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Supports: 3G2, 3GP, 3GPP, ASF, AV1, AVCHD +31 more
12.5) or HH:MM:SS.sss (e.g. 00:01:30.500). Leave Trim "Unchanged" to convert the full duration.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:
| 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 |
| 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 |
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.
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.
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.
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.
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.
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.
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.
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.
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.