When we talk about sample rate and buffer size in Ableton Live, we are mainly concerned with latency during recording and playback. Latency can be annoying since it can lead to quantization errors and can throw you off with your audio playback being a few seconds off the grid.
In music production, ‘sample rate’ is almost always talked about when exporting audio such as MP3 or MIDI. In this case, we shall be looking at sample rate from the recording end. I will show you where you can find these settings and how you should make any changes or adjustments.
Let’s begin by defining what buffer size and sample rate are:
Buffer Size: simply put, this is the amount of time set to allow your computer to process the incoming signal.
Sample Rate: sample rate refers to the frequency (the number of times) your DAW captures an audio sample per second.
Both of these elements rely on your computer’s power to process incoming signals. Ableton makes these parameters controllable so you can adjust them to suit your sessions and accommodate your computer’s processing power. Here is how you find these controls, and adjust them:
Step 1: Open Ableton’s preferences (Cmd +, on Mac, or Ctrl +, on Windows)
Step 2: Find the parameter controls in the ‘Audio’ tab.
Note: what you are looking at with this window is the overall latency in your session. Changing these parameters will affect latency. Also depending on the recording interface you are using, this value may increase or decrease.
In this next segment, I will show you how to change the value of these parameters and give you a brief background on what they are, what they do, and why it is important in some instances to change their values.
Note that this control only affects audio recordings, but can affect the overall latency even when recording MIDI, and during playback. This is how you change the sample rate values.
Step 1: Open Ableton’s preferences, and look for Sample Rate in the Audio tab.
Step 2: now using the dropdown menu, select your preferred sample rate
Note! The higher the sample rate, the higher the overall session latency will be. If you are recording with the intent to export and share, the quality of the audio will remain the same even if the metadata of the exported audio indicates otherwise (a lower sample rate recording with a higher sample rate export)
More on Sample Rate
Sample rate refers to the number of times (frequency) that a recording platform/DAW, ‘samples’ the recorded audio per second. This ‘sampling rate’ in Ableton is marked by nodes, between one point of data. The higher the sample rate, the more the data nodes are created, and the more finely the recorded data is divided and processed.
According to the Nyquist theorem, it is important to sample audio at twice the highest recorded frequency to accurately convert and interpret from an analog signal to a digital signal. With this logic, if the highest frequency of a synthesizer recording is 16kHz for example, then the best possible sample rate should be at least 32kHz.
Buffer Size Adjustments / Changes
Like Sample rate, Buffer size is also measured in samples per second, however, it functions very differently from the sample rate. I shall explain this below but first, this is how you change the value of your buffer size:
Step 1: Open Ableton’s preferences and go to the Audio tab
Step 2: Using the buffer size drop-down menu, select the buffer size that suits your production. By default, this value is set at 512 Samples, as it is generally the most stable across most devices.
Note! While a lower buffer size may reduce latency in your session, the lower the buffer size, the more you are prone to hearing, pops, clicks, and other digital distortions. This effect is cool if you’re producing electronic music but it makes recording difficult and is also annoying during playback.
More on Buffer Size
Think of buffer size this way: when watching a YouTube video, the red line at the bottom of your screen indicates the point (timestamp) you are watching, and the white line shows how much of the video has buffered and can be played without loading. If you increase the video quality, it will take longer to buffer ahead of your playhead.
Buffer size affects the number of samples your computer can process and playback per second. If you increase this value, you will need more CPU power, and there will be more latency when you record. You’ll get more lag when inputting commands to your computer through your keyboard, mouse, or other input devices. If you are not recording, I suggest using the highest possible buffer size in your arrangement, mixing and mastering stages.
Changing the buffer size and sample rate of your session primarily affects the latency of your session. the value of your Buffer size and Sample rate affects the quality of audio playback you are hearing during your session where, the higher the value, the better the quality. I advise that you use the highest possible setting, especially with sample rate, when recording, for the most accurate, and highest quality recordings, and use the highest buffer size when monitoring. See what works for your use case, and see where, when and what you can compromise to get the most out of your productions. Have Fun!