EQing DAX Transmit Audio

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A year ago there was a post verifying that DAX transmit audio bypasses the onboard EQ in SSDR. Is that still the case with v1.5, and will it continue to be in v1.6?
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Charles - K5UA

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Posted 2 years ago

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KY6LA - Howard, Elmer

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YES
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Ken - NM9P, Elmer

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I don't know about 1.6. But in 1.5, when in DigiU or DigiL the DAX transmit audio is unprocessed, thus preserving digital modes. But in SSB, AM and FM the DAX audio comes in before the filtering, TX EQ and processor and gets treated just like it were a Mike input, allowing a recorded Mike audio file to be processed just as your Mike would be, with the same characteristics.
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Charles - K5UA

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Thanks Ken.  Getting ready to try XLR to USB mic converter from Heil so that I can use software audio processing to get audio from the computer through DAX  to the Flex.  Wanted to know if I would still have the Flex internal EQ capabilities if I needed it.

Would you happen to know if the DAX audio goes through the Flex transmit ALC circuitry?  I think I am having some distortion from the ALC attempting to handle my high dynamic range voice characteristics.  Would you mind if I send you a recording a friend made of my audio?  We did a test Friday with a friend on my 6500.  He sounds great on it, I take the mic and sound bad.  I do the same thing to another friend's 6700.  He sounds great, I sound distorted.  We tried the same test on another radio, we both sound great.  I am suspicious that the Flex ALC is having difficulty with my low power voice, and I hear the output power going all over the map. Can't think of any other reason when the only variable in the test is the operator's voice.  Any thoughts?

By the way, I see an icon to upload video and photos, is there a way to upload an MP3 audio file to a post so community members could hear what my audio sounds with my voice as compared to another voice, all things being equal except the voices?
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Ken - NM9P, Elmer

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Charles,
I haven't found a way to post mp3 audio only.  

But, if you have Windows movie maker, you could use it to import your mp3 and put a single simple picture background on it and turn it into a simple move and upload it that way.

It would be even better if you can get the free openbroadcast.com (or it might be .org) software and do a screen capture of your SSDR screen.  Use the FDX trick that I posted and listen to yourself while transmitting into the Transverter port.  You can turn the transverter port power down until you only register about S-9 on your own 6000 receiver.  Then you can send the DAX receive channel from your receiving slice to the OpenBroadcast.  That way someone can look at all of your controls while you talk and also look at your meters and panadapter pattern.  Zoom in so that you have your voice channel filling about 1/3 to 1/2 the screen and you can really see a lot that is going on.

I have used the USB output of my Behringer Q802 Mixer into my computer and selecting 'PC' as a mike input.  In v.1.5 it gets processed in the same way that a hard-wired mike does.  I don't know about 1.6 yet.  I assume it is the same.  But yo might want to check your audio properties in the Windows sound/recording/playback section to make sure that you don't have any processing, noise cancelling, EQ, or other routines that the computer is doing that you don't expect.  My experience is that when I select PC and feed my Heil PR-22 through the USB/computer combo that it sounds good, but not quite as good as the direct feed from my mixer into the BAL input on the 6500.  There seems to be a little bit of conversion loss/distortion going on.  You also want to check the USB drivers properties and make sure that the USB line is set for a sampling rate of 48 KHz instead of the default 44.1 KHz.  The 6000 series uses an even 48 KHz sampling rate and a mis-match requires a conversion to be done which may add to your problems.

The main issue, I suspect, is that your particular voice may be driving the rig into excess.  Double check your mike level meter with the EQ IN and the PROC OFF and make sure it never gets into the red, even on your strongest voice peaks.  (When you're RED, you're dead!)

in fact, if your voice is doing it when others are not, you might even want drop it back a notch or two below that.  When you get that right, then you can turn your processor back on, but don't adjust the mike level.  Sometimes a little less is more!  If you try to squeeze out the very last bit of audio power, you can sometimes cross the line and it starts getting distorted quickly.  Do all of these adjustments without your amp in the line.  Then adjust your power output to the amp with the power level, not your mike levels.  (I know you know this, but other noobies might be listening in.)  

Also, if you are using PROC in DX or DX+ mode, then you must raise your low frequency cut-off up to about 150-200 Hz or higher.  Lower frequencies tend to generate processing artifacts that add to a distorted sound.  On my "extreme DX/Contesting profile I am going from 300-2700 Hz in DX+ mode and it sounds pretty good.  Watch your 1K EQ level....Too high and your voice can get "honky," sounding like the cheap PA Horns at the county fair.  Too low and you lose power.  If you want to add clarity, boost the 2K & 4 K a bit, but not too much of you will get a "lisp."  Too much 500 makes it muddy, and too much 250 makes it boomy.  too much 125 & 63 makes the low frequencies drone on irritatingly. Too much 63, 125 & 250 also add those processing artifacts when in DX and especially DX+ mode.

I hope this helps.  If you can get some recordings, I would be glad to give a listen as time allows.  I wouldn't make any major changes other than those simple mike adjustments until 1.6 comes out anyway.  It should be very soon!

Ken - NM9P
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Charles - K5UA

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Yes, waiting for V1.6 later today. 

I have the feeling that I am going to have a hard time driving the amp if I cut the audio back any more.  I hardly hit yellow as it is.  But, I'll try it.

Also pulled my DEQ2496 digital audio processor out the closet to see what I can do with it acting as a limiter.

Just crazy that I'm the only voice that distorts the audio.

Will record a few friends with Adobe Audition, then adjust all the voices, including mine, to the same peak level.  Then put all the voices back to back in a single wave file to play back thru the Flex, and record the results. Should be interesting.

Now, you mentioned the 48kHz sampling rate of the Flex. That should not make any difference when the Flex is fed an analog signal, but I'm wondering what happens when a 44.1 Khz sampled wave file or mp3 audio file is used through DAX into the TX audio? Hmmmm.... more to worry about.
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Ken - NM9P, Elmer

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The sampling rate is no real biggie.  Just go to the record/playback settings for your USB Audio device in the windows sound driver and click "Properties>Advanced" and there should be a way to set the sampling rate.  Set it to "48,000 16 bit stereo,"  which is also "DVD quality."  Once set, it is good to go, until Windows 10 loads another update and breaks it.  Then you just install and set it up again.  Setting the DAX and other audio sound channel sampling rate and other properties has now become part of my regular configuration routine every time Windows breaks things.  I can get it all done pretty quickly now due to sheer repetition!
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Ken - NM9P, Elmer

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BTW...Charles, what mike are you using with the Heil USB audio converter?  Could it be that you are actually over exciting the mike diaphragm itself?  I have seen a case where a person's mike technique was actually distorting the audio right at the mike.  There is not electrical fix for that!  

(My wife sometimes speaks so loudly into her cell phone that it drives it either into distortion, or overdrives the sampler so that her signal actually drops out.  Some guys will do this on FM, they overdeviate their rigs and their signal, when weak, actually deviates out of the receive filter and drops the squelch.  They don't believe me when I say "talk softer, I can't hear you" or "don't talk so loud, you're breaking up."  but FM is different from their old AM CB when speaking louder was seen as adding to talk power.  When I can finally convince them to soften up a little, their signal stays within the receiver filter passband and I can hear them just fine.  Then I tell them to have their rig adjusted so they don't overdeviate their FM signal.
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Charles - K5UA

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Thanks for the tip on resetting the 40Khz sampling rate.

My mic is a dynamic headset by Audio-Technica called a Pro 8HEx.

I have a very soft voice and can hardly drive any radio I've ever owned, so I doubt that I'm over-exciting the diaphragm.

Will probably won't be able to install V1.6 until Friday (really need to consider retirement).  Will be using an outboard limiter to make sure I don't over-drive the input circuit, not even into the yellow, much less the red.  Will try to hook up with you Friday or Saturday to listen to it.