NEW · FFmpeg.wasm · 100% Browser-Based

Audio Sample Rate Converter

Convert audio sample rates with high-quality resampling — 44.1k, 48k, 96k, 192k, 22.05k, 16k, 8k, or any custom rate. Useful for matching DAW project sample rates or downsampling for legacy systems. FFmpeg.wasm runs locally. No upload, no signup, no watermark.

🎵
Drop an audio file
MP3 · WAV · AAC · OGG · FLAC · M4A
Files stay on your device · Never uploaded

Related Tools on UDT

Audio Converter →
Convert between MP3, WAV, AAC, OGG, FLAC, and M4A — handles codec changes.
Audio Compressor →
Compress with bitrate/quality presets — MP3, AAC, Opus.
Audio Normalizer →
EBU R128 loudness normalization to broadcast standards.
All Audio Tools →
Browse the full Audio Suite: 14 tools, all in-browser.

Why Do This in Your Browser?

Sample rate conversion happens implicitly all the time — every codec change involves a potential resample, every DAW import that doesn't match the project rate gets converted on the fly. But sometimes you need to do it explicitly: a 44.1kHz mix needs to deliver to a 48kHz film post-production pipeline, a 96kHz studio session needs to ship as 44.1k for streaming, a 48kHz voice recording needs to drop to 16k for a Whisper transcription run.

Explicit conversion separates the codec/container question (which the Audio Converter handles) from the sample-rate question. FFmpeg's aresample filter with its built-in swr resampler does the conversion, with a longer and steeper filter than FFmpeg's defaults. Preset buttons cover the common targets, plus a custom-rate input.

How It Works

Drop an audio file. Pick a target rate — the preset buttons cover the most common: 44.1kHz (CD audio, music streaming), 48kHz (video / film standard), 96kHz (high-resolution audio), 192kHz (very high-res, archival), 22.05kHz (low-bandwidth speech), 16kHz (telephony, Whisper input), 8kHz (very low quality, retro). Or type any rate from 4000 to 384000 Hz in the custom field.

FFmpeg applies -ar TARGET_RATE -af aresample=resampler=swr:filter_size=256:phase_shift=10:cutoff=0.95:filter_type=kaiser:kaiser_beta=14: a 256-tap Kaiser-windowed filter. Measured converting 48 kHz to 44.1 kHz: a 20 kHz tone passes at 0.0 dB, and a 23 kHz tone, which no longer fits under the new limit, is suppressed instead of folding back as a false 21.1 kHz tone (−135 dB in a 24-bit file). The output keeps the source's format and codec at the new sample rate.

Tip: For changing codec or container in addition to sample rate, use the Audio Converter — it handles both. To bring the result to a loudness target afterwards, the Audio Normalizer brings the result to target.

Common Use Cases

DAW Project Rate Matching
Convert imported audio to match the host project's sample rate before importing. Prevents implicit resampling artifacts during mixdown.
Streaming Delivery
Convert high-rate studio masters (96k or 192k) down to 44.1k for music streaming services when a distributor asks for 44.1 kHz.
Whisper Input Prep
Downsample to 16 kHz, the rate Whisper works at, so the file is smaller to load. (This tool keeps the channel count; Whisper mixes to mono itself.)
Telephony / IVR
Convert audio prompts to 8kHz for telephony systems. The legacy rate for voice over copper lines.
Film Post-Production
Convert 44.1k music to 48k for film/TV delivery. Standard industry workflow.
Library Rate Matching
Bring a library to one rate before archiving. Upsampling does not add detail (see the FAQ); it only makes rates match.

How We Compare

Honest read on free, paid, and self-hosted options for this kind of job:

UDT Audio Sample Rate Converter (this tool): Free, browser-based, FFmpeg.wasm aresample with high-quality filter settings. Preset buttons + custom rate input. No upload, no watermark, no signup.
Audacity (desktop): Free, full audio editor with Tracks > Resample. This browser tool is the focused alternative — no install, no project.
SoX (command-line): Free, gold-standard CLI resampler. A different, very high-quality resampler (soxr) that this browser build of FFmpeg does not include.
Adobe Audition: Paid desktop editor; Edit > Convert Sample Type.
FFmpeg CLI: Free, command-line; same engine as this tool, and desktop builds usually include the soxr resampler (-ar 48000 -af aresample=resampler=soxr). This tool is FFmpeg with a UI.
Online resamplers (cloud): Several free options. Most require upload and limit file size. UDT runs entirely in your browser.

Frequently Asked Questions

Is this audio sample rate converter really free?+
Yes — completely free, no watermark, no time limits, no signup.
What's the difference between this and the Audio Converter?+
The Audio Converter changes the format (MP3 to WAV, AAC to OGG and so on). This tool keeps the format and changes the sample rate. WAV keeps its bit depth (16, 24 or 32-bit); lossy formats are re-encoded at 192 kbps (MP3, AAC) or 160 kbps (Vorbis), so there is a small generation loss. Formats other than MP3, WAV, OGG, FLAC, M4A and AAC (Opus, for example) come back as MP3. MP3 can only hold 8 to 48 kHz and AAC up to 96 kHz; the tool says so instead of failing.
Which sample rate should I pick?+
44.1kHz for music delivery (streaming, CD, downloads). 48kHz for video/film delivery. 96kHz or 192kHz for high-resolution archival or studio masters. 16kHz for Whisper or other speech AI input. 8kHz for telephony. The presets cover all common cases.
Does converting to a higher sample rate improve quality?+
No. Upsampling cannot create information that wasn't in the original file. Going from 44.1k to 96k makes the file bigger without making it sound better. Upsampling is occasionally useful for keeping a consistent rate across a project, but it's not a quality boost.
What's the quality of the resampling algorithm?+
It uses FFmpeg's built-in resampler (swr) with a 256-tap Kaiser-windowed filter cut off at 95% of the new Nyquist frequency. Measured on 48 kHz to 44.1 kHz: 1 kHz and 20 kHz tones pass at 0.0 dB, and a 23 kHz tone is suppressed to -135 dB in a 24-bit file instead of aliasing to 21.1 kHz. The SoX resampler (soxr) is not in the FFmpeg.wasm build; for it, run desktop FFmpeg with -af aresample=resampler=soxr.
Are my audio files uploaded anywhere?+
No. FFmpeg.wasm runs entirely in your browser. The 32MB engine downloads the first time and your browser usually caches it. Once it has loaded, processing needs no network connection while the page stays open.
What's the maximum file size?+
Limited by browser memory. The whole file and the result are held in memory at once, so very large files, hundreds of MB and up, can fail, and phones run out sooner than laptops.
What's the underlying engine and license?+
FFmpeg.wasm v0.12.x with @ffmpeg/core v0.12.6 (aresample filter, built-in swr resampler). FFmpeg itself is LGPL by default, but the @ffmpeg/core build this tool loads includes x264, which makes that build GPL; the wasm wrapper is MIT.

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