Convert Any Audio File Instantly, in Your Browser

Upload your file into the converter above, choose MP3 as the output format, and select Convert. Processing happens in your browser, and a download link appears when it's ready, no account or payment required.

Yes. The conversion engine runs locally in your browser using WebAssembly, so your audio file is never uploaded to a server, nothing leaves your device at any point.

No. Files exist only in your browser's memory during conversion and are discarded when you close or refresh the page. Nothing is written to any server or database.

You can bring in most common formats as a source, including MP3, WAV, FLAC, OGG, AAC, M4A, OPUS, WMA and AIFF, and export to MP3, WAV, OGG, FLAC, M4A, OPUS or WMA.

It keeps your files private, removes upload wait time, and means the tool works without anyone else seeing your data. It's powered by a WebAssembly build of FFmpeg running entirely on your device.

There's no artificial limit, but very large files depend on your device's available memory and processing power. Most music-length files convert in seconds to a couple of minutes.

Yes. Select or drag in several files at once, choose one output format and set of options, and select Convert to batch process all of them. Each file shows its own progress, and once finished you can download files individually or grab everything together as a single zip.

The converter can read most common audio containers and codecs as input, including MP3, WAV, FLAC, OGG (Vorbis), AAC, M4A, OPUS, WMA, AIFF, AMR and AC3. If your file plays in a standard media player, it will almost always load here without issue.

You can export to MP3, WAV, AIFF, OGG, FLAC, M4A (AAC or ALAC), AAC, OPUS or WMA. Each output format has its own set of quality, bit depth or encoding options once selected.

MP3 output can be encoded at 96 kbps, 128 kbps, 192 kbps, 256 kbps or 320 kbps using the LAME encoder. 128 kbps is a common baseline for speech and casual listening, 192 to 256 kbps suits most music, and 320 kbps is the highest quality MP3 allows.

WAV output can be encoded as 8-bit, 16-bit, 24-bit or 32-bit linear PCM, as well as 8-bit u-law and 8-bit A-law companded encoding for telephony use. 16-bit PCM at 44,100 Hz is the CD-quality standard; higher bit depths suit studio and archival work.

Both are uncompressed PCM audio with essentially identical quality; the difference is container format and platform convention. WAV is more common on Windows and in general use, while AIFF is the standard uncompressed format on Mac and in professional audio tools like Logic Pro and Pro Tools. AIFF output supports the same 8-bit, 16-bit, 24-bit and 32-bit PCM options as WAV.

AAC is a lossy compressed audio codec, commonly used at 96 to 320 kbps. M4A is a container format that most often holds AAC audio, which is why "M4A" and "AAC" are frequently used interchangeably. ALAC (Apple Lossless) is a different, lossless codec that's also packaged in an M4A file, offering CD-quality audio at a larger file size than AAC. This converter offers a plain AAC output for a bare .aac file, and an M4A output where you can choose AAC at various bitrates or lossless ALAC.

You can set the output sample rate to 8,000 Hz, 16,000 Hz, 22,050 Hz, 32,000 Hz, 44,100 Hz or 48,000 Hz, or leave it unchanged from the source file. 8,000 Hz and 16,000 Hz are typical for phone and voice recordings, 44,100 Hz is CD quality, and 48,000 Hz is standard for video and studio production.

FLAC lets you choose a compression level from 0 (fastest, larger file) to 8 (smallest, slowest), with no loss of audio quality either way since FLAC is lossless. OGG Vorbis uses a quality scale from Q2 (about 96 kbps) up to Q10 (maximum). OPUS uses bitrate settings from 32 kbps, suited to voice, up to 192 kbps for high-quality audio.

Yes. A channel option lets you set the output to mono or stereo regardless of the source file's original channel layout, alongside the bitrate, encoding and sample rate options for each format.

Upload your file, choose WAV as the output format, set the encoding to 16-bit PCM (or u-law / A-law, depending on your system), set the sample rate to 8,000 Hz, and set channels to Mono. This produces the classic narrowband WAV format expected by most phone systems.

u-law, also written mu-law or μ-law, is an 8-bit companded audio encoding used in North American and Japanese digital telephony. It's part of the G.711 standard and is a common requirement for call recordings, IVR prompts and VoIP systems in those regions.

A-law is an 8-bit companded audio encoding used in telephony systems across Europe and most of the rest of the world. Like u-law, it's defined by the G.711 standard, and it's a frequent requirement for PBX prompts and call recordings outside North America and Japan.

Both are 8-bit companded encodings defined by the G.711 standard and serve the same purpose, compressing 16-bit audio down to 8 bits for narrowband telephony. The difference is the companding curve each uses and which region's phone networks adopted it: u-law in North America and Japan, A-law almost everywhere else. Using the wrong one for your system can produce distorted or noisy audio.

Most IVR, PBX and VoIP platforms, including systems built on Asterisk or FreePBX, expect 8,000 Hz mono WAV files encoded in either u-law or A-law PCM, sometimes alongside plain 16-bit linear PCM at the same rate. Check your specific platform's documentation for which one it expects.

Select WAV as the output format, choose u-law or A-law as the encoding, set the sample rate to 8,000 Hz and channels to Mono, then convert. The resulting file matches the G.711 format these platforms use for call prompts, voicemail greetings and recordings.