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SM7B, streaming, UA Solo USB, and the API channel strip
Old 2 days ago
  #1
Here for the gear
SM7B, streaming, UA Solo USB, and the API channel strip

I am an absolute beginner in the audio space. So this is probably why you will see obvious ignorance and a lot of wrong assumptions, and maybe even the wrong choice of gear for the job. With that out of the way...

TL:DR: I am fumbling around with new hardware/software, and cannot seem to find any tutorials on setting up voiceover on the API channel strip in the Unison slot.

I am streaming on Twitch, and have a lot of commentary going on. With that much talking, nobody wants to hear a crappy sounding mic. So I bought an SM7B and the Apollo Solo USB.

I think I bit off more than I could chew, because what I thought was a few tweaks here and there would make the mic sound great. Oh, how I was so wrong. Buying plugins will not magically solve my issues if I have no idea what I am doing. To what was sounding OK, said tweaks made it much much worse.

So where is a good place to start on how to learn this thing? Youtube is making me really lost.

*puts on flame ******ant suit*
Old 2 days ago
  #2
Lives for gear
 
๐ŸŽง 10 years
Welcome to the world of audio! You have to start somewhere. You bought good equipment... no worries!

The most esoteric thing about UAD is their "unison" technology. It is used to imitate famous pieces of equipment.

My advice for you right now: skip it! You don't need to use it to operate your interface, and with the SM7b, you aren't trying to imitate anything. You have the mic you want: an SM7b. It's a very popular mic for podcasting. There is no reason to try to make it sound like a u87 or a c12.

Also, vintage preamps are not intrinsic to what the SM7b is. Broadcasters simply plug an SM7b into a clean, high-powered preamp and hit the record button.

So you have all of this exotic signal path modeling processing technology with the UAD unison technology, but it's irrelevant and distracting to your particular goals. You don't need to use Unison technology to run your UAD hardware/software. So just skip all of that.

What you DO need to focus on is good basic compression and eq. A compressor reduces dynamic range. If a sound gets too loud, the compressor will pull down the volume quickly and automatically for you.

Because a compressor pulls down the loudest sounds, this reduces the maximum volume peak. This allows you to subsequently raise the overall track volume, because those original peaks that would have produced digital overload have been backed down by the compressor. If you don't know what a compressor is, this is the magic you are looking for to get professional results... not "unison circuit modeling technology".

The BEST compressor to get started with is the Waves Renaissance Compressor. You will get great results with it, and you will quickly learn how compression works because the interface is so simple. I know you have invested a lot in the UAD stuff and want to stay in that world. But trust me, the Waves Renaissance is the way to go. It's on sale right now for $29.99.
https://www.waves.com/plugins/renais...lugins-marioso

As far as EQs go, get one where you can click on the screen and drag/draw the eq curve (these are sometimes labeled "paragraphic eqs". I will refer to this style as a draggable eq). You want less bass? Click on the eq graph where the bass frequencies are and drag downward. You not only will get less bass, but you can see your eq graph on the screen... very educational, very useful. This is actually much easier to understand, and faster to get results with, than trying to figure out APIs dual-concentric knobs.

Fabfilter ProQ3 is one of the leaders in draggable eqs. But you can find some free ones. Fabfilter has a 30 day free trial, so you can use it to understand how draggable-eqs work and then go find a free one once you understand what you are working with. ProQ3 is such a good teaching tool that it's worth it to download and practice on. The ProQ3 has a lot of advanced features. But you can skip all of those. Simply click on the horizontal line at the frequency range you are interested in, then drag up/down to boost/cut that frequency range. It's that simple to get started.

Watch some videos on "graphical eqs" (not graphic eq), "parametic eqs", "click and drag eqs" "paragraphic eqs"... that sort of thing. Like these:
https://www.youtube.com/watch?v=Ko-XuaIaYgw
https://www.youtube.com/watch?v=gMlb2oYNQJc

After watching about six of them, you will get the idea: you want a screen where you can shape the eq curve visually with your mouse.

Waves makes a plugin that has the Renaissance Compressor and a draggable eq all-in-one: the Renassaissance channel. That's on sale for $35.99, and will allow you to do compression and draggable eq all in one unit. However, because it is an all-in-one unit, the interface is cluttered, which can be intimidating for a new user. So if it looks too technical, buy the Renassiance Compressor and find a separate equalizer.
https://www.waves.com/plugins/renaissance-channel#image
https://www.youtube.com/watch?v=fTt_coz32JA

The Renaissance Channel also has a noise gate on it. Since you are doing streaming, you might particularly like that feature. You can set it so very quiet sounds are eliminated all-together. That way if you are not speaking and you want complete silence, the noise gate will shut off the sounds when you are not speaking. If you're not intimidated by the busy-looking screen, the Renaissance Channel directly matches your signal processing goals: compressor to smooth the dynamic range, eq to shape the signal a bit, and a noise gate to put silence in between speaking parts.

So ditch the API stuff for now. It's not matched to your current needs. You didn't waste money by not using the API software. The UAD solo bought you high quality preamp/conversion. For you, that's what really matters. You can paint whatever signal processing you want on top of it.

YES:
Waves Renaissance Compressor, draggable eq

NO:
API plugins, unison technology

You ironically and unknowingly walked into the wrong part of the jungle to get started: API, Unison. I can think of few *worse* places to get started than that combination. Get out of there! That's not where you need to be. Follow my advice, and you're understanding, results, and productivity should take off... quickly.

Last edited by gearstudent; 2 days ago at 02:27 PM..
Old 1 day ago
  #3
Here for the gear
Thanks for the tips. It truly, truly helps.

OK, as for the plugs you suggested. I have the Waves Gold pack I bought on sale a few years back. It came with the compressor, but not the channel. So I just purchased the channel as I typed this. As for the EQ, it's funny you mentioned the Fabfilter. I saw it advertised on Facebook, and it looked like something I can use later down the road, so I bought it the other day. I do not understand why it was horrendously expensive. Ah well. The money is spent lol I have not used it yet, but I suppose I have use for it now. I have zero idea how any of the plugs work. I just bought a whole crapton of them. I even bought the LX480 Complete because I saw it on a lot of gear porn when I was a young kid in the early 1990s. Anyway...

As for the EQ, I also purchased the Maag, because my friend was saying the "air" feature was great for vocals. He just did not say what kind. I dropped it in, played with it a bit, looked at the manual, but I was hopelessly lost. Maybe I can understand it later. All I know is, it just did not sound right on the settings folders.

OK, I removed the API strip from the Unison, and will just use the default preamp. I bought the API because that had the gate. Along with 4,000 other settings. I just need the gate which the DBX 386s had, because I am in an untreated space. I have to be here with people walking around, talking, dogs, and just general chaos. Welcome to my crazy life. I was told I could ditch the dbx once I bought the Apollo. I primarily purchased the Apollo because I was tired of the 2i2's latency, and no line inputs. The software was very limited, too. But here's the problem - The Apollo has zero latency, or so it was said. I just cannot figure how to route the DAW to the Apollo and vice versa to get the feature I paid a lot of money for. Maybe it is my DAW? I have Studio One Professional. I thought I could use the Artist, but sadly, it did not support plugins. So, I had to upgrade. I am not certain if it can be used for live applications, and early, early on, I tried Ableton 10.

If you can point me on how to route the thing (Studio One, interface, and OBS streaming I just know it uses ASIO, whatever that is), that would be great. That way, I can use the plugins I purchased, including the Waves and Fabfilter.
Old 1 day ago
  #4
Lives for gear
 
๐ŸŽง 10 years
It's great you're making progress. You won't regret buying the FabFilter EQ. It's a fully professional industry standard. It's super clean. I think you said you just bought the Renaissance Channel? If so, that has gate, compression, and eq all in one package. The interface can be intimidating looking with so much stuff on one screen. So the secret is to focus your eyes on one section of the interface at a time.

So we have two challenges: you need to get comfortable with comp/eq/gate, and you also want to sort out the zero-latency monitoring features of the UAD interface. Here is the strategy: work on them separately. They are completely separate processes.

I would work on getting good at compression/eq/gate first. How do you do that? Simply put some narration audio... any audio from anywhere... into your DAW. It doesn't have to be you on the audio. That's not the point. The point is for you to practice compression/eq/gate.

Here are some quick tips on how to use the Renaissance Compressor:
1. Load some vocal audio onto a track.
2. Play the track back, adjust the fader so the levels are reasonable (maybe -6db or so on peaks). Put the track on loop so you can have it continuously play.
3. Look for the ratio control. Set it to 2:1
4. Look for the threshold control. Slowly pull that slider down about a "centimeter". Watch to see the gain reduction meter light "bounce" about a centimeter. The size of the light rectangle that is bouncing corresponds to how much gain reduction you are doing. Listen to the audio while turning the compressor on and off. You should see/hear some nice gain reduction without destroying the audio.

higher ratio settings will compress more aggressively. a 10:1 ratio means "for every input gain of 10 decibels, only allow 1 decibel of signal level increase". a lower setting such as 2:1 or 4:1 is usually good for vocals, unless you are going for something very aggressive.

the threshold control tells the compressor when to start compression. lower thresholds (pulling the slider down further) mean the compressor will start kicking in sooner.

so let's say you have the threshold set to -10db, and the ratio set to 2:1. it translates to this: "hey compressor: for any signal strength stronger than -10db, i want you to kick in and start compressing. for those strong signals, i want you to reduce any signal stronger than -10db by a two-to-one ratio. so if you give me a signal that is -6db, that's 4db higher than -10db. so i want you to cut that difference in half since it is a two-to-one ratio. so only allow 2db more (i.e. -8db on output) rather than 4db more (-6db on output)."

let's look at another example: threshold set to -14db, ratio set to 4:1 (much more aggressive compression). "hey compressor: any signal strength stronger than -14db, i want you to activate. signals levels below -14db i want you to ignore and not compress. so if you give me a -2db signal, that's 12db higher than the -14db threshold. we're using a 4:1 ratio, so we divide by 4. 12/4 = 3. that means only allow the signal to be 3db higher than -14db. so the peak level on that sound transient is going to be -11db, not the original -2db."

so the compressor still allows a signal above the threshold to be louder, but the amount it is allowed to be louder is controlled by the ratio. this allows you to contain the dynamic range while still keeping things natural sounding. it would be strange sounding for the dynamics of a voice to abruptly hit a wall above a threshold. so the compressor reduces level above the threshold fractionally so it's not so destructive and obvious.

very high ratios are known as limiters... thinks like 1000:1 ratio is "brickwall limiting". those ratios are used for safety purposes to avoid digital overs, and for some other reasons. for your basic work, you want to stay in normal ratios: 1.5:1, 2:1, 4:1, 10:1. anything more than that is extreme and should be avoided. 2:1 and 4:1 are good basic ratios.

regarding the maag eq: that's a knobs eq... stay away from those for now. it's simpler than the api, but it's not what you should be focusing on (i have the maag, the api, fabfilter... trust me).

since you have the fabfilter, try using it. you can use the renaissance channel eq, but the fabfilter is great so try using it. you will see a horizontal "blank" line. on the screen. that is your canvas waiting for you to paint your eq onto. left-rigth on the screen corresponds to low-high frequencies. click on the middle of the line. a circle will appear. that is a draggable eq point. move the circle up and down. it will boost/cut the frequencies in that area. slide it left-right. you make it boost lower (left) or higher (right) frequencies. look at the readout on the screen. fabfilter will tell you what frequency the eq point is centered on.

click on the line in another spot. now you have a second draggable eq point. in fact do that again. so now you have three draggable eq points. spread them out across the screen. now you can control the low-mid-highs of your audio by raising/lowering any or all of the three eq points. make big, broad moves while the audio is playing, so you can hear what the eq is doing.

there are other things you will need to learn. the "q" tells fabfilter how wide or narrow to make the eq point be. a high value of "q" will make it very narrow. a low q value will make broad, less aggressive slopes.

there is bell and shelf shapes. bell creates a bell-shape to the eq. the eq curve will be at a peak at the draggable centerpoint, and fall off proportionally on both sides. bells are often used on midrange centerpoints.

shelfs are often used on the extreme lows and extreme highs... the farthest left and farthest right eq points on your graph. for example, you might want to roll off all frequencies below 60hz on your voice. so you drag the far-left eq point to 60hz and set that eq point to shelf. now, that curve will drop down to zero. or in the high frequencies, you might want some "air". so you set the far-right draggable eq point to 10khz, set its filter type to shelf, and then drag it up a bit above the center line. now you have a rising shape from 10khz onward.

fabfilter also has really cool things like "dynamic eq". but you can skip that for now.

so with each eq point you ask yourself some fundamental questions:
1. where do i want an eq point?
2. how much do i want to boost or cut?
3. how wide/narrow do i want the eq point to be?
4. do i want a bell or a shelf shape for the eq point?

you can do very similar things in the renaissance channel's eq as well. definitely work with "click and drag" eqs for now. skip all of the stuff that is trying to imitate old-fashioned analog hardware with knobs.

a gate works like a compressor in that it has a threshold. but this time the threshold tells the processor "silence all signals below this level". a quick way to set a gate is as follows:
1. play back your audio. look at where the audio meter is frequently reaching
2. pull the threshold down until it is near that area.
3. pull it down further (overdo it). you will now hear unwanted chatter. the gate will be blocking sounds you want to pass through, so some of the good audio is getting cut off.
4. slowly raise the threshold up until the good audio is not getting cut off.

there is also a "hold" feature on the gate. this can prevent the gate from opening and closing over and over when it is on the borderline of the threshold. the "hold" feature will "hold" the gate open (i.e. allow signal to pass through) for a miniumum amount of time, regardless of signal strength. so if you set the hold to 100ms, you are telling the gate this: "hey gate: every time you open up, stay open for at least 100ms. even if the signal drops quieter, stay open anyway. i don't want you closing down the audio too quickly. stay open for at least 100ms".

for your podcast audio, you want to not accidentally chop away the good audio. so set the gate's threshold very mildly. that way, it will chop away low decibel background noise when you are not speaking, but will not chop away the good audio when you are speaking.

real-time monitoring is something very specific to UAD. that involves (cue drumroll...) reading the manual and watching videos.

trying to learn that AND eq/comp/gate all at once is too much. get comfortable with comp/eq/gate first. the audio doesn't have to be you, and it doesn't have to be recorded live while you are practicing. simply get some spoken word audio from somewhere, and practice on it. first get comfortable with compression. then eq, then gate. then put them all together. it won't take long if you focus on learning each one separately.

once you understand the comp/eq/gate controls and how to get the results you want, then solve the live tracking-UAD hardware challenge.

right now, you're problem space is too big: you are trying to learn compression/eq/gate/live tracking/direct hardware monitoring all at once. you need to break this down into solvable pieces.

no one part of this is too challenging. but you're trying to learn five things at once, which is four things too many. take these things one at a time. gain practice/fluency/understanding. then move onto the next one.

whichever audio you practice on, make sure it is not already processed with eq/comp too heavily. if you have to read a newspaper article into the computer to get some sort of raw audio, that's fine. you just need something to practice on.
Old 23 hours ago | Show parent
  #5
Here for the gear
Your info seems very helpful. I will try that, and update when I get the hang of it.

Edit: The Solo is having some severe stability issues. I am not sure if it is the drivers, messing with settings, ASIO, etc., but the crashes and freezes are so constant, it is unusable. I sent a support ticket to UA, and see what they say. They are closed weekends, but looks like I will need to go back to the Focusrite for now.

Edit 2: I cannot monitor real time with the plugins you suggested. It seems I have to record the raw audio first, then drop the plugins afterward then tweak it. I am not sure if there's a way without latency, but I seriously doubt it. That's one of the biggest downsides I am seeing.
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