Audio Converter

About Audio Converter

This online audio converter allows you to convert audio files between different formats directly in your browser. Supports MP3, WAV, OGG, AAC, FLAC, and OPUS formats with adjustable quality settings. All processing happens locally for complete privacy. See also our Audio Normalizer and Audio Recorder.

How do I convert MP3 to WAV (or any other format) with this tool?

Drag your source file into the upload area or click to browse. Pick the target format from the dropdown (MP3, WAV, FLAC, OGG, AAC, M4A, Opus, WebM, AIFF, etc.), choose a bitrate or quality preset if it applies, then click Convert. Conversion happens entirely in your browser using the Web Audio API and FFmpeg.wasm, so your files never leave your device. Once finished, a download button appears with the converted file. Batch conversion is supported by uploading multiple files at once; each output is named with the new extension. Typical conversion takes 2 to 10 seconds for a 4-minute song on a modern laptop.

Which audio formats does this converter support for input and output?

Input supports virtually any browser-decodable container: MP3, WAV, FLAC, OGG Vorbis, AAC/M4A, Opus, WebM audio, AIFF, AMR, and even raw PCM. Output covers the same set plus several legacy formats. MP3 remains the most compatible lossy choice; AAC offers better quality at the same bitrate (used by YouTube and Apple Music); Opus is the most efficient modern codec, ideal for voice and low bitrates; FLAC and WAV are lossless. If you need a format not listed, choose WAV — every audio app on Earth reads it, and you can re-encode later without further quality loss from the WAV master.

Will converting from MP3 to WAV improve the audio quality?

No. Converting a lossy file (MP3, AAC, Opus) to a lossless format (WAV, FLAC) does not restore the data that was thrown away during the original encoding. The WAV will be larger — typically 10 to 15 times the MP3 size — but it will sound identical to the source MP3, not better. The only reasons to do this conversion are technical compatibility (a video editor or DAW that refuses MP3 input) or to avoid a second lossy generation when you re-edit and re-export. For true audio quality improvement, you must re-rip or re-record from the original lossless source.

What bitrate should I pick — 128, 192, 256, or 320 kbps?

For MP3, 128 kbps is acceptable for spoken word and old radio, 192 kbps is the practical floor for music, 256 kbps is transparent for most listeners on most material, and 320 kbps (the MP3 maximum) is the safest archival choice when storage allows. AAC delivers similar perceived quality one tier lower — AAC at 192 kbps roughly matches MP3 at 256. Opus is even more efficient: 96 kbps Opus rivals 192 kbps MP3 on music and 64 kbps Opus is excellent for podcasts. Variable bitrate (VBR) usually beats constant bitrate (CBR) at the same average size because the encoder spends bits where they matter.

What sample rate should I use: 44.1 kHz, 48 kHz, or higher?

44.1 kHz is the CD standard and the default for music distribution (Spotify, Apple Music, MP3s). 48 kHz is the standard for video and broadcast (YouTube, film, professional cameras) — pick it whenever your audio will be muxed into video so the player avoids on-the-fly resampling. 88.2 and 96 kHz are useful only during production for processing headroom; the final delivery should be downsampled because higher rates inflate file size without any audible benefit at the consumption stage. Avoid converting between 44.1 and 48 kHz repeatedly — every resample introduces tiny errors. Pick the rate that matches your delivery target and stay there.

What is the difference between lossy (MP3, AAC, Opus) and lossless (FLAC, WAV, ALAC) codecs?

Lossy codecs use psychoacoustic models to discard sound information they predict you will not hear — masked frequencies, signals below the hearing threshold, joint-stereo redundancy — achieving 5 to 20 times size reduction at the cost of permanent data loss. Lossless codecs (FLAC, WAV, ALAC, APE) preserve the exact PCM samples and achieve only about 30 to 60 percent compression, but can be decoded back bit-for-bit to the original. Use lossy for streaming, mobile, and final distribution where file size matters. Use lossless for archival masters, audiophile listening, mastering chains, and any file you may want to re-encode in the future without compounding quality loss.

How can I convert audio without re-encoding to avoid generational quality loss?

Pure container-swap is possible only when the source and target use the same codec — for example, extracting the AAC stream from an MP4 video into an M4A audio file, or moving an MP3 stream into a different container. This tool performs full decode and re-encode for every format change, which is necessary when you cross codecs (MP3 to AAC, AAC to Opus, etc.). To minimize generational loss, always re-encode from the highest-quality source you have, never from an intermediate lossy export, and use VBR with a high quality target. If you plan multiple edit-export cycles, work in WAV or FLAC and encode to a lossy delivery format only at the very end.

Why does Opus often sound better than MP3 at half the bitrate, and when should I still pick MP3?

Opus, developed by the IETF and used by WhatsApp, Discord, and YouTube, combines the SILK voice codec with the CELT music codec and uses modern techniques — temporal noise shaping, range coding, frame sizes from 2.5 to 60 ms — that simply did not exist when MP3 was finalized in 1993. The result is roughly 2x the compression efficiency: 96 kbps Opus is indistinguishable from 192 kbps MP3 in double-blind listening tests (Hydrogenaudio public tests). Pick MP3 only when you need universal hardware compatibility — older cars, cheap MP3 players, embedded devices, voicemail systems — that do not implement Opus. For anything web-based, mobile, or modern, Opus is the better choice at any bitrate from 24 to 256 kbps.