S = t × f × C × b / 8
Audio fileDrop an audio file here or choose an option below.
Audio waveform
Start your audio project
Drop an audio file here or choose an option below.
Recording…
00:00
00:0000:0000:00
Drag to browse the audio
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How does this tool work?
Convert audio files locally to MP3, WAV, FLAC, OGG, M4A or Opus. The engine uses Web Audio for WAV when possible and loads FFmpeg/WASM only when the requested format or decoding requires it.
The file remains in your browser. For WAV, Bethemesh prefers Web Audio decoding followed by local PCM encoding. For MP3, FLAC, OGG, M4A and Opus, FFmpeg/WASM is loaded on demand in a Worker and performs the conversion without uploading media to a server. The first FFmpeg-powered conversion can therefore take a little longer to start. MP3 is selected initially. Unlike the specialist pages, this general converter offers all six outputs. MP3 and AAC in M4A use 192 kbit/s; Opus targets 128 kbit/s; OGG uses Vorbis quality 5. For a custom bitrate, use audio compression. WAV output is 16-bit PCM: native Web Audio decoding may resample to the browser context rate and does not copy metadata. FFmpeg may retain compatible tags, but artwork, chapters and format-specific fields are not guaranteed. Changing formats cannot restore an already degraded recording. Opus uses FFmpeg’s native CELT encoder at 128 kbit/s instead of libopus, which can crash in the pinned WASM core. This profile does not include SILK speech mode.
How to use this tool
Choose a file
Import an audio file supported by the browser or FFmpeg/WASM.
Choose the format
Select MP3, WAV, FLAC, OGG, M4A or Opus for your target use.
Convert and download
The file is converted locally and then downloaded to your device.
Audio converter — Calculation reference
S ≈ 1000 × R × t / 8
- S
- PCM payload in bytes, excluding headers
- t
- Duration in seconds
- f
- Sample rate in Hz
- C
- Channel count, dimensionless
- b
- Bits per sample
- R
- Bitrate in decimal kbit/s
Use cases
MP3 for sharing
Convert audio to MP3 for a widely compatible format with a practical file size.
Lossless FLAC
Create a FLAC file when you want to preserve audio quality without destructive compression.
Batch processing
Reuse the same conversion in Workspace across multiple audio files with one common output format.
Examples
A two-minute MP3 copy
192,000 × 120 ÷ 8 estimates the payload before tags and container overhead. Duration remains approximately the same, subject to encoding precision.
Input
120 s, MP3 output at 192 kbit/sResult
About 2,880,000 audio bytesStereo WAV for editing
PCM storage explains the much larger size. This sample rate is an example assumption, not a promise that the source rate will be preserved.
Input
120 s, 48,000 Hz, 2 channels, 16-bit PCMResult
23,040,000 PCM bytesTips and best practices
- MP3 is a useful compromise between size, quality and compatibility.
- FLAC preserves audio data losslessly but usually creates larger files.
- The first MP3, FLAC or OGG operation must load the FFmpeg/WASM core before conversion.
Sources and references
- FFmpeg project, FFmpeg Formats Documentation (opens in a new tab)Page in English
- J.-M. Valin, K. Vos, T. Terriberry, RFC 6716 — Definition of the Opus Audio Codec (opens in a new tab), RFC Editor, 2012Page in English
- Microsoft, WAVEFORMATEX structure (mmeapi.h) (opens in a new tab)Page in English
- FFmpeg project, FFmpeg Codecs Documentation (opens in a new tab)Page in English
- IETF, RFC 9639 — Free Lossless Audio Codec (FLAC) (opens in a new tab), 2024Page in English
- Xiph.Org Foundation, Vorbis I specification (opens in a new tab), 2020Page in English
Frequently asked questions
Is the audio file uploaded to a server?
No. Audio content remains processed locally in your browser.
Why can the first MP3 or FLAC conversion take longer?
FFmpeg/WASM is deliberately loaded on demand so the rest of the site stays lightweight.
Which output formats are available?
The converter offers MP3, WAV PCM, FLAC, OGG Vorbis, M4A (AAC) and Opus.
Can I set the MP3 bitrate here?
No. This conversion uses 192 kbit/s. Audio compression exposes an adjustable bitrate; it shares the encoding engine but offers different controls.
Is converting to WAV an exact copy of the original file?
No. It creates a new 16-bit PCM stream. Decoding may resample the signal and not all metadata survives. Keep the original as your reference.
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