With the M-Audio audio interfaces, you can create flawless studio-quality recordings with an intuitive and easy-to-use audio interface. For experienced users and beginners alike, issues may arise that result in audio input becoming distorted. This article can assist in troubleshooting problems you may experience with the input, output, and initial setup of your M-Audio audio interface.

Troubleshooting

• Video Series
  • Troubleshooting Audio Signal & Playback  
  • Troubleshooting Distortion, Clicks & Pops
• System Setup and Management
  • Drivers and Updates
  • Buffer Size
  • Sample Rate
  • Monitoring System Resources
• Hardware Setup and Gain Staging
  • Gain Staging
  • Connecting and Working with External Hardware
  • Distortion in Output, Metering, and Monitoring
• Further Support

Video Series

Before you browse this article, be sure to check out our video tutorials demonstrating some of the steps we describe below. Remember, although features may vary between each model, these troubleshooting steps are universal. 

Troubleshooting Audio Signal & Playback
 

Troubleshooting Distortion, Clicks & Pops

System Setup and Management

Static, clicks, pops, and distortion that creeps in over time are all common descriptions of audio distortion that could be generated while audio is being processed by your computer, software, or drivers. Every system will have its limits, and distortion/noise (and even crashing) is usually an indication that those limits have been reached.

The good news is that resolving these problems often means simply adjusting the speed at which the system is handling the incoming and outgoing audio signals, or changing the amount of processing the system is expected to handle at any one time. 

Drivers and Updates

First, check that you have the correct drivers installed. The M-Audio AIR Series interfaces are fully class-compliant for macOS users, so this step is only necessary for Windows users. 

If you are using a Windows operating system and have not yet installed the driver for your M-Audio AIR interface, please use the M-Audio Downloads page to locate the latest driver for your interface model:

1. Navigate to the support -> downloads page at m-audio.com.
2. Locate your product from the list, and expand the product tab to find the download link.

If you are unable to access the website or downloads page, please get in touch with our support team here, who will be able to assist.

1. Navigate to your Windows Control Panel > Hardware and Sound. There, you will see a listing for your M-Audio AIR Series device. Clicking this will open the Driver Control Panel.
2. Under the Windows Start menu, find the M-Audio folder in your app list, and inside you will find your Driver Control Panel.

 

Once your control panel is open, you can access the following:

1. Driver/Control Panel version
2. Streaming Status - Indicates whether or not audio is actively streaming through the interface
3. Buffer Size - See the Buffer Size section for more details.
4. Sample Rate - See Sample Rate section for more details. If this is greyed out and cannot be changed, this indicates that an application is already using the driver, or the interface is not connected. 

Using the process explained above, check the M-Audio Drivers & Updates page, or your user account for the latest available driver for your M-Audio AIR interface. Compare this with your driver/control panel version, and if necessary, download and install the updated driver. You will not need to uninstall the previous driver before installing the newer version. 

Buffer Size

The buffer size is the amount of time allotted for your computer to process incoming and outgoing audio signals. An appropriate buffer setting is entirely dependent on what your computer is capable of performing and may need to be adjusted depending on how much your system is doing at one time. In other words, you may find yourself adjusting this from time to time, and that's ok. 

Buffer Size vs. Latency

The buffer size is connected directly to what's known as "latency", i.e. the time it takes for the system to process audio and push it out to the interface and your speakers or headphones. Latency is inherent to some degree in all digital audio processing, so the goal is simply to manage it.

• A low buffer size means that the system will process the audio faster and latency will be reduced. But, this will be more taxing on your system and could mean more opportunities for errors, which become clicks and pops in the signal. Eventually, this may begin to compound and evolve into a deeper distortion that gets worse over time to the point that your audio signal is completely unrecognizable. 
• On the other hand, a high buffer size will increase latency but will allow your computer more time to safely process the audio and reduce errors (clicks/pops).
• The number of plugins in your project, the number of tracks you are recording, and anything else your computer is doing at the time (video processing/streaming, other open applications, etc.) will all require their own piece of the system processing. This will, in turn, affect the system's ability to process audio efficiently.

Managing your Buffer Size

You'll know that it's time to adjust your buffer size if you're experiencing one or all of the following:

• Clicks and pops
• Heavy distortion over time (only resolved by disconnecting/restarting the interface)
• Your DAW or software is throwing error messages or disconnecting the interface in some way.

If you are experiencing these symptoms, open the driver control panel and raise the buffer size. Because this will depend on the capability of your system and what you are doing, there is no default setting that will work for everyone. Here are a few tips to help you stay in the sweet spot:

1. Start with your buffer at 256 samples and go from there. If you find you still have trouble with clicks/pops/distortion, raise this a little and check playback again. If moving the buffer even a little bit temporarily resolves the errors, this is usually a good sign that a low buffer size was the problem. Continue to check playback from your software and move the buffer up as needed. 
2. A lower buffer size is optimal for recording because the reduced latency will generally make it easier to monitor your incoming audio signal while performing. There are a few ways to compensate for this while using your M-Audio interface:
   • Use the direct monitoring functionality on your AIR interface to monitor your recorded signal directly off of the interface. This virtually eliminates latency by allowing you to monitor the sound of your recorded signal before it is sent to the computer. To do this, use the USB/Direct knob to balance the incoming audio signal against the playback from your software. If the knob is set closer to Direct you'll hear more signal from the audio inputs, and if set closer to USB, you'll hear more playback signal from your computer/DAW. 
   • If you prefer monitoring through your DAW, try using fewer plugins and closing other software while tracking. This will let you set your buffer size lower during recording. 
3. If your project is in the mixing stage, don't be afraid to set your buffer much higher! This will give you space to add more plugins, tracks, automation, and anything else your system may need to do in the mix. You are only monitoring playback at this point, so latency is far less important. Be aware that the playback you hear will be delayed from what you see on your computer screen, but your audio signal will sound better and the system will run smoother.

Sample Rate

Another factor that may contribute to the audio distortion you are experiencing is the sample rate of your software platform.

Sample rate dictates how many times a sound is sampled per second. A "sample" in this case is like a snapshot of the sound itself at a particular moment. Much like a video camera would do for light to form a series of pictures that stream into video, audio samples are taken at a chosen rate and pieced together in a digital form. Higher sample rates will take more "samples" of the sound in less time, which results in a more detailed, higher fidelity sound. But, more samples means bigger files and more work for your system - your computer will work faster to capture and convert audio at the rate, as well as any plugins in your project, or any other form of processing that audio will hit before reaching your ears.

For reference, the standard sample rate for consumer audio, such as audio playback on a CD, is 44.1khz. That is 44100 samples of audio per second. The M-Audio AIR Series interfaces are capable of sampling audio at a rate of 192khz - over 4 times the standard rate! That is a very high rate of fidelity (some may argue too high), and there are plenty of reasons why you may choose to process your audio at this rate, or not. But knowing how it works will help you make an informed decision that keeps your project running smoothly from start to finish.

So, what does this have to do with audio distortion? If you are experiencing pops and crackling during playback, it's possible that your sample rate (in combination with the buffer size, project size, and anything else your system is doing) could be eating up more system resources than your system can handle. Remember that sample rate matters not only for what audio is coming in and out of your system, but how it is processed in between.

Unlike buffer size, a sample rate is inherent in the audio file itself, so it's not easy to change on the fly. If audio already exists in your project, or you are streaming audio live over the internet (i.e. twitch), changing the sample rate will either:

1. Change the playback speed of your audio (faster or slower depending on the difference in sample rates). If you are streaming live audio through Twitch, Discord, Facebook, or any other host, changing the sample rate while actively streaming will result in a different playback speed until the same sample rate is set within the host application, or the application is restarted. 
2. Require that the audio files in your project are re-sampled at the new rate (something most DAW software will request to do upon noticing a new sample rate). Re-sampling to a lower sample rate is generally considered ok, but going higher will not produce better-sounding files. 

For both buffer size and sample rate, it is important to note that these settings can be adjusted both within the Driver Control Panel, as well as within your audio software. Streaming platforms may not give you an option to set this and will default to the sample rate of your audio device upon startup. So there is a possibility that sample rates may not match if not set ahead of time or changed suddenly.

But what sample rate do I choose?

Generally, 44.1khz  or 48khz will be enough fidelity and easy enough for virtually any current computer system to handle efficiently, while giving you plenty of room to record multiple tracks at once, build larger projects, or simultaneously stream intensive video and audio, such as for online gaming.

96khz will be the next step up and is generally used for acoustic recording that requires higher fidelity and dynamics, such as classical music. If your computer has a fast processor and at least 16GB of RAM, this could be used for any style of music, though the benefits would be debatable in some cases. One thing is for sure though: you will need twice the amount of hardware space to store all those audio files!

Deciding what works for you will take some trial and error, but there are often tools available right in your software to help you monitor your system performance and decide if that high sample rate is worth the extra tax on your system. We've collected some links to system monitoring instructions for some of the more popular DAW and music production software below for your convenience, which includes in most cases how to monitor system performance right from your software and even tips on adjusting your Windows or Mac operating system to more efficiently process audio:

• ProTools
• Logic
• Cubase
• Ableton
• StudioOne
• FL Studio

Monitoring System Resources

A surefire way to know whether or not the distortion you are hearing is being caused by your computer is to check your system's resource monitor. Audio interfaces process data and send it to your computer at a speed dictated by several factors such as buffer size and sample rate (as discussed above). If your system is not processing that information quickly enough, distortion can occur.

To check and see your current system resources, you can use the:

• Windows - Windows Resource Monitor 
• Mac - macOS Activity Monitor

These will display your system's current CPU usage, physical memory, and disk space, as well as how much of these resources each application is using. If these applications show CPU usage and physical memory are at or approaching 100%, you may need to take action as discussed earlier in the article to try to reduce some of the strain.

Hardware Setup and Gain Staging

Gain Staging

If you are only experiencing distortion from specific input devices, your issue may pertain to Gain Staging. Gain Staging is the process of optimizing an audio signal through several stages of gain. More simply, it's making sure all of your volume and gain knobs are adjusted properly in a signal chain to keep the noise floor low and your signal distortion-free. For more info, check out our article on Gain Staging 101.
 

Proper gain staging is an important factor for attaining a robust signal without introducing distortion or clipping into the signal. Say, for instance, you are working with multiple devices - all of which have variable gain controls. Your devices could include instruments, compressors, equalizers, different types of modulation effects, etc. (Remember that plugins can also add gain digitally!) Since gain is a relatively common parameter shared among devices and plugins, you might find yourself in a situation where just about everything you're using has a gain parameter! While it may be intuitive to think, "I want a strong, defined signal; I should turn the gain way up!", this is not quite so, as that will likely result in an overloaded signal with unwanted noise and audio clipping. So what can be done to avoid this? Proper gain staging!
 

Each gain stage is certainly important as the sum of all gain stages will ultimately end up being your output signal but the first stage cannot be overlooked as it will set the tone, so to speak, for the rest of your gain stages. Assuming the AIR Series is the first stage (what you are connecting your audio source to), starting off with a quality input signal is paramount. The AIR Series's meters will provide a visual representation of the signal that is being input to each channel. The gain knobs are used to attenuate this signal in order to find just the right level for that particular input source whether it be a microphone or a musical instrument. Each subsequent gain stage has the potential to boost a signal too far into distortion/clipping territory. Properly setting your gain stages will ensure that you are left with a clear and workable signal. If your signal begins to clip at one stage, the clipping will carry over to all subsequent stages.
 

Something else to keep in mind when working with plugins in your DAW - Some plugins you may use will have a gain level as well as an output level. Hang on, they aren't the same thing? Not quite. To illustrate this, let's use guitar amplifiers as a starting comparison. It is not uncommon to find an amplifier that has both a gain control and a master volume control. Increasing the gain parameter will push the signal until you reach the edge of signal breakup and eventually an overdriven to a distorted signal. Yes, you will notice that a side effect of increasing the gain is an increase in apparent volume but the secret sauce is the breakup in the signal that can be achieved. But wait, what if you want the sonic qualities of an overdriven signal without the volume that will wake up your neighbors at two in the morning? This is where the master volume (output, in plugin terms) is useful. Gain is shaping your sound's timbre, master volume (output) is controlling the overall volume of the signal without affecting gain. These same principles apply to the plugins you are working with as well as other external hardware that has these types of gain and volume controls.

Connecting and Working with External Hardware

Instruments like guitars and basses that use passive pickups should be connected to the instrument inputs on the front of the AIR Series with standard 1/4" TS cabling. As passive pickups will yield a low-impedance signal, they will need to be boosted (the AIR's internal preamp will take care of this) to achieve a workable signal.
 

The opposite goes for guitars and basses that use active pickups - for these types of instruments, you should connect them to a combo input in the back of your AIR Series with standard 1/4" TS cabling. As these signals are already receiving a boost from the preamp within the instrument, they do not require an additional boost in the signal. Microphones and other line-level devices that connect via XLR or 1/4" TRS cabling should also be connected to the combo inputs. If you are using a microphone that requires 48v phantom power, be sure to enable that +48v switch!
 

If you connect your input device and it is very quiet, the AIR Series will register the signal around the -20dB meter mark or maybe even not at all. If that is the case, increase the gain. You should notice two things:

1. An increase in the apparent volume of the signal
2. The lights in the input meter are moving closer to the top/CLIP.

Adversely, if you see that your input is jumping right up to the red light and hitting CLIP, decrease the gain until your signal is no longer clipping.

Distortion in Output, Metering, and Monitoring

If you are receiving distortion or noise in your monitor output - pops, crackles, etc. a few things could be causing this.

1. Check your gain stages. If you are seeing red lights in the meters, something is clipping - reduce that something's gain level.
2. Check your system output volume. If your system volume is pegged at 100%, try backing this off by about 10 - 20% to give your signal some additional headroom.
3. The AIR Series interfaces were designed with a slightly hotter output than your garden variety audio interface. If you are experiencing unwanted audio artifacts in your output when setting the USB/Direct knob all the way to USB, simply back the knob away from USB slightly to give your signal more headroom.

Remember, there are plenty of reasons why you could have noise or distortion in your signal. Be sure to check back to the top of the article and the Buffer Size section for tips on resolving noise that may be coming from the system and not your interface.