Audio & video / Local FFmpeg tool

Audio Sample Rate Converter

Resample audio to 22.05, 44.1 or 48 kHz and export a verified MP3 at a bitrate that rate can actually carry.

● Files stay in your browser
Files stay in your browser

The first run loads an approximately 31 MB FFmpeg WebAssembly core from this site. It does not use a CDN or upload your files.

Input stream details

Choose a file and the sample rate, channels and codec detected on this device appear here.

Files are limited to 200 MB total and 20 minutes; mobile devices use an 80 MB limit. Reverse, GIF and boomerang tools have lower duration caps.

Choose files, set the controls, then start.Progress: 0%

Processed result

The first run loads an approximately 31 MB FFmpeg WebAssembly core from this site. It does not use a CDN or upload your files.

Validated downloads will appear here after processing.

Practical guide

Choose the sample rate your destination actually needs

Sample-rate conversion changes how many samples are written per second. Use the rate required by the destination editor, player or delivery specification instead of choosing the largest number automatically.

Steps

  1. Choose one audio file from this device.
  2. Select 22,050 Hz, 44,100 Hz or 48,000 Hz.
  3. Choose the MP3 bitrate for the exported file.
  4. Process locally, verify playback and download the result.

Settings and output

22.05 kHz
Use only when a voice, preview or legacy workflow requests the lower rate.
44.1 kHz
A common choice for music and general audio delivery.
48 kHz
A common target for video, broadcast and production timelines.
How to useHow to use Audio Sample Rate Converter

What the tool actually does

Resample audio to 22.05, 44.1 or 48 kHz and export a verified MP3 at a bitrate that rate can actually carry.

Reproducible example

Example input
Audio file: sample-1.wav
Sample rate = 48,000 Hz
MP3 bitrate = 192 kbps — balanced
Expected result

Creates a probed and validated .mp3 (audio/mpeg). Actual method: Resampling changes the number of samples per second; it cannot recreate frequencies missing from the source. MP3 bitrate is not independent of that choice: 44.1 and 48 kHz use MPEG-1 Layer III and reach 320 kbps, while 22.05 kHz uses MPEG-2 Layer III and stops at 160 kbps. The bitrate picker narrows itself so the encoder is never asked for a combination it would silently downgrade. Output boundary: The result is probed after encoding and the actual sample rate, codec and bitrate are shown next to the target. MP3 is lossy at every bitrate — use the Audio to WAV converter when you need an uncompressed 16-bit PCM master.

  1. 1

    Choose Audio file sample-1.wav and first confirm that each required audio stream exists.

  2. 2

    Use Sample rate = 48,000 Hz; MP3 bitrate = 192 kbps — balanced. The actual FFmpeg rule is: Resampling changes the number of samples per second; it cannot recreate frequencies missing from the source. MP3 bitrate is not independent of that choice: 44.1 and 48 kHz use MPEG-1 Layer III and reach 320 kbps, while 22.05 kHz uses MPEG-2 Layer III and stops at 160 kbps. The bitrate picker narrows itself so the encoder is never asked for a combination it would silently downgrade.

  3. 3

    A download appears only after duration, stream type and .mp3 container validation. Also check this boundary: The result is probed after encoding and the actual sample rate, codec and bitrate are shown next to the target. MP3 is lossy at every bitrate — use the Audio to WAV converter when you need an uncompressed 16-bit PCM master.

Local AI / Speech to text

Transcribe the current audio or video

Open-source Whisper Tiny runs in your browser. The file, audio and result are never uploaded.

◎ First use loads about 64 MB of model and runtime files

Choose an audio or video file in the workbench above first.

◎ The model loads only from 321Kit; MiniMax is not called and no transcript history is stored.

Transformers.js + Whisper Tiny (Apache-2.0)

Transcript

How it works

Audio Sample Rate Converter

Resampling changes the number of samples per second; it cannot recreate frequencies missing from the source. MP3 bitrate is not independent of that choice: 44.1 and 48 kHz use MPEG-1 Layer III and reach 320 kbps, while 22.05 kHz uses MPEG-2 Layer III and stops at 160 kbps. The bitrate picker narrows itself so the encoder is never asked for a combination it would silently downgrade.

How does this tool process my file?

Resampling changes the number of samples per second; it cannot recreate frequencies missing from the source. MP3 bitrate is not independent of that choice: 44.1 and 48 kHz use MPEG-1 Layer III and reach 320 kbps, while 22.05 kHz uses MPEG-2 Layer III and stops at 160 kbps. The bitrate picker narrows itself so the encoder is never asked for a combination it would silently downgrade. Processing uses single-thread FFmpeg.wasm. The page does not upload, store or inspect file content outside this device. Cancel, error and page close terminate the worker and revoke temporary object URLs.

What limits apply to the output?

The result is probed after encoding and the actual sample rate, codec and bitrate are shown next to the target. MP3 is lossy at every bitrate — use the Audio to WAV converter when you need an uncompressed 16-bit PCM master. Files are limited to 200 MB total and 20 minutes; mobile devices use an 80 MB limit. Reverse, GIF and boomerang tools have lower duration caps.

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