Change Speed
Change audio playback speed without altering pitch — from quarter-speed to triple-speed.
100% in your browser — files never leave your device
Pitch-preserving speed change
Changing audio speed is useful in many practical situations: speeding up a recorded lecture to save listening time, slowing down a music track to learn an instrument part, or stretching a clip to fit a specific duration. A naive speed change — like resampling — shifts pitch proportionally to the tempo. That produces the "chipmunk" effect at high speeds and a sluggish, bass-heavy result at low speeds.
This tool uses FFmpeg's atempo filter, which uses time-domain harmonic scaling (TDHS) to change duration without touching frequency content. The pitch of voices and instruments stays the same at any speed setting. Supported range is 0.25x to 3x; the output duration scales inversely with the speed factor.
Processing runs entirely in your browser with FFmpeg.wasm. No file upload, no account, no limit on file size.
Frequently asked questions
Does changing speed affect audio pitch?
No. This tool uses FFmpeg's atempo filter, which changes playback speed while preserving the original pitch. It does this with time-domain resampling (TDHS), which stretches or compresses the audio without shifting frequency content. You get faster or slower audio that sounds the same tonally.
What speed range is supported?
0.25x to 3.0x. The atempo filter natively supports 0.5–2.0; speeds outside that range are handled by chaining multiple atempo filters. For example, 0.25x is achieved with atempo=0.5,atempo=0.5 and 3.0x with atempo=2.0,atempo=1.5.
Is my audio uploaded anywhere?
No. All processing runs in your browser using FFmpeg.wasm — a WebAssembly build of FFmpeg. The WASM binary (about 25 MB) downloads from a CDN on first use and is cached by your browser. Your audio never leaves your device.
Why does the first run take a moment to start?
The FFmpeg WebAssembly engine (~25 MB) has to download on first use. After that it is cached by the browser and subsequent operations start immediately.
What formats does this support?
Input: any format FFmpeg can read — MP3, WAV, OGG, M4A, FLAC, AAC, OPUS, AIFF, and more. Output: MP3, WAV, OGG, M4A, FLAC, or AAC. Lossy formats are encoded at 192 kbps; lossless formats preserve full bit depth.