Optimal Gain Settings for your Microphone Preamp
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- čas přidán 22. 08. 2024
- If you're looking to improve your audio recordings, then you need to know about microphone gain settings. In this video, we'll show you how to set your microphone gain to achieve the optimal level for your audio recordings.
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By setting your microphone gain correctly, you'll be able to compensate for any audio quality issues that may be present in your recordings. By learning how to set your microphone gain, you'll be able to produce recordings that are of the highest quality possible!
In this video I explain how to set your preamp gain, and provide clear audio examples that demonstrate the noise floor, and why certain gain settings will sound better than others. This is great content for anybody with a home recording studio, or who wants to learn about microphones, audio recording, preamps, and how to get the best possible recordings of vocals, guitars, and any music. Good preamp settings also help with mixing and mastering.
A masterclass that even a beginner could grasp. Great style and info, thanks!
Finally a video that makes preamp gain comprehensible, thank you. I can set and forget the gain now without having to worry about clipping in loud parts and about the noise floor on silent parts of my recordings.
hi my friend..i'm from indonesia..i don't understand english..i've watched this video..can you help me to explain what this video means..?? because I just started home recording with the Audio interface, and I'm confused about the Volume gain setting on the audio interface..I want to get a loud sound input but low noise
Hello, obviously you do understand the written language a bit. Why not just activate the subtitles and translate what remains unclear?
Great video. Useful info well explained too. There are actually at least four major sources of noise: ADC, preamp, spurious/EMF and mic noise.
Yes, but of those 4 sources of noise, the preamp noise is usually the greatest. Thanks for the comment :)
Finally a video on this topic I can understand!! Thank YOU!
Oh, that was great, thank you. The audience can understand snr very well and completely.❤🎧✅
Beautiful. Thank you man.
RE: bits available..there is a plug in to measure that..but Im never at 16B so I see! Good points on that. Thanks
Great info! But i thought you were going to give me a recording level in dB .. like record your vocal levels at - 10dBFS or -12dBFS or 6dBFS to be above the noise floor to get a clean recording and below full scale not to clip. Yes my session setup is in 24bit. How about sample rate .. which one is better? 44.1 or 48KHz? Thank you for your helpful tips.
There’s several strategies for watching the meters, however, I find these to be more time consuming, and less reliable than simply looking at the recorded waveform, but if those strategies work for you, there’s nothing wrong with it, as long as you don’t clip.
As for sample rate, higher number has better sound quality but uses more computer processing power and hard drive space.
Most people, and even audio recording courses, suggest to use the input meter and record at a specific db level. They often say your loudest peaks should hit -18, or -12, or -8 or whatever number they think is right. There is no value that is agreed upon because that method doesn't take into account average volume, and also because many people don't understand the signal to noise ratio that I demonstrate in this video.
As for 44.1 vs 48Khz, 48 is slightly better, but it puts slightly more strain on the computer. If you would like to know more about this, I have a video explaining sample rate here:
czcams.com/video/RbqyMiHncXQ/video.html
It's been a year for you, everyone says go 48k or 44.1k. Go 96k instead of spending 3 years wondering why your audio sounds like it has a blanket on it.
@@Hello-pl2qethe only issue with 44.1 will be if you use saturation or analog emulation effects that will need oversampling. 96kHz is useful for mixing and mastering engineers, but not for recording.
This was pretty simple, for a beginner. Thanks for the video!
I am wondering, do you have a more advanced tutorial? Where is it best to set gain for a chain? Mic preamp -> compress -> audio interface. They all have input and output levels, where is best to boost the volume a bit to avoid added noise? Does it depend on the quality of gear? Let's ignore that and assume all of the preamps are the same, do the functions (preamp vs compress vs audio interface) have any inherit benefits of boosting earlier or later in the chain? Or do you just boost them all the same until you get a usable audio level?
Thank you so much! I was just sitting in front of my preamps and realized that I have not actually a concept on what level I should aim for... this video gave me a clear understanding and answered my question very clearly!
One question remains: Can I conclude from this, that adding Gain in the the DAW (of course without clipping) has no negative effect on the signal?
I landed here because I switched from a GoPro to an Ace Pro camera to use for my motorcycle recordings. Using the exact same microphone (Movo) wired to both devices. The resulting waveform in the GoPro is great and I can see motorcycle noise and then my talking easily stands out. With Ace Pro, it's all small and looks alike. Can't figure out how to fix it.
Get.Beautiful...
Great video and concise and clear explanations!
Thank you!
Hi, I use an SM7B with a Cloudlifter and Go XLR. How quiet should the noise floor level be when recording voice over? My room/ambient noise hovers around -50 dB, and my normal speaking voice is around -20 to -18 dB, while the peak is around -16 to -8 dB. I want to eliminate as much noise as possible while maintaining the quality of my voice. I can clean it up in post, but I can still hear some lingering noise when I'm speaking. Any help be greatly appreciated!
Ya, the sm7b is notorious for that. I don’t know why it’s so popular, but in my opinion, it’s not a very good microphone.
Literally any other dynamic or condenser microphone will solve this for you. I recommend the iSK Vibrato, or the 512 limelight.
Excellent presentation. Thank you for this!
the portion about 16 vs 24 i never knew that. i do know that when saving as 24bit the hdd space doubles but after your explanation i can understand
Ya, 16 bit is ancient history. It was useful in the 90’s when 24 bit was too much for those computers, but computers have come a long ways since then, and even a really weak modern computer can handle 24 bit with ease.
Sir my mic records low sound vocal i can't hear it properly why is that plzz rply me is the problem of my mic or mobile
Finally best advice so far.
Thanks:)
Im an absolute beginner.. so does a Fethead gain of up to 27db count too? if i give like 30db of gain in my Wave XLR do i add the 27db and say i just gave 57?
I’m trying to understand what causes a pinched audio also what exactly pinched means ?
But like what do I set my windows sound recording level setting. No one talks about it. (Bottom right speaker, right click. Recording. Right click mic/mixer then properties, recording level)
Bruh max
@chasescordo7995 doing that makes all my mics push far into the red. But on the mixer they aren't even close. I'd have to set my mixer lights to like -30 to just be -6 on pc
What program do I need to see the waveform? I only have levels on my Mackie and Behringer Mixer. No waveform on either of those
Have you covered how Windows or Mac mic volume also interacts with your mic gain/volume in a video? I'm interested in trying to boost my mic enough to get rid of the elements in my noise-floor as much as possible but I wandered how Windows mic volume interacts with the quality or potential distortion of the mic / elevating the noise-floor if that's something worth looking into! I personally have a PreSonus Revelator io24 and a RODE Procaster mic but I find my mic is very quiet and compression settings have been a real difficult thing to try and work out as an amateur - I'm still not sure what some of the settings even mean in the various compressors included in the Revelator io24's preamps but this video definitely gave me some insight into realising that pushing my raw output from the mic as close to 0db as I can without having another preamp like maybe an sE Dynamite DM1 in the signal chain is not helping things when it comes to trying to get my compression settings correct!
Thank you for this
You’re welcome!
Allow me to introduce a new unit of measurement for sound volume,
which I call vovol* (abbreviated: volume voltage).
THIS IS HOW IT WORKS: You use two pre-stages, one with positive volume values, and the other negative.
For example, so has the pre-step with positive values
a measurement from +0 decibels up to +20, while the one with negative values, has from -0 decibels down to -20 decibels.
The highest voltage occurs when the value is
+20 and -20 decibels (or: 20 vovol),
while there is low voltage, when the value is on
+0 and -0 decibels (or: 0 vovol).
You can possibly also combine two different values with each other, by adjusting the value to +10 -20 vovol, which gives a crisper effect. Have experimented with this myself at work and at home.
The adjustment can of course be set to taste, but the purpose of vovol for me is to equalize the sound volume, so that you better hear weak sounds and at the same time avoid high deafening sound levels (loudness war's).
Thank you for reading this!
Take care of your hearing...
:-)
Awesome video anyway!
I have never heard of the vovol. How would it relate to non-voltage sound waves, such as acoustic or digital?
@@Get.Beautiful.Recordings
Well, this is how it works:
When the sound has high volume voltage, the speaker membrane is stretched like a guitar string, so that the sound becomes more sharper, realistic and noise-free.
This can be achieved by using an equalizer with a preamp. First, all frequencies are lowered, to, for example, -12 decibels.
Then you raise it up to +12 decibels, with a preamp.
This is how I usually experiment, with vovol, because currently there are no tools, or apps, to do this.
If you, or any of you reading this, have an equalizer with a preamp, I think you should try it too.
Thanks for asking!
And thank you all for reading!
:-)
Again, great video!
So, if I'm correct, it seems like you need a preamp in order to optimize your signal to noise ratio. Or can you do this just by setting the gain on an interface?
Do this by setting the gain on the interface, which is the preamp gain.
Awesome video !
The only prob for me is my GR mic pre..going to watch your vid lesson #23
this is really useful
Helpful.
great, informative video!
Helpful approach 👍
Thank you :)
Great explanation! Was wondering, the Interface has a Gain knob and then Windows in it's configuration you can also set a Volume Level. How do you know whats the right level to set in Windows?
The gain knob on the interface is the correct one to use for setting the recording volume. The volume in windows is for setting the playback volume.
@@Get.Beautiful.Recordings thank you so much. I'm referring to the volume knob in the microphone for recording. In the interface your setup a gain value and on Windows in the mic settings you also have a volume/gain recording setting.
Great video! Question- how do you see the ratio in the track header?
I have know added a preamp I'm trying to find a good medium between louder and quieter while using it as I'm a quitter person in general
Excellent video. Thank you.
Working with some older equipment (Roland VS-880EX), and trying to negotiate the SNR considering its built-in preamps, which I know nothing about their settings or quality. Any knowledge of these old machines?
Yes, I have some experience. Not with those specific pieces of gear, but with those brands from that era, and the preamps are very noisy, around the same level as the hairball Lola, or presonus audiobox.
@@Get.Beautiful.Recordings Thank you for the reply. Yeah, I find myself using the Bertom Denoiser when I bring the tracks back over on my PC with each song. Well, thank you again.
Great video
Thanks!
Could you please help me a bit? I am using: micro Audio Technica AT4033 + preamp Warm Audio WA73-EQ + audio interface Focusrite 2i2. The preamp has 2 button: mic volumn and output volumn. I don't know how to setup these 2 buttons properly. If I set mic volume and output volume at 12 o'clock. And if I stand 30 centimeters away from the microphone, almost nothing can be heard (The signal light on the audio interface is not lit). If I increase the max volume and output volume the microphone sensitivity increases but the sound is very bad. Please help me some idea.
P/S: I am using Studio One, and I didn't adjust volumn in it. Just kept default value.
Hi can you please share the device and software which you use
GRATE!
Muito grato por mais esse Ótimo vídeo
CONGRATULATIONS
Cheers. :)
thank you!!
Your welcome:)
Thank you ❤
the bigger wave length recording requires at least 80 % gain on an audio interface (ex. 2i2 ) and AT 2050 mic
What is the difference between the input source Of Line level and MIC level
actually im using a ZOOM F8n and a Sennheiser G4 and it's always clipping when they are shouting when im on mic levels
Hi Emery Alexis Tomboy,
Clipping is obviously bad, but the solution is simple, you need to turn the preamp gain down to a level which is not clipping during the loudest parts. If the gain is already very low, and cannot be turned down more then you can turn down the output of the G4 receiver.
If turning the gain down makes the quiet parts, or normal volume parts too quiet, then I would suggest using a compressor, which will raise the volume of the quiet parts and also reduce the volume of the loud parts, when used correctly.
Thanks for the question
when you say turn up my preamp, are you speaking of the volume knob of the channel of the interface i'm using? I'm using an audient ID44
Yes, it’s labelled “Gain”. There’s 1 gain knob for each channel, and this is what I’m referring to
What are you recording through (signal chain)?
Great info, but you really need to get more speakers in the background dude 😂
I’m using a Sudotack ST-800 Microphone and the microphones clips and distorts the audio by itself on a certain dB. Adjusting the gain on the computer doesn’t change anything. Distance mattered but I had to be like 5 feet away from the mic to actually record louder audio.
Is there a way i can adjust the gain using software?
That’s unusual to need that much distance, or else you get distortion. It sounds like the issue is with the microphone. Unfortunately, there’s no way for software to fix this.
If you are using a compressor with this level, is the threshold set pretty low?
If this level is lower than what you are used to, then yes, you would need to set the compressor threshold a bit lower.
My gain is set to max and it’s too quiet/ low… wtf
That can happen with a very low sensitivity microphone. You can get an in line amplifier such as a cloudlifter, or an external high gain preamp capable of like 75 db of gain.
What preamp were you using during this test?
My main studio preamp is the cranborne audio Camden.
Thanks for the question:)
Cool
Thanks :)
@@Get.Beautiful.Recordings btw as long as I'm recording and hitting-18 . Someone taught me that while the volume knob is at 0. The called it u unity gain?
Hi, and thanks for the question:)
Forget the unity gain concept, it’s not relevant for setting your preamp gain. The -18 strategy is good, but I find it awkward trying to establish what the input meter is reading because it bounces around so much, and doesn’t give an accurate perspective of the average.
This is the most unscientific and amateurish way of setting gain. This is absolutely NOT how an audio engineer should set gain.
1. On many DAWS, there is a global setting that magnify the waveforms within the arrange window. Using your visual waveform concept, the magnify function would lead the engineer to believe he has good gain staging when in fact the level is simply too low.
2. Just because the waveform doesn’t clip, it doesn’t mean the levels are correct. The absolute numbers (and therefore meter readings) are important! A signal can be too ‘hot’ without clipping. Equally a signal can be too ‘cold’ without appearing to have a small waveform.
Why is this important? Well, most analogue processing (and thus consequently the digital emulations) worked their magic most effectively when the signals they were receiving were around 0dBVU. This is the case with a lot of plugins also, because they are emulations of the original circuitry. Providing less signal will increase the noise at each stage of the signal chain. Providing too hot a signal will introduce clipping not just at the preamp conversion, but potentially at each stage of the signal chain too. PROPER GAIN STAGING REALLY MATTERS! Reading meter levels really matters.
Additionally, trying to eyeball the waveform instead of reading the meters properly will be a surefire way to run out of headroom quickly once you start mixing more than a couple of sources. You will end up having to reduce your levels to avoid overloading your master bus.
Guys, you will never ever hear a professional advise you to just eyeball a waveform and hope for the best. There’s a reason equipment and DAWS have meters… it’s because it actually matters.
I run a pro studio with clients on a regular basis. This is how I set my gain. It’s fast, simple, and has never given me trouble.
@@Get.Beautiful.Recordings while that may be true, it is more likely that the high degree of headroom and low noise floor implicit within the 24bit DAW is helping to cover poor gain staging. Proper gain staging is a very necessary habit to get into, especially if you hope to ever work in the analogue world where signal levels really do matter. Furthermore, just because you’ve done it one way without any perceived negatives, doesn’t make it optimal. Optimal is to load the circuitry with the correct voltage levels throughout the chain, weather analogue or digital, in the sure knowledge that these good habits are immediately transferable to any system, no matter how high the noise floor.
@@pete3816Hi Pete, I want to use a leveler plugin to maintain 0 dBVU. Should I set the preamp knob on the interface to hit around 0 dBVU while tracking? Also, I'm using POWAIR (a plugin by sound radix) to handle dynamic changes and keep me in the 0 dBVU range. Your thoughts?