Audio Latency and Buffer Size: Settings for Recording vs Mixing

What is buffer size in a DAW?

Buffer size is the number of audio samples your computer collects and processes in one chunk before passing them on. A bigger chunk gives the CPU more time to finish its work but makes you wait longer to hear the result.

Ableton's help center describes all audio in Live as processed in defined parcels of time called audio buffers, which smooth out events and help prevent dropouts. The bigger the buffer, the bigger the latency, and your driver type and interface affect latency too.^[2] Every DAW works on the same principle, whether the setting is called buffer size, I/O buffer size or buffer length.

Latency is the delay between a sound entering the system and coming out of the speakers. You feel it when you hit a pad on your controller and the drum arrives a beat late, or when your own voice in the headphones sounds like a slapback echo.

• Small buffer (32 to 128 samples) Low latency, high CPU load. Good for playing instruments and recording through plugins.
• Medium buffer (256 to 512 samples) Noticeable but usable delay. A common middle ground for producing with a few software instruments.
• Large buffer (1024 to 2048 samples) Long delay, low CPU strain per block. Best for mixing heavy sessions where nobody is performing live.

How do you convert buffer size to milliseconds?

Divide the buffer size in samples by the sample rate in kHz. Ableton gives the example of 256 samples at 44.1 kHz producing 5.8 ms of expected latency for one stage of conversion.^[1]

The table shows one buffer's worth of delay. Your real monitoring delay is at least twice this, because audio passes through an input buffer and an output buffer.

Notice that doubling the sample rate halves the time per buffer. Apple's Logic Pro support notes that higher sample rates result in less input monitoring latency, with the trade-off of larger files and more load on the processor.^[3] For most home studios, lowering the buffer is a cheaper fix than running the whole project at 96 kHz.

What is round-trip latency?

Round-trip latency is the total delay from the moment sound enters your interface input to the moment it leaves the output, including every buffer and converter in between. It is the number that decides whether monitoring through your DAW feels natural.

Apple defines roundtrip latency in Logic Pro as the total input monitoring latency from audio input to audio output and shows it under the I/O Buffer Size menu.^[3] Ableton explains why its Overall Latency readout is higher than the simple formula: it includes both input and output latency (roughly the formula times two), extra layers of processing vary, and some interfaces report inaccurate values.^[1]

• A/D conversion The interface turns the analog signal into data. Fixed by the hardware, usually small.
• Driver and input buffer Audio waits for a full buffer before the DAW can touch it. This is the part your buffer setting controls.
• DAW and plugin processing Plugins with lookahead, linear-phase filters or oversampling add their own delay, which the DAW compensates by delaying everything else.
• Output buffer and D/A conversion A second buffer and the output converter before sound reaches your headphones.

For a feel of scale: sound travels about 343 meters per second in air at 20 °C,^[6] so 10 ms of latency is like standing about 3.4 m (11 ft) from your amp. Drummers and vocalists usually notice that; a pad player often does not. Treat 10 ms round trip as a practical upper target for tracking and adjust to what the performer tolerates.

What buffer size should you use for recording vs mixing?

Use the smallest buffer that plays without crackles while recording, then raise it for mixing. Apple's advice for Logic Pro is to set the buffer to the lowest setting when recording and increase it one step if you get system overload alerts.^[3]

1. Open the audio settings
   In Logic Pro it is Settings, Audio, Devices. In Ableton Live, Settings, Audio. In FL Studio, Options, Audio settings.
2. Lower the buffer before recording
   Start at 128 samples. If playback stays clean, try 64. Watch the latency readout change.
3. Play the session for a full minute
   Crackles, pops or CPU overload warnings mean the buffer is too small for this session. Go up one step.
4. Record the take
   Keep heavy mastering plugins bypassed or frozen while you track, so they do not add delay.
5. Raise the buffer for mixing
   Once everyone is done performing, switch to 1024 or 2048 samples so you can add plugins without dropouts.

ASIO or Core Audio: which audio driver should you use?

On Windows, use your interface manufacturer's ASIO driver. On a Mac, use Core Audio, the audio system built into macOS.

ASIO (Audio Stream Input/Output) was developed by Steinberg, and Steinberg says it is required by practically all professional audio programs under Windows, partly because of its low latency. The same page explains that ASIO drivers are not required on macOS because of Apple's Core Audio infrastructure.^[4]

Ableton notes that low buffer sizes are hard to reach with the default MME/DirectX driver on Windows, recommends installing your interface's ASIO driver, and suggests ASIO4ALL only when no native ASIO driver exists. It also warns that Bluetooth and Wi-Fi audio add much higher latency.^[2] If your driver refuses to show up or keeps dropping out, the Windows ASIO driver troubleshooting guide covers the fixes. Mac users combining two devices can follow Core Audio multi-interface setup.

Universal ASIO drivers

Generic drivers such as ASIO4ALL, FL Studio ASIO and Steinberg's built-in ASIO driver let a laptop's onboard sound chip talk to a DAW. Steinberg notes that its built-in driver uses a fixed buffer size that cannot be changed.^[4] They are fine for sketching beats, but a dedicated interface with its own driver gives you real control over latency.

How do you record without hearing latency?

Use your interface's direct monitoring, or switch your DAW into its low-latency monitoring mode so heavy plugins are skipped while you record. Either way, the performer hears themselves on time.

Focusrite describes direct monitoring as sending the input straight to the headphone or line outputs with near-zero latency while the signal still goes to the DAW for recording. Mute the track you are recording to, or you will hear yourself twice.^[5] The catch is that you hear the dry signal, not your DAW reverb or tuning plugin.

When you do need effects in the headphones, DAWs offer shortcuts. Logic Pro's Low Latency Mode bypasses latency-inducing plugins such as dynamics with look-ahead so the delay stays under your set limit.^[3] Ableton's Reduced Latency When Monitoring skips the session-wide delay compensation on monitored tracks, and freezing latency-heavy tracks removes their delay entirely.^[1] For a full vocal setup, see how to record vocals at home.

Why does a small buffer cause crackles and dropouts?

With a small buffer, the computer has to deliver each chunk of audio much sooner, so any moment where the CPU falls behind becomes an audible gap. Ableton describes those missing chunks as crackles or dropouts.^[1]

If you get crackles at a buffer your machine should handle, the cause is often outside the DAW: power saving, a busy USB hub, background apps or an outdated driver. The companion guide on Windows 11 optimization for music production walks through those system settings. Inside the DAW, freeze or bounce CPU-heavy synths and check CPU optimization for large projects before you raise the buffer for good.

Buffer size does not change the sound of your mix or your exported file. It only decides how long you wait and how hard your CPU has to sprint, so there is no quality reason to keep it low while mixing.