The sound is breaking up. Crackles, clicks, pops or short dropouts come and go while you play back or record. Almost every time this is the computer failing to deliver the audio in time, not a broken plugin, and there is a short list of ordinary causes with a sensible order to check them, the same for any host on Mac or Windows.
This is host-agnostic. The idea is the same in every DAW or video editor; only the names differ, so check the wording in your own audio settings and mixer as you go.
Because the computer cannot deliver the audio fast enough to keep the stream smooth, so a gap or a late block comes out as a crackle. Your machine builds sound in small batches and hands them to the audio hardware on a strict clock. When a batch is not ready in time, you hear a click, a pop or a short dropout. This is nearly always about the computer keeping up, not a faulty plugin. The usual causes, in the order they are worth checking, are these.
There are two different things people call a click. A live crackle happens as the audio plays, moves around or comes and goes, and it is about the buffer, the CPU or the interface. A baked-in click is already recorded into the file, usually at an edit point or a fade that cuts across the waveform, and it plays back in the exact same place every single time. If the click is always in the same spot, leave the buffer alone; find the edit and add a short crossfade, or redraw the fade, instead.
If you are getting no sound at all rather than a crackle, that is a different problem: see why there is no sound from your plugin. This page assumes the audio is playing and only breaking up.
Most crackle clears with a few quick changes, in this order. Stop as soon as the sound is clean.
If it still breaks up, the sections below explain what each of these is doing and when to change it. The table maps the most common symptoms to a likely cause and a fix.
| Symptom | Likely cause | Fix |
|---|---|---|
| Crackle right through playback | Buffer size too small for the project | Raise the buffer to 512 or 1024 samples in the host's audio settings |
| Clean at first, crackles as the session fills up | CPU running low on headroom as tracks and plugins add up | Freeze or bounce heavy tracks, disable unused plugins, raise the buffer |
| Crackle only while recording or playing live | Buffer set low for low latency, so blocks arrive late | Use the lowest buffer that stays clean, or monitor through the interface and raise it |
| Pops, or the audio plays at the wrong speed or pitch | Project and interface at different sample rates | Set both to the same rate, usually 44.1 or 48 kHz |
| The same click in the same spot every time | A click recorded into the audio at an edit or a bad fade | Find the edit and add a short crossfade, or redraw the fade |
| Crackle on the external interface but not the built-in output | A driver, USB cable or port problem | Use the interface's own driver, try another cable and a port on the computer, update the driver |
| No sound at all, not a crackle | A different problem | See why there is no sound from your plugin |
Buffer size is the small batch of audio your computer prepares in advance before it plays, measured in samples. A larger buffer gives the machine more time to build each batch, so it is far less likely to fall behind and crackle. The cost is latency, a small delay between an action, such as pressing a key, and the sound you hear. That is the whole tradeoff, and it is why there is no single best number.
For playback and mixing, set the buffer high, such as 512 or 1024 samples. Nothing is being played live, so the small delay does not matter, and the extra headroom keeps playback rock steady. For recording or playing an instrument live, set it lower so the delay stays small enough to play in time, and accept that a very low buffer is the setting most likely to crackle. Raising the buffer is the single most effective fix for crackle during playback.
Latency is the delay between doing something and hearing it. A small buffer means low latency and responsive playing, but less time for the computer to keep up, so more risk of crackle. A large buffer means the opposite: steady, gap-free playback, but too much delay to play a part in time. You are not picking one number forever. You raise it to mix and lower it to record, and you change it as you go.
Because every track, plugin and effect adds to what the computer has to calculate in the same fixed slice of time, and at some point it runs out of headroom and starts delivering audio late. A session that played cleanly when it was small can crackle once it is full of instruments and effects, even though nothing is broken. This is CPU overload, and most hosts show it on a performance or load meter that climbs toward the top as it happens.
The fixes all come down to giving the machine less to do, or more time to do it in.
A few very heavy plugins can cost more than many light ones, so look at what is actually loaded rather than the track count alone.
Recording is the one time you need a small buffer, and a small buffer is exactly what causes crackle, so the aim is to find the lowest setting that still plays cleanly and to change your approach rather than fight it. When you record you want low latency, so you can play or sing in time, which means a low buffer, which leaves the computer little room to keep up. That is the tension to manage.
Some plugins add a little latency of their own, so a busy chain on the track you are recording can add up even at a low buffer. Keeping that track light while you play, and adding the heavy effects afterward, keeps you in time.
If the buffer is high and the project is light and it still crackles, the cause is usually one of these three.
Your project has a sample rate and your interface has one, and they need to be the same. Common values are 44.1 kHz and 48 kHz. When they disagree, playback can crackle, pop, or come out at the wrong speed and pitch. Set the project and the interface to the same rate, in your host's audio settings and in the interface's own control panel, then reopen the project.
The driver is the software that lets your host talk to your audio hardware, and using the right one matters. On Windows, choose the ASIO driver written for your specific interface rather than a generic or shared driver, which tends to be higher latency and less stable. On a Mac, Core Audio is built in, so just select your interface as the device. With an external interface, a poor USB cable, a hub, or an underpowered port can also cause dropouts, so try a different cable and a port directly on the computer, and check the maker's site for an updated driver.
Anything else demanding the processor or the disk at the wrong moment can make audio arrive late. Large downloads, backups, virus scans, sync services and spare browser tabs are common culprits. Close what you do not need while you work, and if your laptop has a battery saver or low power mode, audio is usually steadier with it off and the machine on mains power.
Most of this page is about your own computer keeping up rather than a faulty plugin. The fastest way to tell the two apart is to hear a plugin you know is clean, with none of your own routing or settings in the way.
Every Tunary instrument runs as a browser demo that needs no install and no account, and they are light on the processor, so you can hear one play in seconds. The Collection is three of them: ARGISH, SILT and REHEAT. If one plays cleanly in your browser but crackles in your DAW, you know the plugin is fine and the cause is a buffer, a driver or a sample rate in your own setup, which is exactly what this page is for.
Try one free in your browser first, no install and no account. The Collection is three instruments in one purchase.
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