Convert audio files between WAV, MP3, OGG, AAC, M4A, FLAC formats online. Adjust bitrate and quality settings. Free browser-based conversion with no file uploads to servers.
Audio formats split into two families: lossy and lossless. Lossy codecs (MP3, AAC, OGG Vorbis, Opus) use psychoacoustic models to throw away sound data that the ear is unlikely to notice, masking effects from louder nearby frequencies, content above the upper limit of most adult hearing around 16-18 kHz, and subtle stereo details. A 3-minute song at 320 kbps MP3 is about 7 MB; the same song as uncompressed 44.1 kHz 16-bit stereo WAV is 32 MB. That roughly 4.5x ratio is the lossy compression at work, and for most content at most bitrates it is audibly transparent. Lossless codecs (FLAC, ALAC, WAV, AIFF) keep every sample bit-exact. FLAC typically compresses WAV by 40-50% through entropy coding without discarding any audio data, so a song that is 32 MB as WAV is 17-19 MB as FLAC. WAV is uncompressed PCM (the raw sample stream), with larger files but universally decodable at zero CPU cost. The practical matrix: use MP3 or AAC at 192-256 kbps for casual distribution, Opus for web streaming (better quality per bit than MP3), FLAC for archival of lossless masters, and WAV only when a downstream tool specifically requires uncompressed input.
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A podcast host requires MP3 but your editor exported a 60 MB WAV.
Input
episode.wav (60 MB, 48 kHz) → MP3 192 kbps
Output
episode.mp3 ≈ 8.5 MB, ~86% smaller, transparent at 192 kbps speech
WAV is uncompressed; MP3 at a sensible bitrate keeps voice indistinguishable while shrinking the file ~7×, which matters for upload limits and listener bandwidth. The conversion runs locally so the raw audio stays private.
Audio formats split into two families: lossy and lossless. Lossy codecs (MP3, AAC, OGG Vorbis, Opus) use psychoacoustic models to throw away sound data that the ear is unlikely to notice, masking effects from louder nearby frequencies, content above the upper limit of most adult hearing around 16-18 kHz, and subtle stereo details. A 3-minute song at 320 kbps MP3 is about 7 MB; the same song as uncompressed 44.1 kHz 16-bit stereo WAV is 32 MB. That roughly 4.5x ratio is the lossy compression at work, and for most content at most bitrates it is audibly transparent. Lossless codecs (FLAC, ALAC, WAV, AIFF) keep every sample bit-exact. FLAC typically compresses WAV by 40-50% through entropy coding without discarding any audio data, so a song that is 32 MB as WAV is 17-19 MB as FLAC. WAV is uncompressed PCM (the raw sample stream), with larger files but universally decodable at zero CPU cost. The practical matrix: use MP3 or AAC at 192-256 kbps for casual distribution, Opus for web streaming (better quality per bit than MP3), FLAC for archival of lossless masters, and WAV only when a downstream tool specifically requires uncompressed input.
Convert files to a format your player, phone, or car stereo actually supports.
Turn large WAV recordings into compact MP3s for email or cloud storage.
Convert lossy files to FLAC or WAV when you want a lossless master copy of new recordings.
Bitrate is the main quality knob on lossy codecs. MP3 at 128 kbps is the floor where most listeners can hear compression artifacts (a slight "swirl" in cymbals, reduced high-frequency detail); 192 kbps is where most listeners stop hearing artifacts in casual listening; 256 kbps is transparent for nearly everyone on nearly all content; 320 kbps is the MP3 format's ceiling and the point of diminishing returns. AAC produces noticeably better quality than MP3 at equivalent bitrates: AAC at 128 kbps is roughly as good as MP3 at 192 kbps, which is why streaming services largely moved to AAC. Opus is the modern champion for new applications: at 96 kbps Opus matches MP3 at 192 kbps, and it handles both speech and music in one codec. For speech-only content (podcasts, voicemail, audiobooks) you can go much lower, with Opus at 24-32 kbps usable for speech.
Converting between formats is always a decode-re-encode operation except when both formats are lossless: FLAC to WAV is a decode-to-PCM operation with no information loss, and WAV to FLAC is a lossless compression pass. Everything else accumulates some degradation. Lossy-to-lossy conversion (MP3 to AAC, for example) decodes the MP3 to PCM, then encodes to AAC, so you lose whatever the AAC encoder would throw away on top of what the MP3 encoder already threw away. For this reason, keep lossless masters when possible and convert from the master for each distribution target, rather than chaining lossy conversions.
Sample rate and bit depth are separate parameters from bitrate. 44.1 kHz 16-bit is the CD standard and covers the full audible range (Nyquist gives 22.05 kHz bandwidth, comfortably exceeding the 20 kHz upper limit of human hearing). 48 kHz is the video and broadcast standard; convert audio for video work at 48 kHz to avoid resampling at the NLE stage. 24-bit depth matters for recording and mixing because it provides extra headroom and a lower noise floor, but for delivery 16-bit is indistinguishable from 24-bit on any consumer playback system. Higher rates (96 kHz, 192 kHz) are marketing rather than audibility: they make sense for pro audio archives but are wasted bits for casual listening.
No. Once audio data has been discarded by lossy compression, converting to a lossless format cannot restore it.
For music, 192-256 kbps offers a good quality-to-size balance. For speech, 128 kbps is usually sufficient.
Audio is decoded and processed locally with the Web Audio API. Your files are never uploaded to a server.