How to convert 48 khz audio to 44.1 khz — free, no upload
- Step 1Open the Sample Rate Converter — Go to the sample-rate-converter tool. Everything runs locally — FFmpeg 8.1 (WASM) loads into the page on first use.
- Step 2Drop your 48 kHz file — Drag in your
.wav,.mp3,.flac,.m4a,.aac,.aiff,.ogg, or.opusfile. The file card shows the detected sample rate (e.g.48,000 Hz), duration, and channel count so you can confirm it really is 48 kHz before converting. - Step 3Confirm the source is 48 kHz — If the card reads
48,000 Hzyou are good to go. If it reads44,100 Hzalready, no conversion is needed — converting 44.1→44.1 just re-encodes with no benefit. - 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 picker on the page — 44.1 kHz WAV is the fixed output, which is exactly the target for CD/music delivery.
- Step 5Download the 44.1 kHz WAV — Download the result. It is a standard 44.1 kHz, 16-bit, PCM
.wav— verify it in your DAW or with a media inspector if your distributor is strict about specs. - Step 6If you need a different rate or a compressed file — The page only emits 44.1 kHz WAV. For 48 kHz delivery use video-to-wav (which exposes a 48 kHz option), and to turn the WAV into a smaller distribution file use wav-to-mp3 or wav-to-flac.
Why your file is 48 kHz and your target is 44.1 kHz
The two dominant sample rates, where each comes from, and which workflows demand which.
| Sample rate | Comes from | Expected by | Use this 48→44.1 tool when |
|---|---|---|---|
| 48,000 Hz | Cameras, phones, OBS/screen recorders, broadcast, most NLE timelines | Video editors, broadcast (EBU/ATSC), film post | You exported audio from a video project but need a music-spec deliverable |
| 44,100 Hz | CD masters, legacy music tooling | Spotify/Apple delivery, DistroKid/TuneCore, CD authoring | A distributor or CD plant rejected your 48 kHz file |
| 44,100 Hz (already) | A track already at CD rate | — | Never — no conversion needed; you would only re-encode |
| 96,000 / 192,000 Hz | High-res studio captures | Hi-res download stores (rare) | Not this tool's fixed target; the web page always lands on 44.1 kHz |
What the in-browser tool actually produces
Verified against the shipped client and processor. The web page has no options UI for this tool; these are the fixed defaults.
| Property | Value | Notes |
|---|---|---|
| Output sample rate | 44,100 Hz (fixed) | Passed to FFmpeg as -ar 44100; no on-page rate picker |
| Output format | WAV, 16-bit PCM (pcm_s16le) | No on-page format picker; always .wav |
| Resampler | FFmpeg 8.1 swresample (swr) | Invoked implicitly by -ar; not an explicit aresample/soxr filter |
| Channels | Preserved from source | Mono stays mono, stereo stays stereo — channel count is not changed |
| Tags / cover art | Carried where the container allows | -map_metadata 0; WAV carries limited tags vs MP3/FLAC |
| Upload | None | Processing is entirely in the browser via WASM |
Tier limits for this tool
From lib/tier-limits.ts (audio family). Duration is a separate cap from file 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
Real 48→44.1 situations, what you drop in, and what comes out. Output is always a 44.1 kHz 16-bit WAV from the web page.
Music exported from a video editor
You bounced a song's stereo mix out of Premiere or DaVinci Resolve. NLEs default to 48 kHz, but your distributor wants 44.1 kHz. Drop the 48 kHz WAV in and get a CD-rate WAV back.
Input: mix_master.wav — 48,000 Hz, 2ch, 16-bit Action: resample to 44,100 Hz Output: mix_master.wav — 44,100 Hz, 2ch, 16-bit PCM File-card readout before: 48,000 Hz · stereo File-card readout after: 44,100 Hz · stereo
Screen-recording audio destined for a music platform
A live-stream or OBS capture is 48 kHz. You want to release the audio as a track. Convert to 44.1 kHz first so the upload passes spec.
Input: stream_set.flac — 48,000 Hz, 2ch Action: resample to 44,100 Hz Output: stream_set.wav — 44,100 Hz, 2ch, 16-bit PCM Note: output is WAV, not FLAC. To re-compress, follow with wav-to-flac or wav-to-mp3.
Checking before you convert (no-op detection)
The file card shows the source rate. If it already reads 44,100 Hz, skip the conversion — resampling 44.1→44.1 only re-encodes and can only ever match or slightly soften the original.
Drop file → card reads: 44,100 Hz · stereo Decision: already CD rate — do NOT convert. Resampling 44.1 → 44.1 adds a re-encode with zero benefit.
Distributor rejected the upload for wrong sample rate
DistroKid/TuneCore-style platforms accept 44.1 kHz WAV cleanly. A 48 kHz file may be auto-resampled server-side (out of your control) or bounced. Do it locally so you hear the result first.
Rejected file: single.wav — 48,000 Hz Fix: resample to 44,100 Hz here Upload: single.wav — 44,100 Hz, 16-bit WAV ✓ accepted
Stereo stem set, one file at a time
The web page processes one file per run on Free/Pro single jobs. Resample each 48 kHz stem to 44.1 kHz, keeping channel count intact.
drums_48k.wav (48,000 Hz) → drums_48k.wav (44,100 Hz) bass_48k.wav (48,000 Hz) → bass_48k.wav (44,100 Hz) vox_48k.wav (48,000 Hz) → vox_48k.wav (44,100 Hz) Pro tier allows up to 10 files; Free is 1 file per run.
Edge cases and what actually happens
Source is already 44.1 kHz
By designConverting 44.1→44.1 is a re-encode, not a true resample. There is no quality gain and you add one lossy/lossless pass. Check the file card first — if it reads 44,100 Hz, you do not need this tool.
You need 48 kHz output, not 44.1 kHz
Not supported hereThe web page's target is fixed at 44.1 kHz with no picker. To produce 48 kHz audio from a video, use video-to-wav, which exposes a 48 kHz option. This tool only down-converts toward the 44.1 kHz music spec.
You wanted a compressed output (MP3/FLAC)
Not supported hereOutput is always 16-bit PCM WAV — there is no format dropdown on the page. Resample to 44.1 kHz WAV here, then run wav-to-mp3 or wav-to-flac to compress.
Non-integer ratio (48000 / 44100 = 1.0884)
HandledThis is exactly where cheap resamplers introduce artefacts. FFmpeg's swresample handles the non-integer ratio properly, so the conversion is clean for normal program material — no audible aliasing at sensible levels.
FLAC or Opus source on Safari
PreservedSafari's Web Audio decoder cannot decode some FLAC/Opus files for the on-page waveform/metadata probe, so the card may show a fallback duration. The FFmpeg resample itself still runs because FFmpeg decodes the file independently of the browser's decoder.
File over the tier size or duration limit
RejectedFree tier caps each file at 50 MB and 30 minutes; Pro at 200 MB / 120 minutes. Duration is checked separately from size, so a long, low-bitrate file can hit the minutes cap even under the MB cap. Upgrade or split the file first with audio-splitter.
Mono camera scratch track
PreservedChannel count is not altered — a mono 48 kHz source becomes a mono 44.1 kHz WAV. If you need stereo for delivery, run mono-to-stereo after resampling.
Expecting 24-bit WAV output
16-bit onlyThe WAV encoder is pcm_s16le — 16-bit. There is no 24-bit option in the in-browser tool. 16-bit is correct for CD/standard distribution; if you specifically need 24-bit you will need a desktop tool.
Distributor silently resamples server-side
AvoidedSome platforms accept a 48 kHz upload and resample it themselves with a resampler you cannot audition. Converting locally first means the 44.1 kHz file you ship is the exact one you have already listened to.
Frequently asked questions
Why do I even need to convert 48 kHz to 44.1 kHz?
Because the video world standardised on 48 kHz and the music/CD world on 44.1 kHz. Audio bounced out of a camera, phone, screen recorder, or video editor is almost always 48 kHz, while CD authoring and most music distributors expect 44.1 kHz. Shipping 48 kHz where 44.1 kHz is expected gets you a rejection or an uncontrolled server-side resample.
What exactly does the in-browser tool output?
A 44.1 kHz, 16-bit PCM WAV. The web page has no rate or format dropdown for this tool — 44.1 kHz WAV is the fixed default, which happens to be exactly the 48→44.1 job. Channel count is preserved from the source.
Does down-converting from 48 to 44.1 kHz lose quality?
Resampling itself is essentially transparent with a good resampler, and FFmpeg's swresample is good. You do discard the band between 22.05 kHz and 24 kHz — content above human hearing for almost everyone — but the audible range is preserved. The bigger 'quality' risk is doing it twice or via a cheap tool, which this avoids.
Is the resampler really FFmpeg, or a browser hack?
It is real FFmpeg 8.1 compiled to WebAssembly, the same codebase as desktop FFmpeg. The rate change is performed by FFmpeg's swresample library, invoked with -ar 44100. There is no upload — the WASM build runs inside your browser tab.
Can I get a 48 kHz file out instead?
Not from this page — its target is fixed at 44.1 kHz. If you are pulling audio from a video and need 48 kHz, use the video-to-wav tool, which offers a 48 kHz option.
Why is the output WAV and not MP3?
The tool emits 16-bit PCM WAV with no format picker. WAV is the right intermediate for distribution masters. If you need MP3 or FLAC, resample here then run wav-to-mp3 or wav-to-flac.
Will my ID3 tags and cover art survive?
The conversion uses -map_metadata 0 and copies embedded artwork where the container supports it. WAV's tag support is limited compared with MP3/FLAC, so if rich tags matter, convert to a tagged format afterward and set them with id3-editor.
Does it handle the odd 48000/44100 ratio cleanly?
Yes. That 1.0884:1 ratio is non-integer, which is where weak resamplers add artefacts. FFmpeg's swresample manages it correctly for normal program material.
How big a file can I convert for free?
Free tier: up to 50 MB and 30 minutes, one file per run. Pro: 200 MB / 120 minutes / 10 files. Pro + Media and Developer: 100 GB with unlimited duration.
Is anything uploaded to a server?
No. The entire pipeline — decode, resample, encode — runs in your browser through WebAssembly. Your unreleased mix never touches a server, which matters for confidential or pre-release material.
My file is already 44.1 kHz — should I still run it?
No. Converting 44.1→44.1 just re-encodes with no benefit and adds a needless pass. The file card shows the detected rate; if it reads 44,100 Hz, you are already at the target.
What other sample-rate jobs can JAD do?
For pulling 48 kHz audio out of video see video-to-wav; to compress the resulting WAV use wav-to-mp3 or wav-to-flac; to change bitrate without touching sample rate use bitrate-changer. See also the downsample-for-speech guide and the no-upload privacy 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.