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Quality Microphone Audio
David Holmgren
Member ✭✭
I think most people are lured into a Flex based upon how great they sound on the air. I was in that camp so when I couldn't get mine to sound good I did some digging through the Flex forums. I learned that it's best to use as little DSP inside the radio as possible and to rather process your audio before it hits the Flex mic jack. With that in mind I stopped by Guitar Center to pick up a tube mic pre-amp and I came home with the PreSonus TubePre V2 $129.00. I then connected that with a "y" cable to an old RANE ME-15 10 band stereo EQ from my former DJ days. I use one channel of the EQ for the Flex and the other channel for my VHF / UHF radio. My goal was to share my ElectroVoice EV320 with both radios. Everything is working great and finally my Flex sounds like a Flex. While on the air another Flex owner I was talking to suggested to change the low cut from 0 to 100 and that took away the echoy, bassey, hollow sound to my audio. My amp was also happier to not have to work extra hard amplifying sound we can't hear. So my advice is to skip the internal EQ keeping it disabled and go outboard with your sound contouring because it works much better. I know the Flex veterans know this already but the newbies like me should find this useful.
Best 73,
Ruffers K9RUF
Best 73,
Ruffers K9RUF
1
Comments
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As a long time Flex owner, I totally disagree. The Flex with internal EQ can be made to sound great. You just have to adjust a bit for your voice.. No need for external EQ.8
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None of the other Flex users with great audio have such a system as yours. How do you explain that they have great audio? I 100% disagree with your approach.1
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Good sound can be realized using out board gear, but many of the best audio I have heard is using the Flex EQ,,,mine is a good example...2
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Well glad all of your internal EQ's work OK. I just couldn't get mine to sound good.
2 -
David, I can understand that. What some people do not realize is that the EQ has a relationship with the PROC. If you set the EQ in such a way that is drives the PROC hard then artifacts will show up.
After many hours of testing I learned that by setting my sliders below the mid way yields the best sound.
With this type of EQ when you want to add more base you lower the higher Freq.. and if you want more highs,,you lower the low freq sliders. So you subtract not add....
Here is my settings now,,notice that I am well below mid way on most my settings. As a result my sound is well balanced and rich. And I can now run the PROC on DX with great audio with 0 distortion.3 -
To outboard process,
or NOT to outboard process,
that is the queston.
It is often a mater of personal preference.
One thing I have discovered is that with a decent mic, and the internal EQ, Filtering, PROC, and DEXP, you CAN get the rig to sound very good.
You can also, by misadjusting the gain, EQ, and PROC, make the flex sound really BAD!
Most of the errors I see that make it sound bad include....
1) running WAY too much BASS in the EQ, especially when running PROC on DX or DX+.
Even if you keep the mic gain out of the red, you can overload the processor's input and generate digital artifacts and distortion in the signal. WHen running PROC on DX or DX+, I recommend setting TX Filter Lo Cut at 150 Hz or higher.
2) Unless you are into extremely high fidelity ESSB , there is usually no need to run TX Lo Cut below 50-100 Hz. And then, GO EASY on the Low Frequency EQ. I often see signals where the component below 150 Hz is a full 20-30 dB above everything else. This is **** the amp, **** the other station's ears, and in most cases is wasting power transmitting information that is not heard by most rigs. "Super-Bass" is extremely difficult to copy on weak signals or in a pileup or noisy band. (My Rag Chew Profile goes from 65-2900, but I have others for DX, weak signals, and Contests that run anywhere from 150-350 on the Low Cut.)
3) In the same way as the "Super-Bass" error, WHen you give your signal some needed boost to the highs, be careful not to run the 1K, 2K & 4K too high. Too much 1K makes you sound harsh and "****." Too much 4K makes you lisp and whistle. Too much 2K can override your vowel sounds.
Properly adjusted, you can even see a nice balanced frequency pattern on your panadapter as you talk. If you need more articulation, ad a little more 1-4 K as needed. Too much of these can also overpower the processor, and you might need to drop Mic Gain a tad.
For a Rag Chew or Hi fidelity profile, you won't need to run more than NORM on the PROC.
For DXing, Weak Signals and Contesting, you may want to use DX or DX+. But adjust your low end filter and EQ to match. You don't want to overcompress a lot of low end stuff.
THere are a number of audiophiles who swear by expensive external processing equipment. Properly adjusted, they can make your rig sound really nice.
However, I think that properly adjusting the internal stuff can make is sound almost as good, except for those that add a touch of reverb, or do fancy phase-shifting of the waveform, multi-band compression, etc. But that all can get pretty expensive!
My 6500 and Heil PR-22 get consistent good reports - both rag chewing and DXing.
Ken - NM9P
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Ruffers, One clue you gave in your post is that it was only AFTER hooking up all of your outboard processing gear that you changed your Low-Cut TX filter from 0 to 100 Hz. That may have been one contributing factor to your difficulty in getting things adjusted to your liking.
In any case, if you like the sound you are getting with your outboard gear, Great! That's ham radio. It isn't about acheiving some elisive standard of perfection, or even pleasing everyone else, it is about getting the rig to sound the way YOU want it to sound.
There are many ways to get there.
Enjoy your rig!
Ken - NM9P1 -
Thanks Ken and Bill for your comments.0
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For more information, check out my audio tutorials on Youtube.com/nm9p
They are a slight bit dated regarding the names for the Mic Profiles, but they might help you understand the need for different profiles for different purposes.0 -
Using Ken’s method with full duplex is a good way to hear exactly what you sound like. To go a step further I use a DAX channel on the receiving slice to feed Audacity record. That way I can speak, record a clip, then play it back to hear what it sounds like. Trying to monitor your own audio real time is difficult. Dave wo2x1
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I disagree with your smile eq approach.1
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I constantly get unsolicited positive comments about the quality of my audio. This has been true on each of the 3 Flex's I have owned. I've used high end condenser microphones such as the Rode NT-1A, various Heil microphones, and inexpensive headsets like the Koss SB45. All can be made to sound great using the internal equalizer and related adjustments. The profile feature of the Flex allows saving of microphone profiles to suit various purposes/modes, which makes it easy to configure the Flex as needed. I also like to keep my station simple, minimizing cables/connections, and equipment.
Larry, W1IZZ
2 -
I have done this both ways ; on my flex 3000 I didn't care for the EQ curves that Flex had set, so I used either an out board setup like a 2496 - I wasn't happy with that either so I setup using a DAW and dumped that back into the 3000 using VB cables keeping as much as I could in the digital realm and it worked well. Esp with Darrin's modded PSDR which opened the 3000 bandwidths.
When I got the 6400 I again used the DAW but soon found that the EQ was much better then older models and now only use it with a PR781 (I did find that the pre-set EQing didn't come close for my voice and and thus started fresh) (Most everyone's voice requires a slightly to more pronounced adjustment). The EQ is a little touchy and you need time and a good ear to set it but once you do you really don't need the other items.
I also when first started in the wider bandwidths and ESSB used a TubeMic pre-amp similar to one you mentioned and found the circuit they used along with the cheap- really non functioning tube in it as there inst enough voltage applied to be NOISY on the output and it created more garbage then did any good.
Even when I get in with he ESSB groups I only run the internal EQ and just open the bandwidth, many were impressed with just running the mic jacked into the back of the radio and no outboard gear. Less chance for RFI getting introduced into the audio chain with the external equipment, less noise products being dumped from each piece of equipment into the next then only being amplified more and onto the next piece, which was the main reason I has switched to using a DAW setup and the various plug-ins (VST files).
Again I now only use the internal EQ and am very happy with the results and get many unsolicited comments on audio.0 -
Ken - your Youtube video should be required viewing! I was sold after how good you got the Flex hand mic to sound. No small feat!
Seriously Dave - give it a try. It's FB if you use your setup, but Ken's video shows how to get great sound with only the mic and SSDR's EQ.0 -
Thanks. That was with the FHM-1.
I wish it was a FHM-2. I am sure I could really get that one to shine!
Perhaps I will borrow one some day so I can do an update on adjusting the Flex Hand Mic.
I have been tempted to replace the dynamic cartridge in my FHM-1 with an electret cartridge and a resistor and a couple of caps and convert it to a pseudo FHM-2.
But since I use the PR-22 all the time, I would rather spend my time doing other things.
My next Mic project will probably be bodging an old military aviation noise-cancelling boom mic onto my Sony MDR7506 Headphones to create a contest headset.
I did this years ago with a simple electret element & stiff wire on another headset and had a very respectable contest mic. But I haven't done any of these experiments in years, since I got the PR22 and broadcast boom.
Ken - NM9P0 -
I don't disagree with anything. As both an engineer and a user, I can tell you there is more than one way to get great results. Far more than one way. Many people use the internal processing, and many use the external processing. And guess what; They both work great.
Try to keep an open mind. It's not "Your way, or the highway"...
Cheers,
Neal7 -
Ken, if you choose to try that mic conversion, ebay has very inexpensive electret mic cartridges, typically a 10 pack for less than $5. A lot of people use those to fix up old mics.
ref:
https://www.ebay.com/sch/i.html?_&_nkw=electret+microphone+capsule&_sacat=0
Cheers,
Neal0 -
re: "...like to keep my station simple"
Well, that is fine, BUT;
I actually prefer to make mine complicated, looking more like the flight deck of the space shuttle. It is, after-all, a hobby, and my Man-Cave :-)
I am even starting to think about adding a fourth display monitor, and maybe the GeoChron 4K :-) :-)
Life is short, have a ton of fun!
Cheers,
Neal2 -
I agree with the onboard EQ, it took me a bit to figure it out, but if you play around with it, you can have beautiful audio. I use a PR20 Heil, I have a DX and Normal profile for that, as well as my Heil Pro Set with the 4 and 5 elements, I have 4 profiles for those and all sound great.
0 -
I already have a bunch of them. Many of the older, circuit board mount style I got at Radio Shack about 20 years ago. And I got a bag of miniature ones that have wire pigtails a couple of years ago to fix my son's xBox headset that the dog chewed apart! The miniature ones sound about as good as the larger ones!
https://www.amazon.com/gp/product/B00FGV867M/ref=oh_aui_detailpage_o05_s00?ie=UTF8&psc=1
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David (Wo2X):
Regarding: "To go a step further I use a DAX channel on the receiving slice to feed Audacity record. That way I can speak, record a clip, then play it back to hear what it sounds like. "
Where do I learn more about that? Does Audacity have a specific input for Flex DAX, or is it simply that DAX and SmartCAT define a TCP/IP address and PORT, and then Audacity can record an audio stream from a selected TCP/IP # and Port?
Cheers,
Neal0 -
Ah, Never Mind, found it:
https://community.flexradio.com/flexradio/topics/recording-saving-flex-6700-receive-audio-using-audacity
0 -
Hi Neal Make sure DAX 1is selected in the RX slice then in Audacity go to Edit/Preferences and under Devices make sure recording device is DAX Audio RX1. Then select your computer speakers as playback device. Dave wo2x0
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I like a rack of mysterious looking black boxes, along with the requisite large monitors.0
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I thought I would try Audacity as well. I have everything on the Audacity side setup with the DAX RX 1 as my input and my computer speakers selected. I have Dax RX 1 enabled on my Slice flag and I have Dax TX on.... everything seems to be streaming on the Dax panel fine in RX... but when I go to transmit there is nothing streaming... and Audacity is not recording while in Transmit mode. It will record while I am in Receive mode... I have even tried the Mic Stream and still no luck...
If I go to full duplex mode and Transmit using this mode and 2 slices on the same frequency I can make it work.... but That is not the way I am understanding this to work??
Any Ideas?0 -
Did you enable full duplex? Click the FDX in bottom left.
Dave wo2x
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You must have a receive antenna different from the tx antenna for FDX to work. And in order to listen to your own Transmit frequency, you cannot be using much power at all or you will overload your receiver. Thus I Transmit on XVTR and listen on ANT1 or RXA. (Or B if you have a rig with 2 SCUs.). Make sure FDX is turned on. Transmit on one slice (A) and receive on the other slice (B). Make sure the DAX channel you are piping over to Audacity is activated on the receive slice (B). And is turned on in the Dax Control Panel. In Audacity, select as the input channel for recording the sound device for the DAX Channel for the receive slice (for example, DAX Audio RX1) When you hit record on audacity, you should see deflections on the Audacity recording level meter and be able to adjust the recording level in Audacity.0
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What you are doing with this technique is recording the audio that is being heard by your second receiver slice (B) It will record your transmitted signal the way it is heard by the second slice receiver, along with any filtering , RX EQ, AGC time constants, etc...as though you were listening to yourself in a different receiver.0
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Ok, Great! For some reason I did not realize that I would want to still do the Full Duplex... I have used that method many of times... but I see... this will make it so I can record the signal and replay it without trying to listen to it live (which can be difficult). Thanks for the clarification... I have it working great using the full duplex mode and tranverter with low power! Thanks!
0 -
To each his own. Yes for sure onboard EQ can easily be made to sound great, but since when do all hams take the easy way. Why do so many of us overkill our station, overkill our designs and installations. Because we can is the simple answer. I personally have a 31-band Ashly 3rd Octave Graphic EQ and it was fun for a short while to tweak to my voice but then lost interest and bypassed it. So if someone wants to use external EQ to get the sound he wants, great. It's a journey and having 31 frequencies to tweak can surely get you there. Total waste of time using any outboard EQ 8-band or less for the obvious reason that the SmartSDR has 8 bulit-in the software and with no cables and RF or ground loop issues.
1
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