Video to WAV Converter

Convert Video files to WAV format online. Free, fast, no watermarks.

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Supports: 3G2, 3GP, 3GPP, ASF, AV1, AVCHD +31 more

OptionsAdvanced Options - Our defaults are optimized for the best results. We recommend you keeping the defaults unless you have a specific need.
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Audio Channel
Audio Channel
Audio Sample Rate
Audio Sample Rate
Trim

How to Convert Video to WAV Online

  1. Upload Your Video File: Drag and drop or click "Add Files" to select video clips. MP4, MOV, MKV, AVI, WebM, FLV, WMV, M2TS, MTS, AVCHD, MPEG, 3GP, and 30+ other container formats are accepted. Batch is supported — extract WAV tracks from an entire folder in one pass.
  2. Choose Audio Channel and Audio Sample Rate: Defaults are set to ORIGINAL (passes the source's channel layout and sample rate through untouched — the right pick for editing and archival). To force stereo or mono, change Audio Channel. To resample, pick a target Audio Sample Rate (48 kHz for video post, 44.1 kHz for music workflows, 16 kHz for speech-to-text).
  3. Trim (Optional): Expand the Trim section to extract a clip instead of the full audio track. Enter start time and duration in seconds (e.g. 12.5) or HH:MM:SS.sss (e.g. 00:01:30.500). Leave Trim "Unchanged" to convert the full duration.
  4. Convert and Download: Click Convert. Files process on our servers and download individually or as a single ZIP — no sign-up, no watermark.

Why Convert Video to WAV?

Video files mux a visual track with an audio track that's usually encoded as AAC, AC3, Opus, or MP3 — all lossy. Converting the video to WAV decodes that audio track to uncompressed PCM, the format every audio editor, transcription engine, and DAW prefers to work with. The picture is discarded; only the soundtrack remains. Common reasons to extract WAV from video:

  • Audio editing in a DAW — Pro Tools, Logic Pro, Reaper, Cubase, Audacity, and Adobe Audition all treat WAV (PCM s16le or s24le) as the native recording target. Importing a video's AAC track forces a decode-and-resample step inside the DAW; importing the WAV is plug-and-play.
  • Speech-to-text and transcription — Whisper, faster-whisper, Deepgram, AssemblyAI, Otter, Rev, and Google Speech-to-Text all accept WAV directly and many internally resample everything to 16 kHz mono PCM. Feeding them a clean WAV avoids the engine's own decode step and often improves transcription accuracy.
  • Lossless source for re-encoding — If you need both an MP3 for distribution and an FLAC for archival from the same source video, start with a WAV master and re-encode from there. Going video → MP3 → FLAC compounds lossy decoding artifacts; video → WAV → MP3 / FLAC does not.
  • Music sampling and remix work — Producers pulling a vocal hook, drum break, or movie quote from a video clip want PCM, not lossy AAC, before time-stretching, pitch-shifting, or layering in Ableton, FL Studio, or MASCHINE.
  • Forensics, dictation, and legal evidence — Court reporters and investigators need an uncompressed, lossless copy of the audio they can re-process without quality drift across passes.
  • Sound-design library ingest — When pulling Foley, ambience, or SFX from royalty-free video sources, libraries like Sound Particles, BOOM Library, or local Reaper folders expect WAV.

WAV vs MP3 (Extracted from Video) — Format Comparison

Property WAV (extracted) MP3 (extracted)
Compression Uncompressed PCM Lossy (perceptual coding)
Typical bitrate 1411 kbps (16-bit / 44.1 kHz stereo); 2304 kbps at 24-bit / 48 kHz 64-320 kbps
Size of 10-minute audio track ~100 MB at CD quality; ~165 MB at 24-bit/48 kHz ~7-23 MB
Quality vs the video's audio track Bit-perfect decode of the in-container codec Re-encoded — second-generation lossy
Best for Editing, transcription, archival, re-encoding source End-listening, sharing, mobile
DAW / NLE friendliness Native import everywhere Decoded on import
Speech-to-text engines Preferred — no re-decode required Accepted but re-decoded

Sample Rate and Bit Depth Quick Guide

Setting When to pick it Notes
48 kHz, 16-bit, stereo Video post-production, re-attaching to NLE timelines (Premiere, Resolve, Final Cut) The film/TV standard; matches what most video containers store internally
44.1 kHz, 16-bit, stereo Music workflows, CD-targeted masters, Spotify / Apple Music uploads "Red Book" CD quality
48 kHz, 24-bit, stereo High-end post and music production where headroom matters Roughly 1.5× the file size of 16-bit at the same rate
16 kHz, 16-bit, mono Whisper, faster-whisper, and most ASR engines Smallest WAV that still gives you full speech intelligibility
Original (passthrough) "Just give me the audio track as PCM, unchanged" Avoids any resampling; the cleanest possible decode

Frequently Asked Questions

Does extracting WAV from a video lose quality compared to the source?

The decode itself is bit-perfect — PCM samples come out of the demuxer exactly as the encoder wrote them. But the audio track inside the video was almost certainly lossy to begin with (AAC for MP4 / MOV, Opus / Vorbis for WebM, AC3 for AVCHD / MKV). The WAV is a lossless copy of an already-lossy track, not a recovery of the original studio audio. That said, a WAV decode is still the right format for editing and transcription because every subsequent processing step stays lossless from this point onward.

Why is the WAV so much bigger than the source video?

Because video formats compress audio aggressively. A 10-minute MP4 with a 128 kbps AAC track holds about 9 MB of audio data; the same 10 minutes decoded to 16-bit / 48 kHz stereo WAV is ~110 MB — roughly 12× larger. Uncompressed PCM stores every sample at full bit depth with no perceptual modeling. The bloat is the point: it's exactly what your editor or transcription engine wants.

What sample rate should I pick — 44.1 kHz or 48 kHz?

If the audio is going back into a video editor (Premiere, DaVinci Resolve, Final Cut, Audition's video session), pick 48 kHz — that's the broadcast / film standard and what your video container already uses internally. If the audio is going into a music project, CD master, or Spotify upload, pick 44.1 kHz. Mixing the two forces a resample somewhere downstream and can introduce subtle artifacts.

How do I extract just a clip instead of the whole soundtrack?

Use the Trim section. Enter a start time and a duration; both accept either plain seconds (75) or HH:MM:SS.sss (00:01:15.000). You can also do this in two passes: extract the full WAV first, then use Audio Cutter to make fine-grained edits to the PCM data.

Will this work on M2TS, AVCHD, or MTS camera recordings?

Yes — they're handled the same way as MP4 and MOV. The audio inside AVCHD camcorder files is usually AC3 or LPCM; the converter decodes either to WAV. If you have raw .mts or .m2ts files from a Sony or Panasonic camcorder, drop them in directly — no need to remux to MP4 first.

Can I batch-extract WAV from a whole folder of videos?

Yes. Drop in every clip at once and the same conversion settings apply to all files. They process in parallel on our servers and download individually or as a single ZIP. Useful for podcast back-catalog extraction, course-video transcription prep, or migrating a footage library to an audio-only archive.

Should I pick stereo or mono?

Pick stereo if the source has stereo content you care about (music, ambience, dialog with directional cues) or if you're handing off to a video editor. Pick mono if the source is single-channel dialog (interview mics, lavs, voice memos) and you're feeding a speech-to-text engine — mono cuts file size in half and most ASR models internally downmix anyway. Original keeps whatever the source has.

How does this compare to extracting MP3?

WAV is lossless PCM — the right format if you'll edit, transcribe, or re-encode the audio. MP3 is lossy compressed — the right format if you're going to ship the file to a listener or upload to a hosting platform. If your end goal is listening, Video to MP3 gives you a much smaller file with near-identical perceived quality. Use this WAV converter for the editing-and-processing path; use the MP3 converter for the listening-and-sharing path.

What if my video has multiple audio tracks (different languages, 5.1 surround)?

The converter extracts the default audio track that the video player would use. If a file has separate dialog, music, and effects stems on different tracks, only the primary one is converted. For 5.1 / 7.1 surround, the output is downmixed to stereo by default — useful for transcription and most editing, but if you need full discrete surround channels exported you'll want a desktop tool like FFmpeg with explicit channel mapping.

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