WAV Format: Complete Technical Reference

WAV — short for Waveform Audio File Format, and often given the .wav or .wave extension — was introduced by Microsoft and IBM in 1991 alongside Windows 3.1's multimedia extensions. It is a specific application of the more general RIFF (Resource Interchange File Format) container, itself modeled on Electronic Arts' earlier IFF format from the Amiga. RIFF stores data in tagged 'chunks,' and WAV's minimal valid file is a RIFF header followed by a 'fmt ' chunk (describing the audio format) and a 'data' chunk (the raw samples). Because it stored audio exactly as captured, WAV became the default for CD-quality audio on PCs and the interchange format between recording hardware and software. It has barely changed in over thirty years — the core specification is essentially frozen — because an uncompressed container has little to improve. The format's few evolutions have been additive: the Broadcast Wave Format (BWF) extension in 1997 for professional broadcast, and RF64 in the mid-2000s to break the 4 GB ceiling. WAV is the direct Windows counterpart to Apple's AIFF, which uses the same underlying idea with big-endian byte ordering instead of WAV's little-endian.

WAV almost always carries linear PCM (Pulse-Code Modulation) — raw, uncompressed audio samples stored as integers or floating-point numbers. Supported bit depths are 8-bit (unsigned integer), 16-bit (the CD standard, signed integer), 24-bit (the professional recording standard), and 32-bit float (the modern DAW mixing standard, which effectively cannot clip during processing). Sample rates range from 8 kHz up through 44.1 kHz (CD), 48 kHz (video and broadcast), 96 kHz, and 192 kHz, with some tools supporting 384 kHz. CD quality is 16-bit / 44.1 kHz stereo; studio quality is 24-bit / 48 kHz or higher. WAV supports mono, stereo, and multichannel layouts including 5.1 and 7.1 surround via the WAVE_FORMAT_EXTENSIBLE header, which also carries an explicit channel mask. The data rate of 16-bit / 44.1 kHz stereo is 1,411 kbps — roughly 10.1 MB per minute. While the container can technically hold compressed codecs like ADPCM, GSM, or even MP3 data, in practice WAV means uncompressed PCM to virtually everyone who uses it.

PCM is the most literal way to store sound. An analog waveform is measured (sampled) at fixed time intervals, and each measurement is rounded to the nearest value on a fixed scale (quantized). The sample rate is how many measurements per second; the bit depth is how many distinct levels the scale has. That is the entire scheme — there is no psychoacoustic model, no frequency-domain transform, no prediction, and nothing is thrown away. Because the samples are stored directly, PCM is trivially fast to decode (no computation beyond reading numbers) and completely lossless, which is why editing software prefers it: you can cut, fade, gain, and re-save a WAV thousands of times with zero generational degradation. The trade-off is size. Every second of 16-bit / 44.1 kHz stereo audio is exactly 44,100 samples × 2 bytes × 2 channels = 176,400 bytes, whether it is silence or a full orchestra. Lossy codecs like MP3 and AAC exist precisely to escape this fixed cost by discarding inaudible detail; WAV keeps all of it.

These two numbers define WAV quality and file size. Bit depth sets the dynamic range — the distance between the quietest and loudest sound before quantization noise or clipping. Each bit adds roughly 6 dB: 16-bit gives about 96 dB of dynamic range (enough for finished music), while 24-bit gives about 144 dB, providing headroom so that quiet passages and recording levels set conservatively still stay well above the noise floor during editing. 32-bit float goes further, representing values beyond 0 dBFS so that a mix can momentarily 'clip' internally and be pulled back down with no damage. Sample rate sets the highest frequency that can be represented, which by the Nyquist theorem is half the sample rate: 44.1 kHz captures up to 22.05 kHz (beyond the roughly 20 kHz limit of human hearing), while 96 kHz captures up to 48 kHz, giving processing algorithms like pitch-shifting and time-stretching more room to work cleanly. For most finished audio, 16-bit / 44.1 kHz is transparent. For recording and mixing, 24-bit / 48 kHz is the practical professional default; higher rates mainly benefit heavy processing rather than final listening.

Standard WAV has a hard file-size ceiling: the RIFF header stores chunk sizes as unsigned 32-bit integers, capping a file at 4 GB (2^32 bytes). At 24-bit / 96 kHz stereo that is only about two hours of audio — a real constraint for long location recordings, concert captures, and multitrack sessions. Two extensions solve this. RF64, standardized by the European Broadcasting Union (EBU), replaces the RIFF fourCC with 'RF64' and adds a 'ds64' chunk carrying 64-bit sizes, lifting the limit into the exabytes while staying backward-readable as WAV up to 4 GB. Broadcast Wave Format (BWF, file extension often still .wav) adds a 'bext' metadata chunk that carries a timecode reference (SMPTE), originator information, and a coding history — essential for film and television post-production where audio must line up frame-accurately with picture. Most professional field recorders and DAWs write BWF and RF64 transparently, so you often get these extensions without choosing them explicitly.

WAV's limits follow directly from being an uncompressed PCM container. It cannot make files small — there is no compression, so a WAV is five to ten times larger than an equivalent MP3 and cannot be shrunk without converting to another format. It cannot stream efficiently over the web or fit comfortably in email, which is why distribution uses MP3, AAC, or Opus instead. Standard WAV cannot exceed 4 GB without the RF64 extension, which not every player understands. It has no standardized, universally-read metadata system — unlike MP3's ID3 tags or FLAC's Vorbis comments, WAV tagging relies on optional LIST/INFO or BWF chunks that many players ignore, so artist and album fields frequently do not survive. WAV has no built-in integrity check (FLAC embeds an MD5 of the audio; WAV does not), no chapter or bookmark support like M4B audiobooks, and no native gapless or ReplayGain metadata. And despite being lossless, WAV offers no quality advantage over FLAC or ALAC for the same audio — those formats hold identical samples in roughly half the space.

WAV versus FLAC: both are lossless and bit-for-bit identical to the source, but FLAC compresses to roughly 50-60 percent of WAV's size and carries robust metadata and an integrity checksum. The trade is a little CPU to decode and slightly less universal editing support. For archival and lossless distribution, FLAC is usually the better choice; for active editing in a DAW, WAV's zero-decode simplicity wins. WAV versus AIFF: they are technically near-twins — both uncompressed PCM — differing mainly in byte order (WAV little-endian, AIFF big-endian) and heritage (WAV from Windows, AIFF from Apple). Modern software reads both interchangeably; pick WAV on Windows-centric workflows and AIFF only if a Mac-only tool prefers it. WAV versus MP3: not really a comparison — WAV is the lossless master, MP3 is the compressed distribution copy. The correct workflow is to keep WAV (or FLAC) as your archive and export MP3 or AAC for sharing, never the reverse, since converting MP3 back to WAV cannot restore the detail the MP3 encoder already discarded.

WAV is among the most universally supported audio formats. Windows has decoded it natively since 1991; macOS handles it through CoreAudio; Linux plays it via ALSA, PulseAudio, and PipeWire. iOS and Android both play WAV files, and every major web browser supports WAV through the HTML5 audio element. On the professional side, every digital audio workstation — Pro Tools, Logic Pro, Ableton Live, FL Studio, Cubase, Reaper, and the free Audacity — reads and writes WAV as a first-class citizen, and it is the standard interchange format when moving stems between them. Video editors (Premiere Pro, DaVinci Resolve, Final Cut Pro) import WAV audio cleanly. The only real compatibility caveats are at the edges: some older or budget portable players choke on 24-bit or high-sample-rate WAV even though they play 16-bit / 44.1 kHz fine, and a handful of simple hardware players do not recognize RF64 files above 4 GB. For everyday playback of ordinary CD-quality WAV, compatibility is effectively total.

Metadata is WAV's weakest area. The base specification has no dedicated tagging system, so three partial conventions have grown up around it. The most common is a LIST chunk containing an INFO sub-chunk, which can hold fields like IART (artist), INAM (title), IPRD (album), ICRD (creation date), and ICMT (comment) — but support is inconsistent, and many players and phones simply ignore it, showing only the filename. Broadcast Wave adds the 'bext' chunk for professional metadata (timecode, originator, coding history) aimed at broadcast rather than consumer music tagging. Some tools also embed an ID3v2 chunk (the same tag system MP3 uses) inside the WAV, which improves compatibility with music managers that look for it. The practical consequence: do not rely on WAV to carry artist and album information across different apps and devices. If you need durable, universally-read tags — cover art, track numbers, multi-artist credits — convert to FLAC (Vorbis comments) or MP3 (ID3v2) for the tagged distribution copy and keep the WAV as the untagged master.

WAV size is fully predictable because it is uncompressed: bytes per second = sample rate × (bit depth ÷ 8) × channels. Concrete numbers for stereo audio. 16-bit / 44.1 kHz (CD quality): 1,411 kbps, about 10.1 MB per minute, 605 MB per hour. 16-bit / 48 kHz: 1,536 kbps, about 11 MB per minute, 660 MB per hour. 24-bit / 48 kHz (common studio setting): 2,304 kbps, about 16.5 MB per minute, 989 MB per hour. 24-bit / 96 kHz (high-resolution): 4,608 kbps, about 33 MB per minute, 1.98 GB per hour — which means a standard 4 GB WAV holds only about two hours. 32-bit float / 48 kHz: about 22 MB per minute. Mono halves every figure. For planning: a three-minute song at CD quality is roughly 30 MB; a one-hour interview recorded 24-bit / 48 kHz mono is about 495 MB. These sizes are why WAV is the right format to record and edit in, and the wrong format to email or stream — for delivery, convert down to MP3 or AAC.

Use WAV whenever audio will be edited, processed, or archived at full quality. Recording sessions should capture to WAV (or BWF) so nothing is lost before you have even started. Mastering, sound design, film and TV post-production, and multitrack mixing all live in WAV because repeated saves cause no degradation. Use WAV as the neutral intermediate when transcoding — decode a lossy file to WAV, process it, then re-encode once, rather than chaining lossy-to-lossy conversions that compound artifacts. Keep a WAV (or FLAC) master of anything important. Avoid WAV for web delivery, streaming, email attachments, podcast distribution, and phone storage, where its size is a liability and a listener cannot hear the difference versus a high-bitrate MP3 or AAC. The canonical pipeline: record and edit in WAV, archive in WAV or FLAC, distribute in MP3 or AAC.

Converting to WAV is the standard first step before editing a compressed file: decode MP3, M4A, OGG, or Opus to WAV so your editor works on raw PCM. Note that converting a lossy file to WAV does not restore quality — it simply stops further loss during editing; the artifacts already baked into the MP3 remain. From FLAC or AIFF to WAV is truly lossless (same PCM samples, different container). Converting from WAV to a lossy format (MP3, AAC, OGG, Opus) is how you produce distribution copies: choose 192-320 kbps for music or 128 kbps mono for voice. From WAV to FLAC or ALAC gives you a smaller lossless archive with proper metadata and integrity checking. AudioUtils runs every one of these conversions entirely in your browser via FFmpeg WebAssembly — your files never leave your device, there is no upload, no signup, and no software to install, and even large studio WAV files are processed locally on your own machine.