The AudioBuffer interface represents a short audio asset residing in memory, created from an audio file using the {{ domxref("BaseAudioContext/decodeAudioData", "AudioContext.decodeAudioData()") }} method, or from raw data using {{ domxref("BaseAudioContext/createBuffer", "AudioContext.createBuffer()") }}. Once put into an AudioBuffer, the audio can then be played by being passed into an {{ domxref("AudioBufferSourceNode") }}.
Objects of these types are designed to hold small audio snippets, typically less than 45 s. For longer sounds, objects implementing the {{domxref("MediaElementAudioSourceNode")}} are more suitable. The buffer contains data in the following format: non-interleaved IEEE754 32-bit linear PCM with a nominal range between -1 and +1, that is, 32bits floating point buffer, with each samples between -1.0 and 1.0. If the {{domxref("AudioBuffer")}} has multiple channels, they are stored in separate buffer.
Objects of these types are designed to hold small audio snippets, typically less than 45 s. For longer sounds, objects implementing the {{domxref("MediaElementAudioSourceNode")}} are more suitable. The buffer contains data in the following format: non-interleaved IEEE754 32-bit linear PCM with a nominal range between -1 and +1, that is, a 32-bit floating point buffer, with each sample between -1.0 and 1.0. If the {{domxref("AudioBuffer")}} has multiple channels, they are stored in separate buffers.