Audio Converter
Upload any audio file and convert it to your desired format instantly, without uploading to a server.
All processing happens in your browser — your files are never uploaded.
How to use Audio Converter
The Audio Converter transforms audio files between the most popular formats — MP3, AAC, OGG, FLAC, WAV and OPUS — directly in your browser with no server uploads. Use it to convert lossless FLAC recordings to streaming-friendly AAC, re-encode voice memos to OGG for web embedding, or produce uncompressed WAV files for audio editing workflows.
- Click "Select Audio" and open the source audio file (MP3, WAV, FLAC, AAC, OGG or OPUS).
- Choose the output format from the format selector.
- For lossy output formats, set the quality or target bit rate.
- Optionally adjust the sample rate and number of channels (mono/stereo).
- Click Convert and download the output file when the conversion finishes.
Your data never leaves your device — 100% private processing.
Choosing the right audio format
Different audio formats suit different use cases. MP3 is the most universally supported lossy format and is fine for music streaming and podcasts. AAC (used in M4A files) offers better quality than MP3 at the same bit rate and is preferred on Apple platforms and YouTube. OGG Vorbis is an open-source lossy format used heavily in games and web audio. FLAC is a lossless format that preserves every detail of the original recording, making it ideal for archiving. WAV is uncompressed PCM — the largest but most editing-friendly format. OPUS is a modern low-latency codec excellent for voice calls and web streaming.
| Format | Type | Best for |
|---|---|---|
| MP3 | Lossy | Universal music playback |
| AAC / M4A | Lossy | Apple, YouTube, high-quality streaming |
| OGG Vorbis | Lossy | Web audio, games |
| OPUS | Lossy | Voice calls, low-latency streaming |
| FLAC | Lossless | Archiving, audio editing |
| WAV | Uncompressed | DAW import, professional editing |
Sample rate and channel considerations
Sample rate defines how many audio samples are captured per second. 44.1 kHz is the CD standard and is suitable for music. 48 kHz is the broadcast/video standard and is used in most video production. Reducing a stereo file to mono halves the file size and is appropriate for voice-only content such as podcasts. Converting from a lower sample rate to a higher one does not add quality — it only increases file size — so always match or reduce the output sample rate to the source.
Glossary
- Sample rate
- The number of audio samples captured per second, measured in Hz or kHz; CD quality is 44,100 Hz (44.1 kHz).
- Bit depth
- The number of bits used for each audio sample; 16-bit is CD standard, 24-bit is used in professional recording.
- Lossless compression
- Compression that reduces file size without discarding any audio data, allowing perfect reconstruction of the original.
- Transcoding
- Converting a media file from one codec or format to another, which involves decoding and re-encoding the data.
- Mono vs stereo
- Mono audio uses one channel; stereo uses two (left and right). Mono files are half the size of equivalent stereo files.
Related reading
Frequently Asked Questions
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Why use Audio Converter?
- Powered by FFmpeg running directly in the browser — no upload needed
- Supports all major video and audio formats
- Lossless and lossy options for quality control
- Export results immediately without waiting for cloud processing
Common use cases
- Compress a video before uploading to Google Drive
- Extract audio from a video lecture for easy listening
- Trim a video clip for a social media reel
- Convert MOV files to MP4 for cross-platform compatibility
- Merge two audio tracks for a podcast episode
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