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Input: MP3, WAV, FLAC, OGG, M4A, AAC, AIFF, OPUS, WMA, WebM, and MP4 (audio extracted from video). Output: MP3 at 128, 192, 256, or 320 kbps, or lossless WAV (16-bit PCM stereo).

The converter uses lamejs, a JavaScript port of the LAME MP3 encoder. Audio is chunked into 1152-sample frames and encoded at your chosen bitrate. Higher bitrates (256-320 kbps) preserve more detail; 128 kbps is smaller but loses high-frequency content.

Yes. MP4, WebM, and MKV files are decoded by extracting the audio track through a video element. The audio is then encoded to your chosen output format. No video processing happens — only the audio stream is used.

Converting to MP3 is lossy — some audio data is permanently discarded. Converting to WAV is lossless but produces larger files. Converting from one lossy format to another (e.g., OGG to MP3) compounds quality loss. For best results, start from a lossless source.

Up to 30 minutes per file. If the input sample rate differs from 44.1 kHz or 48 kHz, the audio is automatically resampled to 44.1 kHz for MP3 compatibility.

No. All decoding and encoding runs entirely on your device using the Web Audio API and lamejs. Your files never leave your machine — there are no server costs, which is why it's free with no limits.