Ljudkonverterare — Gratis och privat
Konvertera ljudfiler mellan MP3, WAV, OGG, FLAC och AAC. 100% i din webbläsare.
Drop audio to convert
MP3, WAV, OGG, FLAC, AAC • Max 100MB
How It Works
This tool uses FFmpeg compiled to WebAssembly to perform professional-grade audio conversion entirely in your browser. FFmpeg is the industry-standard multimedia framework — the same engine that powers YouTube, VLC, Spotify, and virtually every professional audio/video workflow. The WebAssembly version brings this full power to your browser tab with no installation required.
Select your target format from six supported options: MP3 (universal playback), WAV (uncompressed lossless), OGG (open-source compressed), FLAC (lossless compressed), AAC (Apple ecosystem), or M4A (iTunes/Apple Music). For lossy formats, you can fine-tune the output bitrate from 64kbps (voice-quality, tiny files) to 320kbps (CD-quality, larger files) and adjust the sample rate.
The FFmpeg WASM engine (~30MB) downloads once on first use and is cached in your browser. Return visits start converting instantly with no loading delay. The engine runs in a dedicated Web Worker thread, so your browser stays responsive during conversion — even for large files.
Audio trimming is built in: set custom start and end times to extract a specific section of the audio during conversion. This eliminates the need for a separate editing tool when you only need a clip. All processing is local — your audio files never leave your device.
Features
- ✓Convert between MP3, WAV, OGG, FLAC, AAC, and M4A — all major audio formats
- ✓Adjustable bitrate (64kbps to 320kbps) and sample rate for precise quality control
- ✓FFmpeg-powered — the same engine used by professional studios and streaming platforms
- ✓Handles audio files up to 100MB with fast local processing
- ✓Built-in audio trimming with start/end time controls — no separate editor needed
- ✓Preserves ID3 tags and metadata where the target format supports it
- ✓Engine loads once (~30MB) and caches permanently for instant subsequent use
- ✓Zero uploads, zero accounts, unlimited free conversions
How to Use This Tool
- Upload your audio file by drag-and-drop or file picker. Supported input formats: MP3, WAV, OGG, FLAC, AAC, M4A.
- Select your target output format from the dropdown. For lossy formats, adjust the bitrate and sample rate to balance file size against audio quality.
- Optionally set trim points (start/end times) if you only need a section of the audio. The tool will extract and convert the specified clip.
- Click "Convert" and wait for the FFmpeg engine to process your file. Download the converted audio — ready for playback, upload, or further editing.
Perfect For
- →Podcasters converting uncompressed WAV recordings to compressed MP3 at 128-192kbps for podcast hosting platforms (Anchor, Buzzsprout, Libsyn)
- →Musicians exporting FLAC or WAV masters to AAC/M4A format for Apple Music, iTunes, and iOS device compatibility
- →Video editors converting extracted audio tracks to the format required by their NLE (Premiere Pro, DaVinci Resolve, Final Cut)
- →Voice-over artists converting studio recordings (WAV/FLAC) to compressed MP3 for client delivery via email or file sharing
- →Audiobook producers converting chapter recordings between formats for different distribution platforms (Audible, Findaway, Google Play)
- →Developers converting audio assets between formats for game engines, web applications, and mobile apps
- →Language learners converting downloaded audio lessons to formats compatible with their preferred media player or offline listening device
- →DJs converting between audio formats for compatibility with different DJ software and hardware (Serato, Rekordbox, Traktor)
Under the Hood
The converter runs FFmpeg compiled to WebAssembly (ffmpeg.wasm), bringing the full FFmpeg multimedia framework to the browser. This includes production-grade implementations of all major audio codecs: libmp3lame (MPEG-1 Layer 3 encoding), libvorbis (Ogg Vorbis encoding), the reference FLAC encoder, and the AAC encoder. Each codec is compiled from the same C source code used in desktop FFmpeg, ensuring bit-identical output quality.
The encoding pipeline mirrors what happens in professional audio software: (1) the source file is demuxed (container parsed, codec stream extracted), (2) the audio stream is decoded to raw PCM samples, (3) sample rate conversion is applied via a high-quality resampler if the target rate differs, (4) the PCM samples are encoded using the target codec at the specified bitrate, (5) the encoded stream is muxed into the target container format with appropriate metadata.
The WASM binary runs in a dedicated Web Worker thread, preventing the audio processing from blocking the browser's main UI thread. File I/O uses the Emscripten virtual filesystem — source files are written to a virtual /input path, FFmpeg processes them, and the output is read from a virtual /output path. All of this happens in browser memory with zero network activity.