Voicemeeter Buffer Settings for Reducing Delay (i.e., Latency)

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Using Voicemeeter Pro in this demo: www.vb-audio.com/Voicemeeter/...
This shows how to adjust the buffer settings to reduce latency, or the delay between making a sound and hearing it back through your headsets. In Voicemeeter, the setting of the device assigned to "A1" sets the latency for all the rest of the system outputs, including ASIO. The lower you can safely go, the lower the delay in hearing it back will be across all devices.
It covers the differences between the MME, KS and WDM settings, and how to test between them to find a stable lowest setting -- especially important if you need to mix different kinds of devices.
If you are getting intermittent latency issues, try disabling your WiFi and LAN network cards in the network devices control panel by right-clicking on them and choosing "Disable". Those drivers often are responsible for audio glitches and intermittent increased latency, and disabling them often solves the problem. Re-enable them when you need to use the Internet, of course!

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Опубликовано:

20 июл 2024

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Комментарии : 174

@HalfDayHero 3 года назад

I forgot to say thank you for taking the time to make this tutorial. I also wanted to add that I use this software to support my Hyper-X Quadcast microphone, which, as a microphone undoubtedly sounds great straight out of the box, however it does ever so slightly lack a bit of bass that's needed to really give it that "premium studio sound", thankfully this software perfectly resolves that. So, if you have a Hyper-X Quadcast and you're wondering if using it inline with this software is worth it...the answer is yes (especially considering the Hyper-X Quadcast does not come with any companion software!)

@syLLyTime 5 лет назад

Thanks man, finally solved my delay issues! Great guide, well spoken.

@kaitokei 2 года назад

It's 2022 and your video is still helping many people, including me, solve my problem. thank you so much!!

@CreatorsMediaTools 2 года назад

Awesome! 😀

@MrRyan782 2 года назад

This is exactly what I needed, and gets straight to the point of the settings you discussed. Thank you!

@CreatorsMediaTools 2 года назад

Excellent news! :-)

@vbaudiosoftware 9 лет назад

Very good introduction to latency management in Voicemeeter. Basically i recommend to stay above 256 samples (with WDM and KS device) and 512 with MME. Below 256 samples latency, stream stability is less garanteed. Note that the total latency in your video is also given by the MIC input device (which was MME) and the latency of the VB-Cable (for the loop back). Basically, with 256 sample buffer size, we consider we can have 30 ms latency between MIC and Speaker (output A1) or MIC and virtual outputs.

@arkplot2172 4 года назад

the data you have provided are extremely wrong because they only work on speakers on OBS recording you hear an annoying echo and also the voice is annoying and deformed ..

@LeonimuZ 4 года назад

@@arkplot2172 You're telling the guy who made the program that he's wrong.

@ihatesignupsgrrrrrrr 3 года назад

I think I have a solution to auto-correct latency as software solution without the need of a human to attempt to tune their inaccurate audio sensory default equipment(ears).

@GuillermoMathias 3 года назад

That "KS" device saved my live! Thank you so much

@zatoichiable 8 лет назад

Could you do a video on how we can use and set up midi controllers like korg nanokontrol?

@MrCreativeoutlet 4 года назад

Thanks so much for posting. Saved me a lot of time.

@ibrahimelgeaidy6141 5 лет назад

Thanks a ton!! Saved me hours of trial and error

@HalfDayHero 3 года назад

Been looking for this software for freaking years.

@twistrgaming 4 года назад

This is such a fantastic demonstration video :)

@CreatorsMediaTools 4 года назад

Thank you very much!

@no_categories 3 года назад

This was extremely helpful. Thank you!

@CreatorsMediaTools 3 года назад

So glad!

@Karlculate 4 года назад

It's 2020 but this still helped, THANKS!

@CreatorsMediaTools 4 года назад

Awesome! :-)

@CreatorsMediaTools 4 года назад

@@HeidyUmali Made this video a few years ago, but it still applies today.

@vlogit7159 8 лет назад

Thank you, your video helped me alot

@gabbocxliv Год назад

Thank you so much! Clear and concise!

@CreatorsMediaTools Год назад

Glad it was helpful!

@blueknight_live 4 года назад

THANK YOU SOOO MUCH!!!!!!!!!!!!!!!!!! THIS HELPED OUT SUBSTANTIALLY!!!

@CreatorsMediaTools 4 года назад

Fantastic!

@CharieVanWits 7 лет назад

This has helped, but that tiny delay drives me insane when I'm streaming

@CreatorsMediaTools 7 лет назад

Most streaming software has a delay compensation adjustment for the audio - try adjusting in 100ms increments at a time. Glad this helped a little. Another additional solution is to grab a cheap ASIO interface so you can use its ASIO buffers instead, but that doesn't help entirely.

@bambampewpew32 3 года назад

same Charissa lol

@xxxcozz 7 лет назад

awesome video, thanks

@shauntitude 2 года назад

thank you soo much ur vid was the only tutorial that worked

@CreatorsMediaTools 9 лет назад

*Voicemeeter Buffer Settings for Reducing Delay (i.e., Latency)* This shows how to adjust the buffer settings to reduce latency, or the delay between making a sound and hearing it back through your headsets. In Voicemeeter, the setting of the device assigned to "A1" sets the latency for all the rest of the system outputs, including ASIO. The lower you can safely go, the lower the delay in hearing it back will be across all devices. It covers the differences between the MME, KS and WDM settings, and how to test between them to find a stable lowest setting -- especially important if you need to mix different kinds of devices.

@natireson4140 5 лет назад

thank you for the demonstration

@CreatorsMediaTools 5 лет назад

You're welcome! If you ever have any questions or problems that come up, please do not hesitate to ask. I usually respond the same or the next day. :-)

@sog78 3 года назад

Thanks! Help a lot!

@CameoKillz 7 месяцев назад

I use an audio interface and FL Studio with my microphone. I try to match the bitrate in FL Studio to voicemeeter attempting to see if they can go lower but they only seem to go as low as 768 for the bitrate. I cannot for the love of me fix this. I have Voicemeeter Potato, FL Studio 21, and a Focusrite Scarlett Solo.

@CreatorsMediaTools 7 месяцев назад

Have you added the Focusrite to the A1 dropdown as an ASIO device (appears at the bottom of the list)? Once you do that, Voicemeeter will automatically adopt the same settings as the Focusrite. Those higher numbers are from the Windows drivers.

@SeyyideSultan 4 года назад

Hello! I am trying to use Voiceemeter Banana to mix microphone and music to teach dance classes online, with all studios closing where I was teaching in person. I have learned a lot about audio these days because of the need, and was able to feed the AUX input of voicemeeter into my Zoom meetings. The problem is the delay... the microphone comes well after the music in the mix, so I cannot help my students counting the beat in synchrony with the music... The voice also comes visibly behind he video. If you see me in a closeup talking, you realize that the voice comes later than I said it. Any help? I tried going down the sampling but it's not enough... Using a camcorder as video, with a NIX capture card. Then using a RODE GoWireless plugged to the mic port in the laptop and Windows Media Player to provide the music. I tried mixing the music two ways, both work, both with a delay: 1) making Voicemeeter input my default sound system, 2) feeding the Windows Media Player through Virtual Audio Cable. It seems that's the mic and the plug to the microphone, that I cannot get around. I also tried with the built in mic (same thing) and plugged a lavalier there (same issue). The mic in comes at 48 kHz, and Voicemeeter does not let me change that (all my outputs are 44.1 kHz). I would greatly appreciate it if you have suggestions. Love the ability to work out the volumes so predictably with Voicemeeter but the latency is so bad. By the way, before using Voicemeeter I was trying sharing the music from computer (on Zoom) and that way also there was a delay evident in the Zoom stream. I thought when I was able to mix the lines with this Voicemeeter I would solve it so I'm really sad that after all that effort I could not improve the issue (I could only improve the volume management). THANK YOU!

@CreatorsMediaTools 4 года назад

This video ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-dMu3C3cpVLg.html demonstrates how to take advantage of some Zoom settings to achieve higher quality audio, but using Windows audio (such as your laptop's microphone input) is the stumbling block in achieving lower latency/less delay. I recommend buying an inexpensive ASIO unit, such as the Focusrite 2i2 2nd gen. Windows audio is inherently laggy, whereas ASIO devices can work with considerably less lag, even via Voicemeeter. There also is a VST plugin for a DAW that permits near-instantaneous audio transmission, but I'm not sure you have such software. A free solution might be Cantabile Lite as a VST host - see tutorials elsewhere on my site describing that. cantibilesoftware.com The plugin is "Listen To" from AudioMovers. audiomovers.com/

@CarlKidd 8 лет назад

Will/can your buffer setting affect your audio output when doing a hangout on air? When I'm participating in a HOA in the filmstrip I find my audio is not in sink with my video both when I'm in the live event as well as the recorded RU-vid video. I found your how to set up voicemeeter to send out system sounds in an HOA a life saver but the latency issue is driving me crazy. Thanks in advance for you help/suggestions.

@CreatorsMediaTools 8 лет назад

Yes, the lower you can get the buffer without it crackling, the better the sync with the video will be. Though 512 works with many situations, I find 256 to be the most tolerable for me. The lower the buffer, the lower the latency, so the faster it delivers the audio to the application. But, lower buffer settings also increase the strain on the CPU, so one must find a good balance that allows the other apps to work well. The author of the Cantabile live performance software released a free ebook recently called "Glitch Free" on optimizing your computer for audio and video - you can get it here: www.cantabilesoftware.com/glitchfree/

@CarlKidd 8 лет назад

+Terry Leigh Britton Thanks for the quick response Terry. I'll try lower my buffer settings and see how it goes.

@johnnyxthemacnifient5333 Месяц назад

hi, nice explained, but i have an other issue with lighthost for mic modulation, and i keep getting mic audio direct to my main out A1, thats where my soundcard on pc is with FLEXASIO, even when i disable on the mic strip A1 and A2 (whits were me headset with mic is) audio still sounding on A1 so how cani prevent this, of some setting i overlooktd ?

@CreatorsMediaTools Месяц назад

I cannot help but suspect you are not connected to the ASIO correctly. First, be certain that Voicemeeter is the default sound device for both input (Recording) and output (Playback). Then, click on A1 and go to the very bottom of the list of devices where the ASIO devices are shown. Select your ASIO - FLEXASIO card as the A1 output device. Now the input settings in Menu/System Settings will be hooked to the FLEXASIO card's inputs and outputs. Also, what you hope to accomplish may only be do-able using Voicemeeter Banana or Potato, where the outputs can be separated for the A out buttons.

@ney9207 3 года назад

This video explains everything very well. I still got a different problem, i would like to hear my voice constantly because i have closed headphones and i cant hear my voice properly. When i use WDM Speaker Realtek(R) Audio there is no delay at all, everything like it should be. The problem is that when i use WDM Speaker Realtek(R) as my A1 everything else, like brower sound, ingame sound, everything which is using this speaker to deliver sound to me doesnt work anymore. I would like to use WDM Speaker Realtek(R), because the delay is non-existent but i still wanna keep the sound of all the other applications. Would u have a solution for this problem?

@CreatorsMediaTools 3 года назад

In the middle strip, is the A button lit up? Did you make Voicemeeter the default sound output device? (Required for PC audio to generally go through Voicemeeter.) Which version of Voicemeeter are you using?

@ney9207 3 года назад

@@CreatorsMediaTools Thank you, setting it to default solved the problem with ease.

@-champmii 2 года назад

Thank you very much uncle😅

@ronrandellprivado984 3 года назад

Hello...what im having a problem with the 2 outputs that become out of sync...i mean, they perfecrly sync for a total amount of time ( random, sometimes 30 sec, 20,18, or even 1 min) then they become out of sync for a total amount of time ( also random). How do i solve this? Btw i have put the most default settinga i can, i just change the OUT A1 delay to 200ms, after that they will sync but then get out of sync like i said earlier.

@CreatorsMediaTools 3 года назад

Double check that everything (inputs, outputs, any ASIO devices, etc.) are all set to the same sample rate. That usually helps a lot.

@pasupathi8459 6 лет назад

You saved my day

@jakesnussbuster3565 8 месяцев назад

thank you for this.

@caematos 6 лет назад

im getting delay even with my raw mic plugged in. how did u do to get raw HARDWARE INPUT 1

@CreatorsMediaTools 6 лет назад

Delay is a product of the buffers that hold the audio data for processing before sending it out again. It cannot be eliminated, but can be lessened by shortening the buffer size (somewhat risky, though). I, however, no longer attempt to monitor the microphone in my headphones at all. Even a slight delay drives me bats! Just turn off the A button on the microphone is the best solution.

@Naethunn 8 лет назад

Hey, I have a Corsair VOID Wireless and i use voice meter. My voice is all set and ready for recording and streaming etc but everytime i stop talking, there's a HUGE static what i can hear. Any way of removing it?

@CreatorsMediaTools 8 лет назад

I don't know what is causing the static, but there is a noise gate built-in above the input slider that might help. Turn it up till it ducks out to silence when you stop talking. But, if that static is coming from your wireless, then a gate will not get rid of it, as it is signal input coming in. Let me know what you discover! :-)

@Naethunn 8 лет назад

Hey again, the noise gate doesn't work, still static. You may be right about the wireless. I'll just have to buy a microphone -.-

@weenytoosmol5082 5 лет назад

So in my case, I hear my voice come back to me in my headset. It's actually not very off which is good, but I wish I could turn the in ear monitoring of my voice down because its incredibly distracting. It's kind of like how you had yours doubling your voice, but mine's like that all the time and there's only one channel

@CreatorsMediaTools 5 лет назад

I turn off the "A" button leading to my headphones when actually recording or streaming, only turning it on for a sound check when adjusting the orange square above the mic for sound quality. After that I just watch the meters. That slight delay drives me bats! :-) Use Voicemeeter Banana so you have individual control over each of the monitoring outputs (it has A1, A2 and A3 buttons).

@weenytoosmol5082 5 лет назад

This fix worked for me. I finally got it up and running today, so excited!

@baladassweatharanyan6682 5 месяцев назад

Heyyy It helps meee but the thing is after few secs automatically the sounds didn't match how to solveeeee😢

@CreatorsMediaTools 5 месяцев назад

Not sure. Maybe setting lower than your processor can handle? Lower buffers equal more demand on your CPU to keep up with realtime audio.

@mytho2630 4 года назад

I noticed that my audio stutters. I think it is only affecting my voice. I tried messing around with the buffer and it didn't seem to have any effect. It is usually fine when I start but after the program has been on for a while I hear the stuttering again. Strangely this is only an issue in audacity, not in obs. Any ideas why this may be happening?

@CreatorsMediaTools 4 года назад

I think it might be that Audacity has a setting for the project sample rate - that setting needs to match your Voicemeeter and Windows Audio control panel setting. So, usually either 44100 or 48000. The setting is at the bottom left in Audacity.

@mytho2630 4 года назад

@@CreatorsMediaTools Thank You!

@Owayaya 2 года назад

I dont know. I happen to have a sound delay on my headphones in basically every application. I used Voicemeeter to seperate my Spotify sound from my Desktop Sound for OBS. But outside from streaming I dont need voicemeter and it happens to give me a siginifcant delay while running... idk how to fix it, it really baffles me

@CreatorsMediaTools 2 года назад

You cannot get rid of the sound delay when trying to monitor your mic over headphones when using Windows Audio. That is only able to be accomplished using an audio interface having ASIO drivers. What I do is I only monitor my voice enough to get the sound right, then turn off monitoring entirely of the microphone and just talk and watch the meters. That works just fine - there is no need to hear your voice over the headphones when you can hear yourself with much less latency through your own skull! :-)

@MrKTDStudio 4 года назад

Hello, now i've a problem with my Streaming PC Audio is Delay. Can i use this method to fix it??

@CreatorsMediaTools 4 года назад

Well, if the delay problem is monitoring your microphone through your headphones, *just don't do it* except for initial setup. You can hear your voice is working just fine right through the bones of your skull! :-) Voicemeeter is Windows Audio using the WDM and KS (Kernel Streaming) systems. So, it does not yet use WASAPI, which is almost as low latency as ASIO now. Consequently, you'll always have a LITTLE bit of latency, but it won't usually show as lip-sync being out of whack. If using OBS, you can delay the video just a bit to match the latency of the audio to match it up exactly if you notice it is a problem.

@jollyaxekiller7972 2 года назад

So i have a question, when you talk with microphone in voice meter it opens up right to let sound through and then slowly fades away if you don't talk is there a way to quicken this decay in voice meter? There are times when I talk then stop and 2seconds later breathe in and the mic picks it up cause voice meter hadn't actually closed the audio all the way yet. So it could just be my bad mic but is there a way to make it cut out faster after you are done talking?

@CreatorsMediaTools 2 года назад

I always recommend keeping those dials at very low settings - like .8 or so. For a more versatile gate, consider using a VST plugin gate via Cantabile Lite (see videos on using effects hosted by Cantabile Lite elsewhere here on this channel).

@jollyaxekiller7972 2 года назад

@@CreatorsMediaTools ok cool thanks for the reply

@MickyBane 3 года назад

I have problems when I record remotely (source connect or session link) ADR or dubbing to my Adobe Audtion, always records a crackling sound that makes repeat and repeat. This is annoying for the actor. Is there any solution for that?

@CreatorsMediaTools 3 года назад

Crackling usually indicates a sample-rate mismatch somewhere in the chain. Make certain ALL of your sample rates are set to the same rate (44.1 or 48 typically), including the ones in Windows Sound Settings, any ASIO devices and the apps you are using (Audition). Still, with all of that correct, sometimes you will need to shut down the app's audio engine and restart it again a few times to get the sync to "catch". Once sync is attained, it usually stays consistent after that.

@MickyBane 3 года назад

@@CreatorsMediaTools thank you very much for your answer.

@pkmplayer 7 лет назад

Hello! I have been using voicemeter with my A1 output as Realtek HiDef audio (set to 512). Every now and then it causes the audio to become distorted (not changing settings like you did) for example when i open multiple programs at once it causes that distortion. Is there a fix to this?

@CreatorsMediaTools 7 лет назад

If this is happening after about an hour of use, it may be the well-documented "WASAPI Bug" that affects that otherwise speedy and low-latency audio layer (re-starting the audio engine in Voicemeeter will usually buy you another hour). You can eliminate it entirely by selecting MME or KS drivers. I've noticed this appears more often on Realtek cards I own than on other audio interfaces, so it likely is also a driver problem that needs fixing, but the problem is more on the operating system level fundamentally. If latency, or delay from hearing an input on your headphones, is not an issue, then MME would be the way to go until they sort this all out.

@pkmplayer 7 лет назад

Terry Leigh Britton would there be an alternate solution if latency is an issue?

@CreatorsMediaTools 7 лет назад

I use an ASIO interface (specifically the Focusrite 2i2 at first, but also have upgraded to using the MOTU Ultralite mk3 hybrid and 828 mk3 hybrid models. For those, however, their WASAPI interfaces are terrible, and I interface to them both using Voicemeeter instead.)

@jesusacristo307 5 лет назад

can you change the buffer setting for the virtual inputs?

@CreatorsMediaTools 5 лет назад

The one buffer setting serves all.

@Kitarachi 6 лет назад

Hi, I have tried everything I can from this tutorial including trying each type through each possible output with each different latency setting, on normal and swift and I am still hearing a crackling sound when I use the program Voice Meter Banana. I have also searched online on various different forms to no luck. This issue is new so I am confused as to what could be causing it. Also my Pc is a good one so it should be able to keep up either way. Any suggestions would be appreciated as I am near wits end...

@LiveMusicalPerformances 6 лет назад

Try altering the order you start up programs. For instance, I have to start Cantabile 3 (my performance hosting software) first before starting Voicemeeter in order to avoid crackling. Give that a try and let me know how it works - and let me know which software you are using with Voicemeeter so I can perhaps test it out here as well. :-)

@Kitarachi 6 лет назад

Thanks for the quick reply, I will give it a whirl. For equipment I'm using; Focusrite Scarlet 2i2 with my RØDE NT1-A. For programs/software; Voice Meter Banana. VB-Audio Virtual Cable Control Panel & VB-Audio Virtual AUX Cable Control Panel. (Both set to Bandicam (To capture everything including record sound) The Scarlet 2i2 is supported by "Red Plugin Suite & USB 2.0 Audio driver" Everything is set to 16 bit, 48000 Hz. My PC specs are: Windows 10 Pro 64bit OS Intel Core i7-6700K Quad-Core Processor @ 4.0GHz CPU Ausu Z170-A ATX LGA1151 Motherboard ASUS Nvidia GeForce GTX 1070 STRIX 8GB GDDR5 Graphics card Corsair Vengeance 32GB DDR4 RAM Corsair H80i GT V2 Liquid Cooler Sandisk 240GB Solid State Drive Western Digital Black Edition 1TB Hard Data Drive Corsair 750W 80+ Gold Certified Fully-Modular ATX Power Supply Corsair 750D Full Tower Case I just cant seem to figure this one out :(

@LiveMusicalPerformances 6 лет назад

First the bad news - it may be the drivers for the 2i2, as Focusrite has taken a lot of heat lately for their drivers being flaky. If you write to support, they may send you a link to the beta drivers they are testing now trying to fix the issues. I guess if you are using Bandicam, then try starting that first, then Voicemeeter. I suppose you have Voicemeeter's A1 set to use the ASIO drivers for the 2i2 to get the best latency - if not, definitely try that!

@Kitarachi 6 лет назад

Thanks for the heads up, I will also try that. I am switching between WDM & MME for A1 however I am using Windows 10 so I am cautious about WDM right now. My Voice Meter Banana A1 is "Line In (Scarlett 2i2 USB)" and my A1 Hardware output is "Focusrite USB 2.0 Audio Driver" which is using ASIO. I will let you know how I get on, many thanks Terry :)

@LiveMusicalPerformances 6 лет назад

MME is going to be very stable, usually (however, the latency/delay if you are trying to monitor your voice via headphones will drive you batty), but if you have an ASIO-capable interface, I'd definitely stick with the ASIO input of A1. Doing that sets everything else up in Voicemeeter to utilize the buffer settings of the Focusrite. NOW, if the Focusrite buffers are set too low, that will affect everything, so also watch that. I keep mine between 5 and 10 on the Focusrite slider.

@Juice8767 6 лет назад

Man, Sound Blaster Connect does NOT like playing well with WDM and my SoundBlasterX AE-5 doesn't let me drop MME's delay at all :(

@CreatorsMediaTools 6 лет назад

Yeah - None of them - KS/WDM or MME - are going to give you satisfactorily low latency to be able to monitor your microphone in your headset - there will always be that wonky delay to drive you crazy. I stopped monitoring the mic except in the beginning as a sound check with the EQ setting (the dot in the box), then I turn off the A button and just fly by meters to be sure signal is the right level. You really need an ASIO audio interface to tap into the low-low latency possible.

@CreatorsMediaTools 6 лет назад

Even with ASIO there is some delay. I just don't monitor the mic in the headphones anymore, period after I have the "sound" right.

@Juice8767 6 лет назад

Oh I know there's going to be some delay for sure but the problem I have is that it won't let me drop below 1024 which is JUST enough to make me trip over myself when speaking longer sentences. All that aside, great video!

@gfish2625 4 года назад

it did help ty :)

@AFTER_EMPIRE 6 месяцев назад

I almost packed my microphone back to the retailer,you the best bro 🫡🫡🫡

@jcalag Год назад

How can I disable WDM so I can enable the Swift Mode?

@CreatorsMediaTools Год назад

See these forum posts at VB-Audio: forum.vb-audio.com/search.php?keywords=swift

@ChristianParpartDev 4 года назад

While this helped a lot, I still think this is not a good solution in case you want to hear yourself while talking. I am having the Rode NT-USB microphone, which comes with a voice-back mode forged right into the hardware. So there I just plug in the headset into the microphone and can adjust how much I want to hear myself with a slider on the mic hardware. I am so used to that. Now I wanted to play around with Voicemeeter, which itself is a great software (apart from the crashes, so sad ;( ), and your video actually helped me understand how to lower the unbearable latency. I think this setting is good for reducing lag between video and audio sync (please tell me if there is a better way to do that, though - in case you know) - but for hearing yourself while talking, I sadly can't come around doing that via my mic's hardware-builtin feature. Thanks for your video though :)

@CreatorsMediaTools 4 года назад

I highly recommend always that people ONLY listen to their voice in the headphones as a sound check. If you are all about microphone technique and "play" the mic with changing your distance and all, then of course the hardware monitoring solution is fantastic! But for many uses, a quick sound check to make sure your mic levels match the media player you are using, for instance, is adequate, and then turn off the monitoring of the mic, as we hear our voice plenty well enough with very little latency as it resonates through our skull to our ears. :-) One day, Microsoft hopes to get WASAPI to ASIO latency speeds, but Voicemeeter only supports WDM/KS and MME. (Voicemeeter's ASIO is virtual ASIO via WDM same as ASIO4ALL.) When that day comes, monitoring a live microphone via headphones will become viable. I just got the Rode VideoMic NTG with the on-board monitoring for its USB mode, and it is nice to have that available, but I'd be surprised if I ever use it for anything other than podcasting, as I don't like headphones being in the shot, usually. (But who knows - my take on that for my own vidos might change someday!)

@kevgaming204 6 лет назад

you know how after you talk , the bar goes down, when its going down my friends say they can hear static noise , and when it goes down all the way to bottom then they stop hearing it , how do i fix this? so when i talk the bar goes up to the green then it starts slowly going down, how to i remove it instantly?

@CreatorsMediaTools 6 лет назад

That would be that knob above the slider, in Banana labeled "Gate". That is a pretty basic gate which lacks features of a full-blown one, but the higher you set it, the more it has to "hear" to allow signal to pass, so when you stop talking it would then drop to not-passing audio much more quickly. However, the noise is coming from the mic. If you are compressing also, don't! :-) That raises the noise floor so it is always audible if you have a lot of noise in the room (like laptop or PC fan noise, air conditioning, etc.) I leave compression all the way down, and very seldom use the gate, opting instead to lower the volume and subsequently the sensitivity so that I have to be right on top of the microphone.

@CreatorsMediaTools 6 лет назад

BTW, in regular Voicemeeter, the compressor and the gate are tied together to a single knob. Like I said, it is best to leave those knobs off and use the microphone-closeness technique instead if you can.

@kevgaming204 6 лет назад

so you want me to raise the gate abit higher/? like how much?

@CreatorsMediaTools 6 лет назад

A notch or two, perhaps. But better not to use it at all and simply turn down the mic's level and get closer to it if possible.

@xad1i256 3 года назад

very nice thanks

@polinatorrxtra4568 2 года назад

thanks it worked

@CreatorsMediaTools 2 года назад

Good to hear!

@brorafael1286 6 месяцев назад

thank you so much

@CreatorsMediaTools 6 месяцев назад

You're welcome!

@mauricescka Год назад

noice, thanks a lot

@ethanstidley8327 3 месяца назад

You can also use Potato and use the Monitoring Synchro Delay

@CreatorsMediaTools 3 месяца назад

Cool! Thanks for the heads-up! :-)

@yvessriccaldone9499 7 лет назад

Tried put same settings of audio Buffer MME 480 , all here don't working audio start cut it self many pices, idk how to describe it! Any suggestion

@yvessriccaldone9499 7 лет назад

like start to lag and not receive

@CreatorsMediaTools 7 лет назад

MME has the worst latency of them all. It dates back to Windows 95! Wasabi/WDM is the fastest regular Windows audio stream, with Kernel Streaming (KS) coming in close behind it. ASIO is your best bet in many cases.

@CreatorsMediaTools 7 лет назад

It does take several experiments to discover what works best with your hardware, so test many settings out.

@yvessriccaldone9499 7 лет назад

depends on audio card i think too, then every have different setup, thanks!

@zimty5345 6 лет назад

do you know how to turn off listening to yourself through voicemeeter?

@CreatorsMediaTools 6 лет назад

Merely click off the A button for the items you do not want to audition/monitor ("A" is for "Audition" is the memory aid).

@BrentonChan 3 года назад

Question, Terry: I cannot enable Engine Mode nor WDM Input Exclusive Mode. Both are faded out and when I try to select, this notice appears, "WDM Expert Options: These options are disabled. Engine Mode must stay NORMAL. WDM Input Exclusive mode must de disabled." Any tips to enable?

@CreatorsMediaTools 3 года назад

Are you running version 2.0.5.4? It is recommended in that version when it tried to leave them alone due to Windows bugs. Perhaps if you are running a more recent version Vincent has disabled those settings till it is fixed by Microsoft or he finds a workaround.

@BrentonChan 3 года назад

@@CreatorsMediaTools Yes, I'm currently using 2.0.5.8

@CreatorsMediaTools 3 года назад

I'm sure it will be fixed eventually, but meanwhile I would not worry. It did not improve things that much even when they worked.

@BrentonChan 3 года назад

@@CreatorsMediaTools Thanks for your assistance. Here's to the next update!

@dohnuts4405 6 лет назад

Thank you so much I finally got rid of that eeeecccchhhhoooooooo.

@CreatorsMediaTools 6 лет назад

SUPER!!!! :-D

@Aegir_D4 Год назад

I used to have zero delay with my Synth when I had the AUX output in my Daw, but now all of a sudden I have delay again when using Voicemeter and I have no idea what i've done.. any idea?

@CreatorsMediaTools Год назад

First, are you also using a pro-audio USB audio interface that uses ASIO? If so, select that device's ASIO output from the bottom of the A1 button at the top left, and you will then enjoy the same buffer settings and low latency of the audio interface. The fact that you had zero delay makes me suspect you do have an ASIO interface, but are using the Windows drivers for it in A1. Switch to the ASIO one and your problems may be solved. Let me know!

@Aegir_D4 Год назад

@@CreatorsMediaTools I use a DAW and a midicontroller, so I just just AUX out from Asio4all in the DAW, but well.. pretty much zero delay, not noticable when you play, but now all of a sudden on the new computer I have so much delay so I cannot use it. The reason I do use it in the first place is so I should be able to have sound in my DAW and in windows at the same time, because otherwise if I do not have Reason (The DAW) to aux out on the banana audio, the audio doesn't work in Windows (youtube etc), it only works in the DAW when it's open. Very odd.

@CreatorsMediaTools Год назад

Did you select the ASIO4ALL device from Voicemeeter's A1 dropdown at the top right? I use Voicemeeter all the time to combine Windows system audio with my ASIO audio inputs. But you need to assign the ASIO inputs to a channel strip in the System Settings/Options menu. (This is AFTER you've assigned the A1 dropdown to the ASIO output.) Have you done that? Also, when using Voicemeeter with a DAW, you need to go into the DAW and use the Voicemeeter Virtual ASIO as the audio device. (Voicemeeter then uses the ASIO4ALL itself as the actual ASIO device).

@Aegir_D4 Год назад

@@CreatorsMediaTools No I have not done that, I will try that, thank you. Im not sure how that works, you mean I go into system settings and then under patch asion inputs to strip I change one of them to 1? I've been using the AUX out for the DAW, so the one name Voicemeter AUX instead of the Voicemeter Input or whatever. Cause that's what the video I watched suggested, but I don't know. I just want working sound in windows/youtube at the I have my DAW open.

@CreatorsMediaTools Год назад

@@Aegir_D4 I'd better make a video showing how it is done. I'll let you know when I upload it, likely over the weekend.

@nasithomas4902 3 года назад

No homo but I love you this worked ❤️💯

@VAndy-bz7tk 2 года назад

THAAAAANK YOUUUUUUUU

@Blue-bb9ro 3 года назад

Thanks!

@CreatorsMediaTools 3 года назад

No problem!

@savvyexcalibur 3 года назад

@xxxcozz 7 лет назад

The Asio driver is an all around better driver but there's configurations to be made.

@CreatorsMediaTools 7 лет назад

Indeed, but one also needs an actual hardware ASIO interface on A1 to do it justice. :-)

@arkplot2172 4 года назад

..hum but the second channel is not better to disable it so you avoid all these maneuvers?

@CreatorsMediaTools 4 года назад

Sorry, I'm not clear as to what you are saying.

@Lukinhas4269 Год назад

thnks

@ns556308 9 лет назад

what can i say. you r just great .thnx

@CreatorsMediaTools 9 лет назад

Glad you found that helpful! :-)

@tonyfffuuu 5 лет назад

I can only record in audacity with wdm and none of the settings improved

@CreatorsMediaTools 5 лет назад

The biggest secret I give in the videos is to NOT monitor yourself through the headphones, except for briefly to do a sound check. After it is set, switch off the microphone's A (for audition) output. You really, really, really do not need it. For multi-tracking, you'll have to live with the latency and move the tracks to line up afterward. Let me know if you have any other questions. Have you tried using Voicemeeter Banana's built-in recorder to capture live tracks? Audacity will open the resulting multi-track file just fine.

@tonyfffuuu 5 лет назад

@@CreatorsMediaTools awesome this video is a bit old and i didn't expect a reply but thank you! I figured I'll try your advice i don't have the banana version is it worth getting?

@CreatorsMediaTools 5 лет назад

Yes, if only for the built-in recorder/audio file player (the "cassette" player/recorder).It adds separate control over three A outputs (A1, A2, and A3) and two possible VAIO and AUX outputs (B1 and B2) assignable individually to sliders and to the "Cassette".

@CreatorsMediaTools 5 лет назад

I generally reply to all comments the same day or the next day. So, if you have more questions, ask away! :-)

@mevoogle 3 года назад

i do this and then it starts crackling

@CreatorsMediaTools 3 года назад

That definitely can happen if you set the buffers too low. It depends on your hardware how low you can get away with.

@_BangDroid_ 7 лет назад

Any tips to reduce latency on the B (virtual out)?

@CreatorsMediaTools 7 лет назад

The best way is to use an ASIO interface and make that the "A1" device. Then the rest of your latency settings will follow the setting of that device's buffer settings. Otherwise, use KS or WDM where possible (KS is sometimes more stable if you have those available), and then tweak the buffers in the Settings dialog.

@_BangDroid_ 7 лет назад

I am currently using ASIO4ALL on A1 and the latency is as good as it can be when playing back. But the B virtual out, what other programs will use record the audio (such as Audacity or a RTA) currently experience about 500ms delay this does not change even if i adjust the buffer size. It seems the virtual B out has it's own buffer.

@CreatorsMediaTools 7 лет назад

Another thing that will improve the "B" latency is to connect with Voicemeeter via its ASIO interface directly (similar to connecting to ASIO4ALL). Audacity does not have that feature, but most other DAW recorders will show the "B" connections as ASIO ports (as well as an insert port being available from Voicemeeter). Audacity only offers MME (slowest), DirectSound (KS) and WASAPI (WDM), so that is the bottleneck right there in that you are constrained to using Windows Audio. (Though the DirectSound and WASAPI are pretty good, they are not up to ASIO speeds yet.) The cheapest DAW I know of that can connect to Voicemeeter via ASIO is Reaper, which is an incredibly deep piece of equipment costing only $60 for personal use. (However, do keep in mind that Voicemeeter itself can record directly into its own "Cassette" recorder in Banana, and that is a BWF multi-channel file that can be edited in Audacity or other DAWs). Why do you require that the recorder have low latency, though?

@lorenzoescanor4762 3 года назад

Doing this with my audio already messed up. What a nightmare lol

@CreatorsMediaTools 3 года назад

Heh... Avoid making things worse! 😁

@erthwermmusic 3 года назад

Upon installing Voicemeeter Banana and installing it, my entire computer's audio is delayed. Not by a second, but enough to notice.

@CreatorsMediaTools 3 года назад

Which applications? Some use WDM (pretty fast) others use MME (Pretty darn slow). You can adjust buffers in the settings, as seen in this video. However, you will ALWAYS hear mic delay if monitoring through headphones (which you likely didn't do before), so I advise everyone not to.

@erthwermmusic 3 года назад

@@CreatorsMediaTools I was using WDM! I was not using a mic. I meant my entire audio coming from my computer is delayed like 1/3 of a second. RU-vid video's audio, notifications, my music software, etc. Upon uninstalling VM Banana the small latency/delay went away. Is this short latency/delay just a part of having something handle sound other than my computers default?

@CreatorsMediaTools 3 года назад

@@erthwermmusic Perhaps. I am not sure how you are determining this delay. Gamers use Voicemeeter Banana and Potato all the time, and I've never heard of this kind of delay being bothersome before. Write to Vincent from his email at the bottom of his web page, and he will know more than I do about that. I personally connect my Voicemeeter A1 to an ASIO device, so I enjoy very low latency due to that.

@blakedmc1989RaveHD 2 года назад

ok there is a really bad delay on rhythm games when using voice meeter :/

@CreatorsMediaTools 2 года назад

Where do you mean? The only serious latency you should experience is monitoring your microphone over headphones, and I recommend NOT doing that, but instead turning the mic monitoring off once you have your levels set (and then watch your meters). The monitoring of the game audio should be in perfect sync on the A1 monitoring device, though there COULD be some delay on the A2 or A3 monitoring device,, especially if using straight WDM Windows audio. The situation improves vastly if you link the A1 monitoring to an ASIO hardware device.

@blakedmc1989RaveHD 2 года назад

@@CreatorsMediaTools well when i was testing out a rhythm game, it was very bad input lag when i use VoiceMeeter

@CreatorsMediaTools 2 года назад

@@blakedmc1989RaveHD There should not be significant input lag, but when not using ASIO hardware and using Windows audio, the monitoring lag can be very noticeable. That is why I do not advise trying to monitor inputs, especially on headphones. This isn't the fault of Voicemeeter, but is a failure of Windows audio.

@johnnyxthemacnifient5333 Месяц назад

forgot to say have voicemeter whti lighthost vor mic modulation

@StegiTrash Год назад

@CreatorsMediaTools Год назад

A question mark does not actually create a question. What puzzles you?

@rrnaBo 2 года назад

I know it's been 7 years from this, but it didn't help. for any real time voiceovers or singing, or any live event, you can't use VoiceMeeter due to latency issues, haven't found solution to it, adn I've been using it for 5 years now

@CreatorsMediaTools 2 года назад

You are correct, but it is not for Voicemeeter not trying! The Windows Audio subsystem inherently introduces a ton of latency. Although MS promised they would move WASAPI to "near ASIO latency specs," that never happened. Voicemeeter uses WDM, not WASAPI, and so works under those constraints. Nothing will surpass custom ASIO drivers written specifically for a particular piece of hardware. (Btw, ASIO4ALL is only an interface to WDM similar to Voicemeeter's Virtual ASIO, but they both are constrained by the workings of WDM itself.)

@rrnaBo 2 года назад

@@CreatorsMediaTools Thank you kind sir, for clarifying and answering on 7 year old video :)

@CreatorsMediaTools 2 года назад

@@rrnaBo I should mention that if you do have an ASIO audio interface, adding that as the A1 output device (as an ASIO device from the bottom of the dropdown list) does permit your interfacing with Windows apps like Skype and Zoom or RU-vid Live Streaming via OBS yet you can take full advantage of the monitoring capabilities of the ASIO hardware. This is how I accomplished my (many) live performances - without Voicemeeter those would have been very difficult to have realized! :-)