Jim, If only all users followed your advice. Sadly the last contest showed they don't.
I was amazed at the number of stations who happily went on and on with distorted, over driven or clipped audio. One such station called us on every 3 hour segment and the audio was nothing short of a joke. (I am being polite) I have tried several times over the past couple of years and I confess I was not happy with any of the attempts. The K3 M1-M4 does the job for me and this is reflected (I feel) in the low number of repeats I have to make. Conditions are improving on 15 and 10M so I guess I can expect to hear some great audio coming out of W land...:-) 73's Gary On 23 April 2011 02:33, Jim Brown <j...@audiosystemsgroup.com> wrote: > On 4/22/2011 12:40 AM, Ian White GM3SEK wrote: > > Most people can improve articulation dramatically by slowing down only > > 10-20%, so it only requires a modest increase in the tempo setting to > > restore a normal brisk speed. Time compression is a re-sampling > > technique and it does introduce some artefacts, but these are minor > > compared with everything else that happens to a SSB voice signal. > > You're right, Ian. My advice is really directed at users who are not > skilled in audio editing, and is part of a KISS (Keep It Simple, Stupid) > philosophy. I do a few things with editing that I wouldn't dream of > recommending that others try (and I won't even mention them) because > they are so complex and easy to screw up. My experience with time > compression goes back to the Lexicon D224 hardware product, a pro > product that sold for about $7,000 in the early 1980s. I demoed and sold > them to studios for the purpose of shortening radio spots (commercials). > I heard them on a lot of material, all with very expensive voices. 5% > compression sounded great, 10% was OK, but more than that was artificial > sounding. > > Don suggested keeping a copy of the original file. Yes, a good idea, but > all of the editing software mentioned has an undo function, so if you > listen to each step as you go along, you can get away without that. And, > of course, you can always re-record the message, which I do occasionally > because I don't like the first attempt. In fact, I often record a > message a half dozen time (or more) before I start editing it. > > A few other suggestions. > > When recording, make sure your shack is quiet -- close the door, turn > off all the fans and air conditioners. > > Work with the mic not too close to your mouth so that you don't get > breath pops and low end boost, and make sure that the audio levels are > right as shown on the editing software's meter and waveform display. > You should NEVER see any overload, and it's best to keep the peaks of > the waveform at least 3dB below max (0dB on the display). If you do, > throw out that recording and start over. You CANNOT fix it by turning > it down after it's been recorded. > > After you've finished editing, use the EQ function to roll off the low > end at about 100 Hz, and to roll off the high end at about 6 kHz. > > If you like to use VOX (I do), record a click at the beginning of each > CQ to activate the VOX a few milliseconds before the message starts. > This prevents losing the first syllable of the recording. Adjust the > peak level of the click to be 15-20 dB below the peak level of the > message. Use this click only on messages that will transmitted alone, > like your call, a CQ, and the Thanks message at the end of QSO. Do NOT > use it on an exchange -- you should activate the VOX with the live mic > when you say the other guy's call. When all this is working well the > click should not be transmitted. To get the timing and level right, > play the track through the rig and listen to the result. > > Setting levels is VERY important. I like to keep the highest peaks of > the final recording between about -6dB and -3dB as indicated on the > Audacity waveform display. It's also important not to set the output > gain of the computer too high. Most sound cards have greatly increased > distortion when they get close to full output, so it's best to run their > output a bit lower to keep that distortion low. You don't need to > reduce it a lot -- 3-6 dB is enough. > > When setting levels at the K3, remember that you want to match the level > of the live mic going straight into the K3 with the level of the > playback from the computer. We use the Line Input control to set the > level of the playback audio, and to do that, we must temporarily set the > K3 for Line Input and use the front panel Mic Gain. I usually set the > Line In gain so that I get the same indicated ALC and COMP indications > on the K3 meter display with playback as I do with the live mic (about > 10dB of COMP on the hottest voice peaks). > > 73, Jim K9YC > ______________________________________________________________ > Elecraft mailing list > Home: http://mailman.qth.net/mailman/listinfo/elecraft > Help: http://mailman.qth.net/mmfaq.htm > Post: mailto:Elecraft@mailman.qth.net > > This list hosted by: http://www.qsl.net > Please help support this email list: http://www.qsl.net/donate.html > -- VK4FD - Motorhome Mobile Elecraft Equipment K3 #679, KPA-500 #018 Living the dream!!! ______________________________________________________________ Elecraft mailing list Home: http://mailman.qth.net/mailman/listinfo/elecraft Help: http://mailman.qth.net/mmfaq.htm Post: mailto:Elecraft@mailman.qth.net This list hosted by: http://www.qsl.net Please help support this email list: http://www.qsl.net/donate.html
