Audio-Setup with Mixer - Rookie-Questions ;-)

[oe3ide]

Hi dear Anan community,

atm I am running an FT-2000 with an external audio-rack, all "controlled" via Yamaha MG06-Mixer.
Now I am planning to go for the Anan 7000 DLE MKII.... and the investigation has just started

Ok, the goal for me is, to control most of audio-levels via the mixer. At least mic-gain, headphone-gain, line-out from pc.
I have drawn 3 variants of audio-chains. See attached pictures.

So my questions are:
1) which chain will be good, in terms of low latency
2) would this chains "work"
3) About VAC1 & audio processing via openHPSDR (pre-eq, cfc, leveler, etc)... I have read, that audio-processing is also possible, when using VAC for mic-in/audio-out. But in the transmit-settings, there is only "line-in" and "mic-in" available. So I guess, that if no line-in/mic-in is present/connected and VAC is configured, openHPSDR takes the audio from VAC and processes it. Is this correct?
But what if there is also a VAC2 configured (ie. for digi-modes)... how to prevent the processing from VAC2?
4) The anan doesn't have a line-out. Which connection should be used, for bring the rx-audio back to the mixer? Headphones or Speaker L/R?

For me, the version without USB (using physical connection to mic-in, and physical connection back for rx-audio) looks good, but I have also read, that using the connections on the Anan, produces higher latency...


many thanks!

73 de Ernst
OE3IDE

[w-u-2-o]

Ernst,

Your drawings are wonderful! However, my recommendation would be "None of the above!" Instead I'd recommend the following approach:

1. Use a high quality audio interface with ASIO driver support. I find the Behringer UMC202HD to have the best drivers. Steinberg is good. I don't care for the Focusrite, the driver stability doesn't seem very good, although perhaps it has been improved since I last tried it. Right now I'm using a Presonus, but it is very expensive and complex.

2. Attach the output of your audio rack to the input of the interface. Attach the output of the interface to your speakers and headphones.

3. Dispense with the external mixer. Use Voicemeeter Potato. Put the interface on hardware input 1 and hardware output 1. Put PowerSDR (or Thetis) VAC1 on virtual channel 1. Put VAC2 on virtual channel 2. Put your digi mode software on virtual channel 3. Now you can mix and route audio to your heart's content all in software (it's very low latency), including mixing out some low latency sidetone.

4. Consider dispensing with your audio rack and using the built-in features of PowerSDR: leveler, rotator, pre-EQ, CFC compressor and post-EQ. If you use Thetis you get a fully adjustable noise gate/expander and look-ahead VOX, too.

73!

Scott

[oe3ide]

Hi Scott,

thank you very much for your explanations and suggestions. Below my thoughts:

1. Use a high quality audio interface with ASIO driver support. I find the Behringer UMC202HD to have the best drivers. Steinberg is good. I don't care for the Focusrite, the driver stability doesn't seem very good, although perhaps it has been improved since I last tried it. Right now I'm using a Presonus, but it is very expensive and complex.

Thanks for the hint, I would need the UMC204HD, because I need the 48v phantom power for the Beyerdynamic Mic. I have read about the poor ASIO implementation of Focusrite, but I will give it a try (I have this device already)...at least, its worth a try. But the Presonus AudioBox iTwo looks good

2. Attach the output of your audio rack to the input of the interface. Attach the output of the interface to your speakers and headphones.

Ok, in my case only the headphones, I don't use speakers at all

3. Dispense with the external mixer. Use Voicemeeter Potato. Put the interface on hardware input 1 and hardware output 1. Put PowerSDR (or Thetis) VAC1 on virtual channel 1. Put VAC2 on virtual channel 2. Put your digi mode software on virtual channel 3. Now you can mix and route audio to your heart's content all in software (it's very low latency), including mixing out some low latency sidetone.

The main reason for the external mixer is the quick and easy control. i.e. Watching a youtube-video (audio goes to headphone), on the side I am watching the waterfall (currently SDRConsole & RSP1), if a station of interest appears, I can quickly close the audio from PC (in this case from the browser) and make the qso and avoid transmitting youtube-audio
And all this without going into software, deactivating, adjusting audio.. I run 2 lcd-monitors, and both a filled nearly 90% ..no more space for additional panels... It is more a comfort-"thing" then a technical need.
Doing the mapping using Voicemeeter Potato is definitly a great idea for the audio-routing inside the pc.
I guess the Behringer Xenyx Q802 USB uses a similar USB-interface then the UMC2020HD, that would bring back the mixer But if I look at my current setup (Yamaha MG06) the only knob I really use is the audio coming from the pc (see example "youtube" above)... I have to rethinking about that need. And if I am not wrong, listening to tx-audio from openHPSDR/Thetis could be a complicated job, because via headphone-out on a mixer, I would hear also the audio coming in from the mic....

4. Consider dispensing with your audio rack and using the built-in features of PowerSDR: leveler, rotator, pre-EQ, CFC compressor and post-EQ. If you use Thetis you get a fully adjustable noise gate/expander and look-ahead VOX, too.

Yes, thats the goal... I know the built-in audio-processing is second to none, and I want use that for sure.

And finally "....audio-processing is also possible, when using VAC for mic-in/audio-out. But in the transmit-settings, there is only "line-in" and "mic-in" available. So I guess, that if no line-in/mic-in is present/connected and VAC is configured, openHPSDR takes the audio from VAC and processes it. Is this correct?...But what if there is also a VAC2 configured (ie. for digi-modes)... how to prevent the audio-processing from VAC2?"
I am right?

Thanks & stay safe
73s de Ernst
OE3IDE

[w-u-2-o]

And finally "....audio-processing is also possible, when using VAC for mic-in/audio-out.

That's correct.

But in the transmit-settings, there is only "line-in" and "mic-in" available. So I guess, that if no line-in/mic-in is present/connected and VAC is configured, openHPSDR takes the audio from VAC and processes it. Is this correct?

There can be things connected to line-in and mic-in, and regardless of the setting in the Transmit setup tab, if VAC1 is enabled and none of the VAC "override" settings are enabled, all transmit audio will be sourced from the VAC1 interface. The override options are there for those who want to do something like use a voice keyer during contests but override it with a line-in or mic-in source.

Receive audio will be available on all interfaces and, if VAC1 is enable, VAC1 as well.

But what if there is also a VAC2 configured (ie. for digi-modes)... how to prevent the audio-processing from VAC2?"

VAC2 only carries receive audio from RX2. It has no transmit side functionality. VAC1 carries transmit audio and receive audio only from RX1. The design could, perhaps, be better thought out or more flexible than that, but that's currently how it works.

[oe3ide]

Ok, it's going clearer and clearer.
But this "VAC2 only carries receive audio from RX2. It has no transmit side functionality. VAC1 carries transmit audio and receive audio only from RX1. The design could, perhaps, be better thought out or more flexible than that, but that's currently how it works." confused me a bit.
How can digi-mode work, without transmit functionality?

73s Ernst

[w-u-2-o]

Digi mode app's must be connected via VAC1 to work properly.

It's a long story on how the audio functionality evolved the way it did, longer than I want to type here Perhaps in the future it will become more flexible and robust. In the meantime this is why I suggest Voicemeeter Potato as a way to work around many of the routing and mixing problems associated with the existing interfaces.

[oe3ide]

Ok. Thats an important info
You know, I am just on "dry run". If the Anan is here on the desk, I don't want to start with all the research...
I use digi-modes not very often, maybe once a week.

So, if I understand right, for working SSB and digi-modes there should be several solutions:

• use mic-in/line-in for phone, VAC1 for digi-modes => not good for low latency on phone
• use VAC1 for phone and digi-modes:
  • switching VAC1 sources manually in openHPSDR/Thetis settings
  • switching VAC1 sources via voicemeeter, like you suggest

Last question (I hope )
VAC2 cannot be used for digi-modes. So if VAC1 set to digi-in/out (via voicemeeter or manually), how does powerHPSDR/Thetis know, to NOT process the tx-audio going to the digi-mode-app?

To obtain (and monitoring) a flat frequency curve, I normally use Audacity to record the transmitted audio (currently in my case I receive my tx-audio from audio-hardware-rack + FT-2000 via RSP1 and feed the audio to Audacity). If recording is complete, I use "show plot" to check the curve and then doing adjustments, mostly without listening the recording
My "audio-elmer" Peter, HB9UQX (Anan user and audio-guy) has a very interesting article on his qrz.com page. So for recording and checking the frequency curve, I always use his sentence please orange, I insist that you remove the brown zipper from the blue microphone boom. This sounds funny, but this sentence contains most of our typical vocals and consonants and makes the result reproducable.
Be aware: XYL may ask, if it is all ok with you
Atm I am experimenting with "Spectrum Lab", to get the average-curve in real-time, this works perfect. So no more recording/playback is needed. Adjusting eq and see the resulting curve-plot real-time, makes the job easier

[oe3ide]

Ok ,found the answer:

all audio processing (except VAC RX Gain, VAC TX Gain and ALC) is disabled on VAC audio when in DIGU or DIGL.

[LarryW]

Due to the fact that somehow I ruined my front mic input on my radio and now am attempting to use VAC as my main route.

All I am wanting to do is use my USB headset and control both the radio audio and a online SDR through VoiceMeeter Banana.
I have got some of it working but after a bit it all seems to run together in my brain due to too many selections.

Is there anyone that could mimic the following and let me know how to set it up properly. I have looked at YouTube videos but just have not come across the one that fits our application.

I think most of my problem is not totally understanding the proper routing of things

Rig
PC
Voicemeeter Banana
USB mic with headset

Goals are to listen to both the radio and a online sdr at the same time and control both volumes on Banana.

Thank you in advance for your kindness!
Larry
N7ZDR

[oe3ide]

Hi,

I would suggest:
1) make Voicemeter Banana (or Potato) as your default sound device in windows (I use VAIO 3)
2) select your physical mic-in device as "hardware input 1" in VM
3) "hardware out A" should then be again your physical soundcard

=> so you have all your audio coming from windows (incl. web sdr) routed to your headphones

Cause atm I dont haven an anan, I guess Scott can give you the infos for Thetis/openHPDSDR.

73s
Ernst