How to Convert MP4 to WAV

WAV is the right output format when you intend to edit, mix, or process the audio after extraction. Video editors and DAWs such as Premiere Pro, Final Cut Pro, Logic Pro, and Ableton Live all work best with uncompressed audio on the timeline. MP3's lossy compression introduces artifacts. When you apply processing — EQ, compression, pitch shifting — to a lossy file, those artifacts become more audible. WAV eliminates that risk. Choose WAV when: the extracted audio will go into an editing session, you need to match a project's sample rate precisely, or you are archiving the audio for long-term storage.

The conversion pipeline runs entirely in your browser via FFmpeg compiled to WebAssembly. AudioUtils demuxes the MP4 container, reads the audio stream (typically AAC or AC-3), decodes it to raw PCM samples, and writes those samples into a WAV container with a standard RIFF header. The output is a 16-bit PCM WAV file at the original sample rate of the source audio. No quality is lost beyond what was already in the source stream — if the MP4 had AAC audio at 48 kHz, the WAV will be PCM at 48 kHz with no further degradation. Conversion speed depends on file size and CPU. A 500 MB MP4 typically converts in under two minutes on modern hardware.

WAV files store audio as PCM — Pulse Code Modulation. Each audio sample is stored as a numeric value directly, with no compression algorithm applied. This makes WAV files large but perfectly suited for editing. A one-minute stereo WAV at 44.1 kHz, 16-bit takes approximately 10 MB. At 48 kHz (common for video audio) it is about 11 MB per minute. For a 30-minute MP4, expect a WAV output around 300–330 MB. If file size is a concern for archival but you still need lossless quality, FLAC is a better choice — it compresses PCM data by 40–60% with zero quality loss.

Most MP4 video files use AAC audio at 48 kHz — the broadcast standard for video. Some older files use 44.1 kHz — the CD standard. AudioUtils preserves the source sample rate in the WAV output. If your DAW project is set to 44.1 kHz and the WAV comes out at 48 kHz, import it anyway — your DAW will resample automatically. For precise control, check the source file's audio properties before converting. Bit depth in the WAV output defaults to 16-bit. This matches CD quality. Professional recordings may benefit from 24-bit, but since the source AAC is already lossy, 16-bit captures all the meaningful data present.

Output WAV is very large: This is expected. Uncompressed audio is large by definition. A 100 MB MP4 can produce a 300+ MB WAV because the video was compressed and the WAV audio is not. No audio in the output: Verify the MP4 actually contains an audio track. Some screen recordings and time-lapse videos have no audio. Play the file in a media player before converting. DAW reports unsupported format: Most DAWs want stereo or mono WAV. If the source MP4 had surround audio (5.1 channels), the WAV output may have 6 channels. Use a DAW import dialog to downmix to stereo on import. Conversion fails on very large files: Browser memory limits can affect files over 2 GB. Splitting the video first may help.

The choice between WAV and MP3 as output comes down to what you do with the file next. For editing and production: WAV. Video editors and DAWs expect uncompressed audio. Working with MP3 in a professional timeline introduces re-encoding steps that degrade quality. For listening and sharing: MP3. WAV files are 10 times larger than equivalent-quality MP3 and most listeners cannot distinguish the difference on consumer headphones. For archival: WAV or FLAC. If you are storing the audio as the master copy, uncompressed (WAV) or losslessly compressed (FLAC) formats guarantee no quality drift over time or across future re-encoding steps.