Audio Converter
Convert audio between MP3, M4A/AAC, WAV, FLAC, OGG, Opus and AIFF with bitrate, sample-rate and channel options.
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Tool guide
What this tool does
Audio Converter transcodes music, voice notes and recordings between the common audio formats using FFmpeg in your browser. Set the bitrate for lossy formats, resample to 48, 44.1, 32 or 22.05 kHz, mix down to mono, and convert a whole batch at once.
Audio arrives in whatever format the recording device prefers: WhatsApp and Telegram voice notes are Opus in OGG, iPhones record M4A, studio exports are WAV or AIFF, and music collectors keep FLAC. The app, website or device you want to use next often accepts only one of them, typically MP3.
This converter runs a WebAssembly build of FFmpeg locally. It decodes almost any audio format, including AMR and WMA, then encodes with LAME for MP3, FFmpeg’s AAC encoder for M4A, libvorbis for OGG, libopus for Opus, or writes uncompressed 16-bit PCM (WAV, AIFF) or lossless FLAC. Your files are never uploaded; only the engine is fetched once and cached.
How to use Audio Converter
- Drop one or more audio files onto the upload area or click to browse.
- Choose a preset or pick the format in Convert to.
- For lossy formats choose a Bitrate: 320 or 256 kbps for music, 128 kbps for general use, 64 to 96 kbps for speech.
- Optionally change Sample rate or set Channels to Mono.
- Click Convert and watch the progress for each file.
- Listen to the results and download them individually or as a ZIP.
Tips for better results
- Convert lossless sources (WAV, FLAC) to lossy formats, not the other way round: turning an MP3 into FLAC cannot restore lost detail.
- FLAC is typically 50 to 70 percent of the size of the equivalent WAV with identical sound.
- Opus at 64 kbps sounds as good as MP3 at about 128 kbps for most speech, but some older devices cannot play it.
- CD audio is 44.1 kHz; video and most phones use 48 kHz. Keep the original rate unless a platform requires a specific one.
Common use cases
- Converting WhatsApp OGG/Opus voice notes to MP3 to play on any device.
- Turning WAV exports from a recording app into MP3 for sharing.
- Making mono, low-bitrate MP3s of podcast episodes or lectures to save space.
- Converting FLAC albums to M4A or MP3 for a phone with limited storage.
Common mistakes to avoid
- Choosing a bitrate higher than the source, which grows the file without improving it.
- Resampling to 22.05 kHz for music; it removes everything above about 11 kHz and sounds dull.
- Converting DRM-protected purchases, which cannot be decoded.
Frequently asked questions
Which input formats are supported?
Anything the bundled FFmpeg can decode, including MP3, WAV, AIFF, FLAC, M4A/AAC, OGG Vorbis, Opus, WMA, AMR and the audio inside WebM files.
What bitrate should I choose for MP3?
For music, 192 to 320 kbps is effectively transparent for most listeners. For speech, 64 to 96 kbps mono is plenty and creates much smaller files.
Is WAV better quality than MP3?
WAV is uncompressed, so it keeps every sample of the source. MP3 discards sound that is hard to hear to save space. Converting an MP3 to WAV does not add back what was removed.
Are my files uploaded?
No. Conversion runs on your device using FFmpeg compiled to WebAssembly. The engine is downloaded once from the jsDelivr CDN.
Feedback & suggestions
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