DevToolKit

Audio Converter

Convert audio between MP3, WAV, OGG, M4A, FLAC, Opus, and WebM with bitrate, sample rate, and channel controls — runs entirely in your browser.

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Drop an audio file or click to browse

MP3, WAV, OGG, M4A, AAC, FLAC, Opus, WMA, AIFF, WebM — processed locally, never uploaded

Client-Side Audio Conversion

Conversion runs in your browser with a WebAssembly build of FFmpeg — your audio is never uploaded. The first conversion downloads a ~32 MB engine once; later conversions start instantly. Output keeps the original filename with a new extension.

Processed locally
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How to Use

Convert an audio file in three steps:

  1. Drop your audio file — Any of MP3, WAV, OGG, M4A, AAC, FLAC, Opus, WMA, AIFF, or WebM is accepted. The tool inspects the file's binary signature (magic bytes), so a mislabeled extension won't fool it. Nothing is uploaded — the file is read locally.
  2. Choose the output settings — Pick a target format, then tune bitrate for lossy outputs (128-320 kbps for MP3/AAC/Vorbis, 96-192 kbps for Opus), sample rate (keep source, 44.1 kHz, or 48 kHz), and channels (keep, mono, or stereo). The panel shows a size estimate before you convert.
  3. Convert and download — The first run downloads a ~32 MB WebAssembly engine once; a progress bar tracks the transcode. The output keeps your filename with the new extension (e.g. interview.mp3).

About This Tool

Codecs vs. Containers

An audio "format" is really two things: a codec that compresses the raw PCM samples, and a container that wraps the compressed stream with metadata. MP3, AAC, FLAC, Opus, and Vorbis are codecs; M4A, OGG, WebM, and WAV are containers. That is why this tool offers "M4A (AAC)" — the AAC codec inside an M4A container — and "Opus" packaged in an OGG container. When you convert, FFmpeg decodes the source stream to PCM and re-encodes it with the target codec: LAME for MP3, the native AAC encoder for M4A, libvorbis for OGG, libopus for Opus and WebM, and raw PCM for WAV.

Conversion happens inside a WebAssembly build of FFmpeg running on your device. WebM and MP4-family inputs may contain a video track alongside the audio — the converter drops non-audio streams automatically, so you can also use it to extract the sound from a WebM recording.

Bitrate, Quality, and Generation Loss

Bitrate is the budget a lossy encoder spends per second of audio. At 128 kbps an MP3 encoder must discard detail aggressively; at 320 kbps the result is perceptually transparent for most listeners. Codecs differ in efficiency: Opus achieves at 96-128 kbps what MP3 needs 192+ kbps for, which is why Opus dominates voice recording and streaming. FLAC and WAV are lossless — FLAC compresses PCM by roughly 40-60% with zero quality loss, while WAV stores samples uncompressed.

One caveat: converting between two lossy formats re-quantizes already-discarded detail, a phenomenon called generation loss. An MP3-to-Opus transcode can never recover what the MP3 encoder threw away, and it adds new artifacts on top. If quality matters, keep a lossless master (FLAC or WAV) and generate lossy copies from it rather than from each other. To cut audio before converting, use the Audio Trimmer, which exports lossless WAV that converts cleanly here.

Sample Rate and Channels

The sample rate caps the highest frequency the file can represent: 44.1 kHz covers the full 20 kHz hearing range (CD standard), while 48 kHz is the video/broadcast standard and pairs well with AAC and Opus. Upsampling never adds detail — keeping the source rate is usually right. Forcing mono halves the channel count, which roughly halves the bitrate needed for voice recordings where stereo carries no real spatial information.

Why Use This Tool

Why Convert Audio Locally?

Server-side converters upload your file, transcode it in a queue, and hand back a download link — slow for large files and a poor fit for unreleased music, interviews, or voice notes. This tool runs FFmpeg compiled to WebAssembly entirely in your browser tab: no upload wait, no file-size caps imposed by a server, and no copy of your audio on someone else's disk. After the one-time ~32 MB engine download, conversions work even on a flaky connection.

Which Format Should You Pick?

  • MP3 — the universal choice: plays on every device, car, and app released in the last 25 years. Use 192 kbps for music, 320 kbps when file size doesn't matter.
  • M4A (AAC) — better quality than MP3 at the same bitrate; native to Apple devices, Android, and all modern browsers.
  • FLAC — lossless archival: about half the size of WAV with bit-identical playback. Choose it to preserve a master copy.
  • Opus — the most efficient codec here: superb for speech at 96 kbps and music at 128-160 kbps; ideal for web embedding.
  • OGG Vorbis — open and royalty-free, common in games and Linux ecosystems.
  • WAV — uncompressed PCM for editing in a DAW or feeding another tool without adding compression artifacts.
  • WebM — Opus in a WebM container, the lightest option for HTML5 audio players.

Common Use Cases

  • FLAC to MP3: shrink a lossless music library for phones and car stereos.
  • Voice memos to Opus: iPhone M4A or Android recordings become tiny Opus files that still sound clear.
  • WebM extraction: pull the audio track out of a WebM screen recording into MP3 or WAV.
  • WAV to M4A: turn huge uncompressed exports into shareable files for messaging apps.
  • WMA rescue: modernize legacy Windows Media Audio collections into MP3 or AAC that current players accept.

Privacy

The entire pipeline — format detection, decoding, re-encoding, and download — executes locally via WebAssembly. No file data crosses the network; the only download is the engine binary itself on first use. Related tools: Audio Trimmer, Screen Recorder, File Type Detector, and Base64 File Converter.

FAQ

Which formats can I convert between?
Input: MP3, WAV, OGG Vorbis, M4A, AAC, FLAC, Opus, WMA, AIFF, and WebM (the audio track is extracted automatically). Output: MP3, WAV, OGG Vorbis, M4A (AAC), FLAC, Opus, and WebM audio. The format is detected from the file's binary signature, not just its extension.
Which bitrate should I choose?
For MP3, AAC (M4A), and OGG Vorbis, 192 kbps is a good balance for music; 320 kbps is the maximum for near-transparent quality. Opus is more efficient — 96 kbps suits speech and 128-160 kbps handles music well. WAV and FLAC are lossless, so they have no bitrate setting.
Does converting audio reduce quality?
Converting from a lossy format (like MP3) to another lossy format re-compresses the audio and loses a small amount of quality — this is called generation loss. Converting to WAV or FLAC is lossless and preserves the decoded audio exactly. For archiving, choose FLAC; for maximum compatibility, choose MP3 at 192 kbps or higher.
Why does the first conversion take longer?
The tool runs a WebAssembly build of FFmpeg that is roughly 32 MB. Your browser downloads it once on the first conversion and caches it, so subsequent conversions start almost instantly. No audio data is sent over the network at any point.
Is my audio uploaded to a server?
No. Everything runs locally in your browser using FFmpeg compiled to WebAssembly. Your file is read into memory, converted on your device, and offered back as a download — it never leaves your computer.
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