that using Effects Processor in the EQPlus really helps to break through major pile-ups. I set Delay and Level to about half scale. It gives the signal nice ambience without sounding obnoxious, like a CB reverb.  I also use Downward Expander in the EQPlus to eliminate background noise although it has not been an issue since I sold my Commander amp and replaced it with an Acom amplifier. The Commander's cooling fan sounded like a jet engine.
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Gyprat--you have an external EQ, compressor, limiter, reverb, downward expander/gate and internal EQ, but the guy with a few thousand bucks of audio processing equipment shouldn't? If you meant to say that "technology can be misused just as easily as it can be properly used" then certainly that has a lot of merit. If you are saying it's a waste to spend a goodly amount of money on audio processing then I respectfully disagree.
DirtNasty may have been joking a bit, but he's right: you CAN sound like an FM radio DJ and SIMULTANEOUSLY be highly intelligible. It's not that easy to do, particularly if your voice isn't FM radio DJ quality to start with (like your's truly ) but it can be done and lots of people are very successful doing it.
And it is most definitely a bandwidth sensitive proposition as you point out. I have the following profiles set up on my radio: 2.7, 2.9, 3.5, 4.0 and 5.0KHz for SSB, and 5.0KHz (really 10KHz) for AM. For SSB I use an equalization curve and 5-band compression scheme that maintains outstanding intelligibility over the entire range of TX bandwidths. The 2.7 and 2.9KHz profiles bump the low end up to 100Hz so that the low end does not dominate without the higher frequencies available in the larger passbands. The AM profile requires an entirely different audio processing scheme for good quality. You can be loud, you can be intelligible, and you can be "great"  Of the three choices I'll take the latter. It's taken me a year to get a handle on approaching the "great" level of complexity and I'm still learning. But it is a "great" feeling when the source of the DX pile-up takes time out to mention how nice my audio is, something that now happens quite frequently. That is a higher compliment then pretty much anything else where audio quality is concerned and if someone with my voice can achieve that it can only be because of a triumph of modern audio processing technology I'll also add that playing with this audio stuff has been hella fun and a great learning experience. It's added a whole new dimension to the hobby for me. It doesn't have to be all about making antennas, you know.
At any rate this is a pretty bad level of thread drift; this kind of complexity is not something that Brundoggie needs to worry about...yet! |
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Gyprat--you have an external EQ, compressor, limiter, reverb, downward expander/gate and internal EQ, but the guy with a few thousand bucks of audio processing equipment shouldn't? If you meant to say that "technology can be misused just as easily as it can be properly used" then certainly that has a lot of merit. If you are saying it's a waste to spend a goodly amount of money on audio processing then I respectfully disagree. DirtNasty may have been joking a bit, but he's great" Of the three choices I'll take the latter. It's taken me a year to get a handle on approaching the "great" level of complexity and I'm still learning. But it is a "great" feeling when the source of the DX pile-up takes time out to mention how nice my audio is, something that now happens quite frequently. That is a higher compliment then pretty much anything else where audio quality is concerned and if someone with my voice can achieve that it can only be because of a triumph of modern audio processing technology I'll also add that playing with this audio stuff has been hella fun and a great learning experience. It's added a whole new dimension to the hobby for me. It doesn't have to be all about making antennas, you know.
At any rate this is a pretty bad level of thread drift; this kind of complexity is not something that Brundoggie needs to worry about...yet! What I was saying is that some people buy expensive audio processing equipment but fail to set it up properly. It may have that DJ sound on good speakers with a wide bandwidth amplifier. It does not always work when trying to push it into a SSB signal with a limited bandwidth. It takes a lot of trial and error to tailor this to operator's voice, a specific microphone and limited SSB bandwidth. There is nothing wrong with adding complex, external audio processing equipment. A lot of guys do this with great success. What I heard people say is that they did not want to spend the money on additional audio equipment. When I asked if they tried adjusting TX audio parameters in their radios, they did not know it can be done. Sometimes changing a few menu parameters and properly adjusting compression, microphone gain and ALC level makes a huge difference in TX audio quality. |
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I do own expensive audio outboard gear. I have no desire to use it for amateur radio. The built in speech compressor in my radio is ideal for this use. If you want to do anything else to your audio I recommend doing it in the digital domain. Not older musical analog outboard gear.
If I want to use the DBX compressor in my rack first I have to figure out how to correctly convert/ adapt the signal levels and impedance. Both in and out of both units. Or additional rack devices, mixers, etc. If you don't convert correctly your signal will be much worse than when you started. IMO, its a waste of time and money when I see hams trying to connect a pile of Berhinger crap to low end radios. Spend your time and money on better microphones, headphones, DAC/ADC, speakers and room acoustical treatment. Maybe learn to use digital processing. And of course better antennas. If you want to spend money on audio gear I suggest high end DAC's like RME or Apogee, better quality dynamic mics, headphones and acoustic treatment. |
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Here's a little story about available bandwidth, recording, radio, and such.
Back in the very early days of recording the Victor Talking Machine Company (later Victor Records, RCA-Victor, etc) had a gadget that took over the place of the piano in the parlor, the "Victrola!". Yes, that record player that you cranked up to make it go, played records with bamboo needles, and amplified by a big megaphone type horn coupled to the diaphragm that was vibrated by the needle. Early radio was not much better. In those days a dance orchestra or studio orch was recorded with one microphone hung from the ceiling, suspended inside a hoop with little springs to help isolate from extraneous vibrations. The string bass or tuba were "lost in the mix". Those bass instruments could not be heard either on the radio, nor recordings. For one thing, the mics did not pick up low notes well, and on the other end of the audio chain, the speakers did not do much of a job with the low notes, either. The instuments began to change, favoring those that recorded well. One manufacturer of saxophones came out with their "Radio Improved" line, which had a brighter tone than other similar saxes. The banjo or Dobro were preferred over regular guitar due to the brighter tone. Dobro, I am not talking about guitars played on the lap with a steel bar, leading to the "Hawaiian guitar", but rather the acual guitar itself, which had a big steel resonating disk that gave it a brighter tone. 
They gained the name "steel guitar" not because of playing them with a steel bar (or neck of a bottle) but the steel resonating plate. But back to recording and radio... From the early days of recording it was found that the bass saxophone recorded well. You could hear the reedy tone well over radio and Victrola speakers. The listener was not really hearing the actual fundamental tone, but the reedy harmonics. The bass line could be heard clearly. If you look at pictures of the early Benny Goodman Orchestra there was no string bass or tuba, but a bass saxophone. The bass sax was not sitting up front with the other saxes, but back by the drums, piano, and guitar, in the rhythm section. So, in 1916 when you went down and bought your first Victrola the chances are you also picked up some records, many of which used the bass saxophone for the bass line. The tag at the end of the Three Stooges closing theme was bass saxophone, not tuba. Here's an example of that sound. Listen at 00:48. https://www.youtube.com/watch?v=WveEco2Rq2w https://www.youtube.com/watch?v=QabgfbnCJ6Y Later, as recording and playback equipment improved, going into the 1930's and 1940's, they switched back to string bass or tuba. So, there is an example of the early days, using instruments that fit the recording conditions, available bandwidth. |
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After reading the FLDIGI tuning instructions, I believe I've been overdriving my audio on the digital net. I'd been adjusting the level until the ALC meter is just inside its red zone. But the FLDIGI guys recommend adjusting for minimum compression. The guys on the IC-7000 Yahoo group measured this very precisely and the ALC meter is very conservative, showing ALC action where none exists. On the 7000 you can let some ALC show on digital and I always did. I suspect the same is true of the 7100, they seem almost identical in many ways. Remember to leave your RF power set to 100W (max.) and adjust for average output power using Fldigi. This allows some headroom for the amp to handle the peak-to-average ratio of the digital waveforms. 50W with the occasional peak above that should be GTG, although I've been known to run my 7000 a little harder than that
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I've got a 7100, and I don't use any compression unless I'm really trying hard to break a pile up.
The biggest thing is to drop the bass out of that radio. Go into the TX filters and drop everything below 200hz, and bring the treble up +2. Otherwise you're just very bass heavy. It's a shame you're not closer to me, or I'd help you dial it in. |
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The biggest thing is to drop the bass out of that radio. Go into the TX filters and drop everything below 200hz, and bring the treble up +2. Otherwise you're just very bass heavy. Is that just for the stock mic, or are you doing that with your MXL770, too? And is the audio from the 802 going into the mic, ACC or the DATA jack? |
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I've got a 7100, and I don't use any compression unless I'm really trying hard to break a pile up. The biggest thing is to drop the bass out of that radio. Go into the TX filters and drop everything below 200hz, and bring the treble up +2. Otherwise you're just very bass heavy. It's a shame you're not closer to me, or I'd help you dial it in. This. |
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Is that just for the stock mic, or are you doing that with your MXL770, too? And is the audio from the 802 going into the mic, ACC or the DATA jack? Quoted:
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The biggest thing is to drop the bass out of that radio. Go into the TX filters and drop everything below 200hz, and bring the treble up +2. Otherwise you're just very bass heavy. Is that just for the stock mic, or are you doing that with your MXL770, too? And is the audio from the 802 going into the mic, ACC or the DATA jack? on the mxl770 too. Running audio through the mixer right into the ACC jack. |
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Here's a little story about available bandwidth, recording, radio, and such. Back in the very early days of recording the Victor Talking Machine Company (later Victor Records, RCA-Victor, etc) had a gadget that took over the place of the piano in the parlor, the "Victrola!". Yes, that record player that you cranked up to make it go, played records with bamboo needles, and amplified by a big megaphone type horn coupled to the diaphragm that was vibrated by the needle. Early radio was not much better. In those days a dance orchestra or studio orch was recorded with one microphone hung from the ceiling, suspended inside a hoop with little springs to help isolate from extraneous vibrations. The string bass or tuba were "lost in the mix". Those bass instruments could not be heard either on the radio, nor recordings. For one thing, the mics did not pick up low notes well, and on the other end of the audio chain, the speakers did not do much of a job with the low notes, either. The instuments began to change, favoring those that recorded well. One manufacturer of saxophones came out with their "Radio Improved" line, which had a brighter tone than other similar saxes. The banjo or Dobro were preferred over regular guitar due to the brighter tone. Dobro, I am not talking about guitars played on the lap with a steel bar, leading to the "Hawaiian guitar", but rather the acual guitar itself, which had a big steel resonating disk that gave it a brighter tone. http://upload.wikimedia.org/wikipedia/commons/1/16/Dobro.jpg They gained the name "steel guitar" not because of playing them with a steel bar (or neck of a bottle) but the steel resonating plate. But back to recording and radio... From the early days of recording it was found that the bass saxophone recorded well. You could hear the reedy tone well over radio and Victrola speakers. The listener was not really hearing the actual fundamental tone, but the reedy harmonics. The bass line could be heard clearly. If you look at pictures of the early Benny Goodman Orchestra there was no string bass or tuba, but a bass saxophone. The bass sax was not sitting up front with the other saxes, but back by the drums, piano, and guitar, in the rhythm section. So, in 1916 when you went down and bought your first Victrola the chances are you also picked up some records, many of which used the bass saxophone for the bass line. The tag at the end of the Three Stooges closing theme was bass saxophone, not tuba. Here's an example of that sound. Listen at 00:48. https://www.youtube.com/watch?v=WveEco2Rq2w https://www.youtube.com/watch?v=QabgfbnCJ6Y Later, as recording and playback equipment improved, going into the 1930's and 1940's, they switched back to string bass or tuba. So, there is an example of the early days, using instruments that fit the recording conditions, available bandwidth. damn. pretty cool history post there, Jup. thanks for sharing it. |
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After the last five posts I went back to JT65. Quoted:
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All very interesting, but... So Brundoggie, how are you making out with adjusting your mic and compression levels in your 7100? After the last five posts I went back to JT65. 
yeah, after reading all that i began thinking to myself, "Do I like my CW sidetone at 600 Hz, or 700 Hz? Maybe I should get a little wild & crazy today & practice w/ the sidetone @ 500 Hz tonight."
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