# Resample an Audio File Change any audio file's sample rate — 8 kHz voice to 192 kHz studio — with high-quality ffmpeg resampling, right in your browser. Runs locally, nothing is uploaded, free. ## Run it - **CLI:** `gizza tool audio-resampler 'url=https://example.com/input' 'rate=44100' 'format=wav'` - **Web:** https://gizza.ai/tools/audio-resampler/ - **Agents:** machine-readable descriptor (parameters JSON Schema) at https://gizza.ai/tools/audio-resampler/tool.json ## Inputs - `audio` — Audio _(file; accept: audio/*)_ - `rate` — Target sample rate (Hz) _(field)_ - `format` — Output format _(field)_ ## Output - Resampled audio (audio) ## Query parameters Open the tool pre-filled and auto-run via URL: - `rate` — Target sample rate (Hz) - `format` — Output format - `url` — fetch the input file from a public URL (CORS-permitting) Example: `https://gizza.ai/tools/audio-resampler/?url=https://example.com/input&rate=44100&format=wav` --- ## Resample audio in your browser Change an audio file's **sample rate** — the number of samples per second, in hertz — to a target you choose. Upload a file, enter a rate like `44100` or `48000`, and it's resampled with ffmpeg's high-quality windowed-sinc resampler (the same `swresample` engine used by professional pipelines), complete with an anti-alias low-pass when downsampling. Everything runs locally in your browser tab via ffmpeg compiled to WebAssembly, so your audio is never uploaded to a server. Anything ffmpeg can decode works as input: mp3, wav, flac, m4a/aac, ogg, opus and more. ### Worked example You recorded a voice memo at 48 kHz but your transcription tool wants 16 kHz mono-friendly speech audio. Upload `memo.wav`, set **Target sample rate** to `16000`, leave **Output format** at `wav`, and you get `memo-16000hz.wav` at 16 kHz — smaller and exactly the rate the downstream tool expects. Going the other way (16 kHz → `48000`) changes the rate but cannot invent detail the source never captured; upsampling only makes the file play at the new rate. ### Common sample rates - **8000 / 16000 Hz** — telephony and speech; small files, ideal for transcription and voice models. - **22050 / 32000 Hz** — legacy and broadcast intermediate rates. - **44100 Hz** — CD quality; the safe default for music. - **48000 Hz** — the standard for video, DAWs and most editing workflows. - **88200 / 96000 / 192000 Hz** — high-resolution studio rates for mastering and archival. Any integer from 3000 to 384000 Hz is accepted — the presets above are just the common ones. ### Output formats - **WAV** (default) — lossless 16-bit PCM; the cleanest way to store a resample. - **FLAC** — lossless and compressed; smaller than WAV, still a perfect copy. - **MP3** — lossy; small and playable everywhere, encoded at 192 kbps. - **OGG** — lossy Vorbis; open format, good quality per byte, 192 kbps. - **M4A** — lossy AAC in an mp4 container; small high-quality files, 192 kbps. ### Limits and edge cases - Input files up to 10 MiB. - Choose **WAV** or **FLAC** to keep the resample lossless; the lossy formats re-encode at a fixed 192 kbps (dedicated bitrate control lives in the separate audio-convert tool). - Upsampling to a higher rate never adds fidelity the source lacks — it only changes how many samples per second the file stores. - Embedded album art is dropped: cover images ride along as a video stream, which audio-only formats like wav can't carry. - The output keeps the original filename with a rate suffix and the new extension (`song.wav` → `song-16000hz.flac`). ## FAQ
What's the difference between resampling and converting audio? Resampling changes the **sample rate** (how many samples per second, in Hz) — that's the whole point of this tool. Converting changes the **container and codec** (mp3 → wav, say) while keeping the rate. Use this tool when a downstream system demands a specific rate like 16 kHz or 48 kHz; use audio-convert when you just need a different file format or bitrate.
Does upsampling to 96 kHz or 192 kHz improve the quality? No. Upsampling recomputes the waveform at more samples per second but cannot recover frequencies or detail the original recording never captured. It's useful when a tool or device requires a higher rate, not as a way to "upgrade" audio. Downsampling, by contrast, genuinely reduces size and is applied with a proper anti-alias filter so it stays clean.
Which sample rate should I pick? `44100` for music (CD quality), `48000` for anything going into video or a DAW, and `16000` (or `8000`) for speech and transcription where small files matter. When in doubt, match the rate your target system or file already uses so nothing has to resample it again.
Which output format keeps the resample cleanest? `wav` (the default) or `flac` — both are lossless, so the resampled audio is stored exactly. Pick `mp3`, `ogg` or `m4a` only when you need a smaller file and can accept a lossy 192 kbps re-encode on top of the resample.
Is my audio uploaded anywhere? No. The page downloads an ffmpeg WebAssembly build once and then processes your file entirely inside the browser tab — the audio never leaves your device.
## Related tools - [Fit Audio to an Exact Length](https://gizza.ai/tools/audio-fit-to-length/): Make any audio file an exact number of seconds — pad it with silence when it's short or trim it when it's long. Runs in your browser, nothing uploaded. - [Shorten Long Audio Pauses](https://gizza.ai/tools/audio-pause-shortener/): Tighten speech pacing by shortening long silent gaps to a natural pause length in your browser. Nothing is uploaded, free. - [Add Chorus to Audio](https://gizza.ai/tools/chorus/): Add a configurable chorus effect to an audio file in your browser with ffmpeg: voices, delay, depth, rate, decay, and output format controls. - [Convert AIFF to FLAC](https://gizza.ai/tools/aiff-to-flac/): Convert AIFF or AIF audio to lossless FLAC in your browser — preserve samples and textual tags, choose FLAC compression level 0-12, and download the result. - [Bleep or Mute Regions of an Audio File](https://gizza.ai/tools/audio-bleep-censor/): Censor an audio file in your browser — bleep, mute, or duck the exact time regions you pick. Set a tone frequency and output format. Nothing is uploaded, free.