Free Audio Converter Online

How to Convert Audio

1. Select your target format (MP3, WAV, OGG, or AAC) above.
2. For MP3 and AAC, adjust the bitrate (128-320 kbps). Higher bitrate = better quality but larger file.
3. Drop or select one or more audio files.
4. Click 'Convert All' to start the conversion.
5. Download individual files or all at once as a ZIP.

Audio Format Guide

• MP3 · Most widely supported format. Good compression and quality. Use for general purposes and portability.
• WAV · Lossless format. No quality loss but larger file sizes. Best for professional audio editing and archiving.
• OGG · Open-source format with excellent compression. Smaller than MP3 at same quality. Good for web and gaming.
• AAC · Modern format used by iTunes and Apple devices. Better quality than MP3 at same bitrate. More efficient compression.

When to Use Each Format

• Convert to MP3 · Maximum compatibility. Use for music players, YouTube, and devices from 2000s-2010s.
• Convert to WAV · For professional audio work, archiving, or when you need lossless quality for further editing.
• Convert to OGG · For websites, game development, and scenarios where file size matters more than maximum compatibility.
• Convert to AAC · For Apple devices, iTunes libraries, and modern apps where AAC is preferred.

Bitrate Explained

• 128 kbps · Adequate for speech and podcasts. Noticeable quality loss on music.
• 192 kbps · Good for general listening. Suitable for most streaming services.
• 256 kbps · Very high quality. Recommended for music. Imperceptible difference from lossless.
• 320 kbps · Maximum MP3/AAC quality. Use when quality is critical and file size is not a concern.

Frequently Asked Questions

What happens to audio quality when converting?

Converting to WAV is lossless (no quality loss). Converting to MP3, AAC, or OGG applies lossy compression. At 256-320 kbps, the quality difference is imperceptible to most listeners. Lower bitrates trade audio fidelity for smaller file sizes.

Can I convert multiple audio files at once?

Yes! Add multiple files and they will be converted sequentially. You can download them individually or all at once as a ZIP file for convenience.

Does the conversion preserve metadata like artist and title?

The conversion preserves the audio content. However, metadata (ID3 tags) may not be transferred. You may need to re-tag your files with artist, album, and title information in your media player.

Are my audio files uploaded to a server?

No. All conversion happens in your browser using FFmpeg.wasm. Your audio files never leave your device.

What is an audio converter?

An audio converter takes a sound file in one format (say, a high-quality WAV recording) and rewrites it as another format (say, a smaller MP3 for sharing). The audio content stays the same; only the way it is encoded and packaged changes. Different formats make different trade-offs between file size, audio quality, and which devices can play them. A 5-minute song might be 50 MB as uncompressed WAV, 5 MB as 128 kbps MP3, or 3 MB as 96 kbps Opus, all carrying the same musical content with different fidelity.

The two main format categories are lossless (WAV, FLAC, ALAC) and lossy (MP3, AAC, Opus, Vorbis). Lossless preserves every sample exactly and can be decoded back to bit-identical audio; lossy throws away information that the human ear is unlikely to notice in exchange for much smaller files. For music or speech destined for normal listening, lossy is almost always the right choice. For mastering, archiving, or further editing, lossless preserves headroom for later processing.

This tool uses FFmpeg.wasm, a WebAssembly port of the venerable FFmpeg audio/video processing library. Everything runs in your browser: the file you drop in is decoded, re-encoded in the target format, and offered as a download, all on your machine. Nothing is uploaded. The tool supports the formats most people actually need (MP3, WAV, OGG, M4A/AAC, FLAC, Opus, WebM) with sensible default settings, and falls back gracefully on older formats that FFmpeg knows about (WMA, AU, AIFF, AMR and dozens of others).

What is inside the tool

A drop zone at the top accepts an audio file from your disk. Drag and drop, or click to pick. The tool reads the file's duration, sample rate, channel count and current format as soon as it loads, so you can confirm you picked the right file. There is no upload step; the file is loaded directly into browser memory.

The output-format dropdown lets you pick the target format. For most cases the defaults are fine: MP3 for compatibility, AAC for Apple devices, Opus for the highest compression-to-quality ratio, WAV for editing in a DAW, FLAC for lossless archiving. Quality presets handle the bitrate selection for you (Low, Medium, High, Best); for music, choose at least Medium (128 kbps for MP3, 96 kbps for Opus). The Convert button kicks off the FFmpeg encoding pipeline.

Conversion runs entirely in your browser via the WebAssembly build of FFmpeg. The progress bar updates as the file is processed. A 5-minute MP3 typically converts in 3 to 10 seconds on a modern machine; a 30-minute podcast in 30 to 60 seconds. Hour-long lossless files take longer because the input is bigger. When done, an audio preview lets you confirm the result before downloading, and the download button saves the new file to your default downloads folder.

History and background

From PCM to digital audio (1937 onwards)

Pulse-code modulation (PCM) was invented in 1937 by Alec Reeves at the British arm of ITT. It is the foundational technique of digital audio: sample the analog waveform at regular intervals (44,100 times per second for CD audio), quantize each sample to a number, store the numbers. WAV (Microsoft, 1991) is essentially a thin wrapper around PCM samples. Every modern audio format starts from PCM and then applies compression or other processing on top.

MP3 ships and wins (1993)

MP3 (MPEG-1 Audio Layer III) was standardized in 1993 by the Fraunhofer Institute, using psychoacoustic models to discard sound that the human ear masks naturally. It was the first format to make high-quality audio compression practical on consumer hardware. The combination of small file size, decent quality at 128 kbps, and (eventually) free decoder licenses made it the dominant audio format for two decades. Napster (1999) and the iPod (2001) cemented its position.

AAC takes over for Apple and broadcast (1997 onwards)

AAC (Advanced Audio Coding) was finalized in 1997 as the successor to MP3 in the MPEG-2 and MPEG-4 standards. It produces better quality at the same bitrate, especially at low bitrates. Apple chose AAC as the default format for the iTunes Store (2003) and the iPod, which gave it mainstream adoption. AAC is also the standard for digital broadcast radio (DAB+ in Europe, HD Radio in the US) and YouTube. Files are usually named .m4a (audio-only) or .mp4 (when combined with video).

FFmpeg becomes the universal Swiss army knife (2000)

FFmpeg was started by Fabrice Bellard in 2000 as an open-source command-line tool for converting between audio and video formats. Over 25 years it has accumulated decoders and encoders for hundreds of formats, becoming the de facto reference implementation for media processing. Nearly every audio/video tool you have ever used (YouTube, VLC, Audacity, OBS, Premiere Pro, every streaming platform) relies on FFmpeg internally for some part of its pipeline. This tool runs FFmpeg in your browser via WebAssembly.

Opus enables WebRTC and modern streaming (2012)

Opus was standardized as RFC 6716 in September 2012, designed jointly by Xiph.Org and Skype's audio team to handle both speech (like Skype) and music (like Vorbis) in a single codec. It is the mandatory codec for WebRTC (real-time browser audio), the default for Discord and Zoom voice, and supported by YouTube, Spotify, Whatsapp, Telegram and every modern messaging platform. At 64 kbps Opus matches MP3 at 128 kbps for music; at 16 kbps it produces intelligible speech. It is also royalty-free.

FFmpeg.wasm brings conversion to the browser (2019)

FFmpeg.wasm was released in 2019, compiling the full FFmpeg toolchain to WebAssembly. Before then, audio conversion in a browser required uploading the file to a server that ran FFmpeg server-side. With FFmpeg.wasm, the conversion runs in the browser at near-native speed (within roughly 2x of native FFmpeg). This is the foundation that makes private, client-side audio conversion tools like this one possible. The trade-off is download size (the WebAssembly bundle is several megabytes) and slower performance than native FFmpeg, but the privacy gain is usually worth it.

Practical workflows

Converting WAV recordings to MP3 for sharing

You recorded a meeting, voice memo, or rehearsal in WAV at 44.1 kHz stereo, which is 10 MB per minute. To email it or upload to Drive, convert to 128 kbps MP3 to shrink to roughly 1 MB per minute. For voice-only content, 64 kbps Opus produces an even smaller file with comparable intelligibility, but pick MP3 if you need broad compatibility. Keep the original WAV if you might re-edit later.

Converting MP3 back to WAV for editing in a DAW

Digital audio workstations (Audacity, Logic, Pro Tools, Reaper) prefer to edit lossless audio so cuts, fades and effects do not compound encoding artifacts. Convert your MP3s to WAV before importing. Note that converting lossy back to lossless does not restore the lost quality; the WAV is just a lossless container holding the already-degraded audio. Use the original lossless if you have it.

Converting Apple Voice Memos and iTunes purchases to MP3

Apple Voice Memos save as M4A (AAC inside an MP4 container) and older iTunes purchases came as M4P (DRM-wrapped AAC) or M4A. Modern iTunes (Apple Music) downloads are DRM-free M4A. Convert to MP3 for non-Apple devices, players that do not support AAC, or for embedding in a website or game that needs MP3. For DRM-protected M4P files this tool cannot convert them; remove DRM at the source first.

Preparing podcast episodes

Podcast hosting platforms (Libsyn, Buzzsprout, Anchor) standardize on MP3 at 64 kbps to 128 kbps mono for spoken-word podcasts. Record in WAV in your DAW, edit, then export to WAV for archiving and convert a copy to MP3 for distribution. For interview podcasts where speech intelligibility matters more than file size, 96 kbps stereo is a sweet spot. Apple Podcasts and Spotify both accept MP3 at any standard bitrate.

Converting old audio formats for modern devices

WMA (Windows Media Audio), AU (Sun Microsystems), AIFF (Apple's pre-2000 format), AMR (mobile voice) all worked great in their day but get spotty support on modern devices. Convert them to MP3 (universal) or M4A (best for Apple ecosystem) for guaranteed playback. The conversion is one-way for lossy sources; do not expect any quality gain.

Building a lossless archive from a CD rip

For audiophiles or archivists, rip CDs to WAV (universal, large) or FLAC (compressed lossless, about 50 percent smaller than WAV with bit-identical decoding). FLAC is the standard archival format because it supports tags, multi-channel audio, and is open-source. Convert your existing WAV rips to FLAC to save disk space while keeping bit-perfect audio. You can always go back to WAV from FLAC because the decode is lossless.

Common pitfalls

Lossy-to-lossy conversion compounds quality loss

Each lossy encoding throws away some audio information. Converting MP3 to AAC to Opus to MP3 again will sound noticeably worse than the original MP3 even if every step uses the same nominal bitrate. The artifacts of one codec become input to the next, which sees them as legitimate audio and tries to preserve them. If you must convert between lossy formats, do it once at the highest bitrate that makes sense. Whenever possible, go back to the lossless original (or the source recording) before re-encoding.

Bitrate too low produces obvious artifacts

MP3 at 64 kbps stereo sounds noticeably muddy on music with cymbals, sibilant vocals or complex high frequencies. The codec runs out of bits to represent the upper frequencies cleanly. Keep music at 128 kbps or higher for MP3, 96 kbps or higher for Opus, 96 kbps or higher for AAC. For speech only (podcasts, lectures, voice memos), 64 kbps is fine because speech is much simpler signal than music. The Quality presets in this tool encode these defaults.

Sample rate mismatches degrade quality

If your source is 44.1 kHz (CD standard) and you convert to a format at 48 kHz (DVD/streaming standard), the converter has to resample, which introduces small distortions. Most converters do this well using high-quality algorithms (FFmpeg's swresample uses a polyphase filter that is essentially transparent), but cheap converters can introduce audible artifacts. When converting for playback, match the source rate if you can. When converting for production, follow the target medium's spec (44.1 kHz for CD, 48 kHz for video, 96 kHz for high-resolution audio).

Channel reduction loses spatial information

Converting stereo to mono averages the left and right channels into a single signal. Any audio that was panned hard left or hard right loses its position. Centre-panned audio (vocals, kick drums) survives fine but anything sounding spatial flattens. For podcasts and voice-only recordings, mono is appropriate and saves bandwidth. For music, keep stereo unless file size is critical. For surround-source audio (5.1 from a DVD or Atmos from a Blu-ray), downmixing to stereo loses spatial information; consider keeping the surround for home theatre use.

Metadata is often lost in conversion

MP3 stores metadata in ID3 tags, AAC in iTunes-style metadata, Vorbis in Vorbis comments, FLAC similarly. Converting between formats does not always preserve every tag: cover art may be dropped, custom tags may not have an equivalent in the target format, encoding history may be reset. If metadata is critical (large music libraries, podcast tags), use a dedicated tagging tool (MP3Tag, Mp3tag for Mac, Picard) after conversion to clean up. This tool focuses on the audio content; metadata transfer is best-effort.

Constant vs variable bit rate (CBR vs VBR) matters for some use cases

Constant bit rate (CBR) uses the same number of bits per second throughout the file, regardless of how complex the audio is at that moment. Variable bit rate (VBR) uses more bits for complex passages and fewer for simple ones, producing smaller files at the same average quality. For streaming and broadcasting CBR is required (predictable bandwidth); for files on disk VBR is better. This tool defaults to CBR for compatibility; if file size matters more than streaming, look for a VBR option in the advanced settings of dedicated tools like LAME (for MP3) or qaac (for AAC).

Privacy and data handling

The audio file you upload stays on your device throughout the conversion. FFmpeg.wasm runs the entire encoding pipeline in your browser using WebAssembly; there is no upload step and no remote processing. This matters because the audio you most want to convert privately (voice memos, meeting recordings, medical consultations, legal depositions, song demos before release) is exactly the kind of content you do not want to send to a third-party cloud converter.

Once the page is loaded, the tool works offline. You can disconnect from the internet, drop an audio file, run the conversion, and download the result without your audio ever touching another machine. Cloud-based converters (Online-Audio-Converter, Convertio, Cloudconvert) upload your file before processing, which is precisely the failure mode to avoid for confidential audio.

When not to use this tool

When you need to edit or mix the audio (use a DAW)

If you want to cut sections, adjust volume, add fades, mix multiple tracks, or apply effects, use a digital audio workstation (Audacity is free and powerful, Reaper is paid but excellent, Logic Pro and Pro Tools for professional work). A converter only re-encodes the audio; it does not edit. The DAW workflow is: open in DAW, edit, export to your target format. Converters are for the export step alone.

When you need to manage metadata at scale (use a tagger)

For batch-tagging a music library (consistent artist names, album art, track numbers), use a dedicated tagging tool: MP3Tag (Windows, free), Mp3tag for Mac, MusicBrainz Picard (cross-platform, auto-tags via the MusicBrainz database). Converters touch metadata as a side-effect but do not provide good tools for editing it. Picard in particular can auto-tag entire libraries by matching audio fingerprints, which is huge for older or unlabelled files.

When the source is DRM-protected

Old iTunes purchases (M4P), Spotify/Apple Music streams, Audible audiobooks (AAX) all carry digital rights management that this tool cannot bypass. Conversion fails or produces empty output. The legal way to deal with DRM is usually to obtain the content in a non-DRM format (re-purchase, find an open alternative, contact the rights holder for an accessibility exemption). Removing DRM is illegal in many jurisdictions under DMCA-style laws.

For real-time streaming format conversion

If you need to convert audio in real-time (broadcast streaming, live transcoding, voice-call format negotiation), use a streaming server or library: nginx-rtmp, GStreamer, Janus WebRTC server. This tool is for converting files on disk one at a time. Real-time conversion has different constraints (latency, throughput, recovery from packet loss) that file converters do not address.

More questions

MP3 vs AAC vs Opus: which should I use?

Opus is the technically best choice in 2026: highest quality at any bitrate, royalty-free, supported by every modern browser, phone and messaging app. Use Opus when you control both ends. AAC is the right choice for Apple devices, video files, and broadcast streaming; it produces noticeably better quality than MP3 at the same bitrate. MP3 remains the safest choice for maximum compatibility (every player, every car stereo since 2000, every embedded device). For new audio with no compatibility constraints, pick Opus; otherwise, pick MP3 for maximum reach or AAC for Apple-leaning audiences.

What is a good bitrate for music?

For MP3, 192 kbps is essentially transparent for most listeners on most equipment; 320 kbps is the practical maximum and rarely audibly different from lossless. For AAC, 128 kbps matches MP3 at 192 kbps for quality. For Opus, 96 kbps matches AAC at 128 kbps. The classic listening tests by Hydrogenaudio established that double-blind listeners cannot reliably distinguish lossless from 192 kbps MP3 or 128 kbps AAC on standard equipment. Higher bitrates beyond that are inaudible but waste bandwidth and storage.

Can I convert lossless to lossless without losing quality?

Yes. Converting WAV to FLAC, FLAC to WAV, FLAC to ALAC, ALAC to WAV, or any combination of lossless formats produces bit-identical audio on decode. You can prove this by decoding both back to WAV and comparing the byte streams. Lossless compression (FLAC, ALAC, APE, WavPack) reduces file size by 30 to 60 percent depending on the music, without changing the audio. You can always recover the original.

M4A and MP4 audio: same or different?

M4A is just MP4 carrying audio only, with .m4a extension instead of .mp4. The container is identical (ISO Base Media File Format, ISO 14496-12); the audio codec inside is almost always AAC, sometimes ALAC (lossless). MP4 (with video) and M4A (audio-only) are interchangeable in most software. Some old players that recognize .mp4 may not recognize .m4a; rename the extension or convert if needed. Apple uses M4A everywhere for audio-only AAC.

Why does my AAC file end in .m4a?

AAC is a codec (the actual compression format) and M4A is a container (the file wrapper). AAC audio is almost always wrapped in an M4A or MP4 container because the raw AAC bitstream lacks metadata, seeking tables and timing information that the container provides. Pure raw AAC files exist (.aac extension) but are uncommon. Treat AAC and M4A as effectively the same for normal use; the file inside is the same audio.

Why use FFmpeg rather than a simpler converter?

FFmpeg is the reference implementation for virtually every audio codec. It supports more formats than any competing tool (hundreds), with the highest-quality encoders for most of them. Simpler converters often use FFmpeg internally anyway and just wrap it in a friendlier UI. By running FFmpeg directly in your browser, this tool gets the same quality and format support as the command-line tool used by every major streaming service, with no quality compromise from a simplified encoder.

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