Have you ever sung on headphones and heard your voice arrive with a slight delay, like an annoying echo? Or noticed crackles and dropouts as soon as the session fills up with plug-ins? Both problems depend on the same parameter: the buffer. Understanding the trade-off it represents lets you set it deliberately.
Why a buffer exists
A computer does not process audio one sample at a time: that would be far too inefficient. Instead it collects a block of samples, processes them all together and hands them to the converter. This block is the buffer, and its size is expressed in samples: 64, 128, 256, 512 and so on.
While the computer prepares one block, the converter plays the previous one. If the computer doesn't finish in time, the converter runs out of data: you hear a click or a gap, the so-called dropout.
Calculating latency
The latency introduced by a buffer is calculated by dividing its size by the sample rate:
latency (ms) = buffer samples / sample rate × 1000
| Buffer | At 48 kHz | At 44.1 kHz |
|---|---|---|
| 64 samples | 1.3 ms | 1.5 ms |
| 128 samples | 2.7 ms | 2.9 ms |
| 256 samples | 5.3 ms | 5.8 ms |
| 1024 samples | 21.3 ms | 23.2 ms |
These values cover a single stage. Round-trip latency, from the microphone to the headphones through the computer, includes the input buffer, the output buffer, conversion times and often a driver safety margin. That is why the real value is always higher than the simple division. Many DAWs display the actual latency in their audio preferences.
How much latency is noticeable
Perception depends on what you are doing:
- Singing or speaking on headphones is the most critical situation: your own voice also reaches you through the bones of your skull, so even a few milliseconds of delay in the headphones can give a "doubled" feeling.
- Playing a virtual keyboard or electronic drums requires low latency to stay in time.
- Listening and mixing pose no problem: latency only translates into a short delay when you press play.
The trade-off
A small buffer gives low latency but asks the processor to work under very tight deadlines. A large buffer leaves more time to process, so it allows more tracks and plug-ins, but increases the delay.
A simple and widely used strategy:
- When recording: small buffer, 64 to 128 samples, with few active plug-ins.
- When mixing: large buffer, 512 or 1024 samples, to get all the available power.
Drivers: ASIO, Core Audio, WASAPI
The driver is the software layer that connects the DAW to the interface, and it has a big effect on achievable latency.
- Core Audio on macOS is built into the system and generally offers low latency without extra drivers.
- ASIO on Windows is the standard for professional audio: it lets the DAW talk to the interface while bypassing part of the Windows audio system. You should use the ASIO driver supplied by the interface manufacturer.
- WASAPI is Windows' modern audio system; in exclusive mode it can achieve reasonable latency, useful with devices that lack ASIO drivers.
"Generic" ASIO drivers that bridge to the Windows audio system work, but usually don't match the performance of a dedicated driver.
Reducing crackles
If dropouts appear even with reasonable buffers, the problem is often not processor power but something interrupting it:
- Power saving: set the power profile to high performance and plug the laptop in.
- Wi-Fi and Bluetooth: some network drivers cause latency spikes in the system; turning them off during a recording is a useful test.
- USB hubs: connect the interface directly to the computer.
- Plug-ins with lookahead: limiters and some compressors add their own latency. When recording, it is best to bypass them.
In my work, before blaming the hardware, I use a system latency diagnostic tool to see whether some driver is blocking the processor for too long.
Latency compensation
Modern DAWs automatically compensate for plug-in latency during playback, aligning all tracks. If a recorded track ends up a few milliseconds out of time, check the recording latency compensation setting: many DAWs let you measure it with a cable connecting the interface's output to its input.
In short
- The buffer is the block of samples processed together: the larger it is, the higher the latency.
- Latency is calculated by dividing the buffer by the sample rate.
- Record with small buffers and few plug-ins, mix with large buffers.
- On Windows use the manufacturer's ASIO driver; on macOS Core Audio is almost always enough.
- Crackles often come from power saving, network drivers or USB hubs.
