Audio Format Converter

WAV, MP3, OGG, and FLAC: Choosing the Right Audio Format

Audio format decisions affect file size, playback compatibility, perceived quality, and how platforms process your music. The four formats this converter supports cover every common use case in music production, distribution, and streaming. Understanding how each one works helps you pick the right format for the job rather than guessing.

WAV: The Production Standard

WAV (Waveform Audio File Format) stores audio as uncompressed PCM data. Every sample is written to disk exactly as your DAW rendered it. A stereo 44.1 kHz / 16-bit WAV file uses approximately 10 MB per minute. A 24-bit / 96 kHz session file uses roughly 34 MB per minute per channel.

WAV is the format most distributors and mastering engineers request for delivery. DistroKid, TuneCore, CD Baby, and most other distributors accept WAV as the preferred upload format. They transcode to the formats each platform needs on their end, so you start the chain with the cleanest possible source.

This converter outputs WAV as PCM 16-bit, which matches the CD standard (44100 Hz / 16-bit / stereo). If your source file is 24-bit or 32-bit float, the output is dithered down to 16-bit by ffmpeg. For a full breakdown of bit depth and sample rate trade-offs, see the Sample Rate Calculator.

MP3: The Universal Delivery Format

MP3 uses the MPEG-1 Audio Layer III psychoacoustic compression algorithm, developed at the Fraunhofer Institute. It works by removing audio information that the human ear is least likely to notice, such as very quiet sounds that occur simultaneously with very loud sounds in the same frequency region. This is called masking, and it is the core reason MP3 can reduce a WAV file to 8 to 10 percent of its original size while sounding reasonably close to the original.

The bitrate setting directly controls how much data is kept per second of audio. At 128 kbps, an average listener may notice compression artifacts on busy, high-frequency content like cymbals or acoustic guitar. At 192 kbps, artifacts are present but rarely audible without direct comparison to a lossless source. At 320 kbps, MP3 is considered transparent by most listeners, meaning the compression difference from WAV is not perceptible on consumer playback gear.

For music you plan to share with collaborators or upload as a rough mix reference, 192k is sufficient. For anything going to a final listener or used as a backup copy when a WAV is not available, 320k is the right choice. Once you compress to MP3, converting back to WAV does not restore the removed data. The lost frequencies are gone permanently.

Spotify, Apple Music, and YouTube all transcode uploaded audio to their own delivery formats anyway. They do not stream your original file. Spotify converts audio to OGG Vorbis at Q5 (around 160 kbps) for desktop clients and to AAC at 256 kbps for iOS. Apple Music delivers AAC at 256 kbps. So uploading a 320k MP3 to Spotify is functionally the same as uploading a WAV: both get re-encoded. Use our LUFS Loudness Meter to check your master before distribution, since each transcoding step can affect peaks and perceived loudness.

OGG Vorbis: Open-Source Lossy Compression

OGG Vorbis is a royalty-free lossy format developed by the Xiph.Org Foundation. It uses variable bitrate encoding and a quality scale from -1 to 10. At Q5 (roughly 160 kbps average), OGG Vorbis produces better audio quality than MP3 at 160 kbps in most listening tests. At equivalent bitrates, OGG consistently outperforms MP3 in double-blind evaluations, particularly on complex stereo material.

Spotify chose OGG Vorbis as its primary desktop streaming codec for exactly this reason. Their premium desktop streams are OGG at Q9 (approximately 320 kbps). Their standard desktop streams are Q5 at roughly 160 kbps. The codec is open, patent-free, and decodes efficiently on modern hardware.

OGG is less universally supported than MP3 in standalone hardware players and some older mobile apps. For web and software distribution, it is an excellent choice. For sending files to labels, sync libraries, or clients who may open them on any device, MP3 or WAV is safer.

FLAC: Lossless Without the WAV File Size

FLAC (Free Lossless Audio Codec) compresses audio without removing any data. Decompressing a FLAC file produces a bit-perfect copy of the original. A stereo 44.1 kHz / 16-bit FLAC file is typically 40 to 60 percent smaller than the equivalent WAV. At 24-bit, the compression ratio tends toward the higher end. Tidal and Apple Music both support lossless streaming using FLAC and ALAC respectively, and subscriber counts for their lossless tiers have grown steadily since their launches in 2021.

FLAC is the standard format for archiving audio. If you have original multitracks or mix sessions on an external drive, converting stems to FLAC saves roughly half the storage space without any quality penalty. It also loads faster than WAV in many DAWs because modern CPUs can decompress FLAC faster than a slow drive can read a larger WAV file sequentially.

FLAC is not universally supported in all streaming platforms or web browsers out of the box. For delivery to distributors, WAV is still preferable. For personal archives, high-fidelity playback, or sending lossless files to mastering engineers, FLAC is a practical choice.

How ffmpeg.wasm Runs Conversion in Your Browser

This tool uses ffmpeg.wasm, a port of the FFmpeg command-line tool compiled to WebAssembly. FFmpeg is the most widely used open-source audio and video processing library. It powers conversion features in VLC, HandBrake, YouTube's processing pipeline, and thousands of other applications.

WebAssembly (WASM) lets compiled C and C++ code run inside a browser at near-native speeds. The ffmpeg core, including all its codecs for MP3, OGG Vorbis, FLAC, and WAV, is loaded from a CDN as a WASM binary. Once loaded (typically 5 to 15 seconds on first use), it runs entirely within your browser tab. Your audio file is written to an in-memory virtual filesystem inside the WASM sandbox, converted, and then handed back to your JavaScript context as a downloadable Blob. No data touches any server.

This tool uses the single-threaded ffmpeg core, which does not require the SharedArrayBuffer API. That API requires specific HTTP security headers that can interfere with third-party embeds on other pages. The single-threaded core is fully functional for typical audio files and conversion takes a few seconds on most hardware. Conversion speed scales with file duration: a 3-minute MP3 converts in roughly 2 to 5 seconds on a modern laptop.

When to Use Each Format

For uploading to distributors: WAV at 44.1 kHz / 16-bit or 24-bit. Most distributors explicitly prefer WAV and some reject MP3 for mastered tracks.

For sharing a mix reference with a collaborator or client: MP3 at 192k or 320k. Small enough to email, plays on anything.

For archiving original sessions and stems: FLAC. Half the storage of WAV, bit-perfect quality, supported by all major DAWs.

For web audio players and HTML5 audio tags: OGG Vorbis for Chrome and Firefox, MP3 as a fallback for Safari and older browsers. Most modern web audio players handle both.

For sync licensing submissions: Check the brief. Most music supervisors request WAV at 44.1 kHz / 16-bit or 24-bit. Some accept high-quality MP3 at 320k for initial review, with WAV required for final placement.

Bitrate, Quality, and the Transparency Threshold

The concept of transparency in lossy audio refers to the bitrate above which an average listener cannot distinguish the compressed version from the lossless original in a controlled double-blind test. For MP3, transparency is generally reached between 192 kbps and 256 kbps depending on the content. Complex material with lots of high-frequency information (acoustic guitar, strings, cymbals) tends to reveal compression at lower bitrates. Simple material with limited frequency content (spoken word, bass-heavy electronic music) may sound transparent even at 128 kbps.

For OGG Vorbis, transparency is typically reached at Q5 (about 160 kbps). This is one of the reasons Spotify uses Q5 for standard streams. The codec achieves the same perceptual quality as MP3 at a lower average bitrate.

If your goal is archiving, do not use lossy compression regardless of bitrate. Use FLAC or WAV. Quality settings for lossy formats only matter for listening copies, not for archival master files. Converting a 128k MP3 to 320k MP3 does not improve it. You are re-encoding already-degraded audio with a larger container, and the artifacts from the first compression pass are baked in permanently.

For an overview of how audio quality interacts with streaming income, see the Streaming Royalty Calculator, which shows how stream counts and per-stream rates connect to your revenue across platforms that all use their own codec pipelines.

For more on how streaming platforms handle audio encoding, the Vorbis I Specification from Xiph.Org documents the exact encoding algorithm used in OGG Vorbis, and the FLAC format specification covers the lossless compression algorithm in technical detail.