How to change audio sample rate — free, browser-based
- Step 1Open the converter — Open the sample-rate-converter tool. The WASM FFmpeg build loads in-page; nothing installs to your machine.
- Step 2Drop the mismatched file — Drag in any supported audio file —
.wav,.mp3,.flac,.m4a,.aac,.aiff,.ogg,.opus. The detected sample rate appears on the file card. - Step 3Read the current sample rate — Check the card. If it shows, say,
48,000 Hzor96,000 Hz, you have a candidate for standardising down to 44.1 kHz. - Step 4Run the conversion — Start processing. The tool resamples to 44.1 kHz and writes a 16-bit PCM WAV. There is no rate or format selector — 44.1 kHz WAV is the fixed output.
- Step 5Download the standardised WAV — Download the result. Every file you push through comes out at the same 44.1 kHz / 16-bit spec, which is what makes batch standardisation predictable.
- Step 6Need a different rate (16 kHz, 48 kHz, 96 kHz)? — Use JAD's programmatic API, which exposes the full 8–192 kHz ladder, or reach for a sibling tool: video-to-wav offers 48 kHz and 16 kHz options, and the downsample-for-speech guide covers 16 kHz targets.
Common rate mismatches and what breaks
Where a mismatch shows up and the symptom you actually see.
| Scenario | Source rate | Target system | Symptom of the mismatch |
|---|---|---|---|
| Video audio into a music DAW | 48,000 Hz | 44.1 kHz project | Import refused or playback slightly off-speed/pitch |
| CD-rate stem into a hi-res session | 44,100 Hz | 96,000 Hz interface | Glitches or resample-on-load warnings |
| Old loop into a sampler | 22,050 Hz | 44.1 kHz instrument | Loop maps to wrong key / sounds bright |
| Phone recording into broadcast | 44,100 Hz | 48,000 Hz delivery | Spec rejection at ingest |
In-browser tool vs the full rate ladder
What the web page does versus what JAD's engine supports. Verified against the client, processor, and MCP schema.
| Capability | In-browser web page | JAD programmatic API |
|---|---|---|
| Target sample rate | Fixed 44.1 kHz (no picker) | 8 / 16 / 22.05 / 44.1 / 48 / 88.2 / 96 / 192 kHz |
| Output format | WAV 16-bit PCM only | mp3, wav, flac, m4a, ogg, opus |
| Resampler | FFmpeg swresample (-ar) | FFmpeg swresample (-ar) |
| Where it runs | Your browser (WASM) | Server-side / programmatic |
| Channels | Preserved | Configurable |
Tier limits for this tool
From lib/tier-limits.ts (audio family). Duration cap is separate from size.
| Tier | Max file size | Max duration | Files per run |
|---|---|---|---|
| Free | 50 MB | 30 min | 1 |
| Pro | 200 MB | 120 min | 10 |
| Pro + Media | 100 GB | Unlimited | 100 |
| Developer | 100 GB | Unlimited | Unlimited |
Cookbook
Typical standardisation jobs done entirely in the browser. The web tool always produces 44.1 kHz 16-bit WAV.
Standardise a folder of mixed-rate WAVs to 44.1 kHz
You inherited stems at 44.1, 48, and 96 kHz. Push each through to get a uniform 44.1 kHz set your DAW can import without per-file resampling.
stem_a.wav 96,000 Hz → stem_a.wav 44,100 Hz stem_b.wav 48,000 Hz → stem_b.wav 44,100 Hz stem_c.wav 44,100 Hz → (skip: already 44.1 kHz) Result: one consistent project sample rate.
Confirm the mismatch before converting
The file card reports the source rate on drop. Use it to catch the file that is actually causing the import error.
Drop suspect.wav → card reads: 48,000 Hz · stereo That 48 kHz is why the 44.1 kHz DAW complained. Convert → 44,100 Hz · stereo.
No-install replacement for a one-off FFmpeg command
Instead of installing FFmpeg to run ffmpeg -i in.wav -ar 44100 out.wav, drop the file in the browser. Same swresample under the hood, zero setup.
Desktop equivalent you avoid installing: ffmpeg -i in.wav -ar 44100 out.wav Browser: drop in.wav → download out.wav at 44,100 Hz.
Hi-res capture down to project rate
A 96 kHz field recording needs to drop into a 44.1 kHz session. Down to 44.1 kHz in one pass; channel count is kept.
field_96k.flac 96,000 Hz, 2ch → field_96k.wav 44,100 Hz, 2ch, 16-bit PCM
When you genuinely need 16 kHz (not 44.1)
The page only emits 44.1 kHz. For a speech-model 16 kHz target, use the programmatic API or the dedicated guides instead of this page.
Need 16 kHz? The browser page outputs 44.1 kHz only. Use JAD's API (enum includes 16000) or see the downsample / Whisper guides linked below.
Edge cases and what actually happens
Expecting a sample-rate dropdown on the page
Not presentThe in-browser tool renders no rate selector — the target is fixed at 44.1 kHz. The full 8–192 kHz ladder exists in JAD's engine but is exposed via the programmatic API, not the web control.
Expecting a format dropdown
Not presentOutput is always 16-bit PCM WAV. To get MP3/FLAC/M4A, standardise to WAV here then convert with a sibling tool such as wav-to-mp3 or wav-to-flac.
Source already at 44.1 kHz
By designRe-running a 44.1 kHz file through gives you another 44.1 kHz WAV — a pointless re-encode. Check the card and skip files already at the target.
Need to upsample to 96 kHz
Not supported hereThe page goes to 44.1 kHz only and cannot upsample. Upsampling does not add real high-frequency detail anyway; if a downstream system demands 96 kHz, use the programmatic API's 96000 target.
Safari can't decode a FLAC/Opus for the preview
PreservedThe browser's Web Audio decoder may fail to render the on-page waveform/metadata for some FLAC/Opus files on Safari, falling back to an element-based duration probe. FFmpeg still decodes and resamples the file regardless.
File exceeds tier size or duration
RejectedFree is 50 MB / 30 min; Pro is 200 MB / 120 min. The minutes cap is checked independently of MB, so trim long files with audio-trimmer or audio-splitter first.
Variable-sample-rate source
NormalisedSome webm/opus captures carry an unusual nominal rate. FFmpeg normalises to the requested 44.1 kHz on output, so a quirky source rate resolves to a clean target.
Wanted bitrate change, not rate change
Wrong toolChanging sample rate is not the same as changing bitrate. If your goal is a smaller MP3 at the same rate, use bitrate-changer instead.
Wanted to keep it lossless and compressed
Two-stepThis tool's output is uncompressed WAV. For lossless-and-compact, resample here then run wav-to-flac.
Frequently asked questions
Do I have to install anything to change the sample rate?
No. The tool runs FFmpeg 8.1 compiled to WebAssembly directly in your browser. There is nothing to download or install, and your files are never uploaded.
What sample rate does the browser tool output?
44.1 kHz, 16-bit PCM WAV — fixed. The page has no rate or format dropdown for this tool, so it behaves as a one-click 'standardise to CD rate' button.
Can I pick 16 kHz, 48 kHz, or 96 kHz on the page?
Not on the web page. JAD's engine supports the full ladder — 8, 16, 22.05, 44.1, 48, 88.2, 96, 192 kHz — but those targets are reachable through the programmatic API, not the in-browser control.
Is the result really resampled, or just relabelled?
Really resampled. FFmpeg's swresample library performs an actual rate conversion when -ar 44100 differs from the source rate; it does not merely rewrite the header.
Why standardise to 44.1 kHz specifically?
44.1 kHz is the music/CD common denominator and the rate most DAWs, samplers, and distributors accept without complaint. Forcing a mixed-rate set to a single rate prevents per-file resample surprises on import.
Will this fix my 'wrong speed / wrong pitch' DAW import?
Usually yes. That symptom happens when a DAW plays a 48 kHz file as if it were 44.1 kHz (or vice versa). Converting the file to match the project rate removes the mismatch.
Does it change my channel count or bitrate?
Channel count is preserved (mono stays mono, stereo stays stereo). Bitrate is not a WAV concept — the output is uncompressed PCM. For bitrate work use bitrate-changer.
Can I batch a whole folder?
Free tier processes one file per run; Pro allows up to 10. For larger batches at non-44.1 targets, the programmatic API is the better fit.
What input formats are accepted?
Common audio containers: WAV, MP3, FLAC, OGG/OGA, Opus, M4A, AAC, ALAC, AIFF/AIF, and audio-in-WebM. The output is always WAV.
Does it preserve metadata?
It carries tags and embedded cover art across the conversion where the destination container supports it (-map_metadata 0). WAV holds fewer tags than MP3/FLAC, so consider tagging after converting with id3-editor.
Is there a file-size limit?
Free: 50 MB / 30 min, 1 file. Pro: 200 MB / 120 min, 10 files. Pro + Media and Developer: 100 GB, unlimited duration.
Which related tools should I know about?
video-to-wav for pulling 48/16 kHz audio out of video, wav-to-mp3 and wav-to-flac to compress the result, bitrate-changer for bitrate-only changes, plus the Whisper 16 kHz guide.
Privacy first
Every JAD Audio tool runs entirely in your browser via FFmpeg (WebAssembly) and RNNoise. Your audio files never leave your device — verified by zero outbound network requests during processing.