No Audio After Recording

Okay, that’s what I have it set to is the AISO driver. What should I set it to? I will try that.

Okay, I lowered it to 256, but now there is no pausing and no clicks. I should put it to a higher number or lower? I think the original setting was 512 or something.

Hi, On the second video, I am not sure, but the clicking sounds like the click track or metronome. Make sure it is turned off. Find that in the Transport menu. If it is already off, that is not the issue.

In Audio Connections, I suggest you set the Stereo Out outputs to the left and right sides of your audio interface (or, if applicable, your computer’s audio device). That will most directly allow you to monitor through the interface (or computer) via headphones or monitor speakers.

Also to avoid random-seeming clicks and glitches in audio recordings, you will want to increase the buffer size in ASIO4All until the problem is fixed. Bigger buffer size=lower demand on your processor. That translates to avoiding clicks and glitches. As far as buffer size, the main downside with increasing it is that it also increases latency. Latency could be an issue during recording, but not during mixing or playback. Depending on what you are recording, and in which context, latency may or may not be an issue for you. Trial and error with various buffer settings in your actual use case will help you quickly see what works. You can see how changes in buffer size affect latency in Studio Setup/Audio System, where the latency times are shown in MS, and change with changes in buffer size.

Lastly, be sure to set your recording levels high enough, but not too high, while tracking. That is usually done with the input/preamp level knobs on your interface. If the recording level is too high, you could risk clipping/distortion; recording level that is too low risks increasing the audible noise level when it’s brought back up to proper level during mixing and playback. If you are dealing with an already recorded track that was done at a too low level, try increasing the gain using the Pre Gain knob in the Cubase mixer, or clip gain in the project window.

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I appreciate the tips. I have the Out outputs set to left and right. As far as increasing buffer size I’m not sure how far up I can go. I went up as high as 700 and it didn’t change a thing for me. It has to be something so simple and I’m missing it. I apologize for all the trouble.

It is getting better!!! My goodness… it started to improve around 800. I’m not sure that is normal though… incredible. Thank you I will keep working with this until the issue is gone. I will post with an update.

When you go into Audio Connections, on the Output tab. Click in the Device Port column for Left and Right, under Stereo Out. See if there are multiple output device options available. Instead of VoiceMeeter, select your interface (or your computer’s audio device) for left and right.

You can go as high as you wish on buffer size. On my system, for example, when mixing, I set mine at 1536 Samples!

By the way, I suggest you check M-Audio website to find their driver for your interface. In my experience, manufacturer’s USB drivers usually (but not always) give better results than ASIO4ALL. It’s worth trying.

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If your computer has a small amount of RAM and a slower CPU, setting the buffer too low can tax your system. Audio dropouts, clicks and other odd things can then occur.

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So far it still clicks and pauses, more audio though at least I can hear a snippet of my acoustic LOL … anyway I’m going to fool around with this for a while and see. I will download that driver from M-Audio and see if that will help. Thanks again and I’ll post back here probably tomorrow sometime with results. APPRECIATE YOU ALL!


I agree. Most of the native drivers are much better compared to the generic ASIO4ALL driver.

It depends on your RAM and the driver. There are drivers which allow you to go up to 2048 or even 4096. It also depends on what you are doing. While recording, you need lower latency (it means lower buffer size). While editing and mixing, you don’t need low latency, so you can go up as much as you can.

I would advice take VoiceMeeter out of the equation first, deal with real audio card output first before starting to introduce that virtual thing. VoiceMeeter’s routing is not that intuitive. You might think it will solve some routing/splitting problem for you - but it in the end does not.


I removed voicemeter as suggested. I’m still not getting any improvement other than what was done earlier from changing the buffer size. So, I decided I will uninstall the app and reinstall. I’ll update the post when I can get to it. It may be a few days or could be when I return from the Netherlands later this month. Appreciate you guys!

Good luck! I hope you can fix it