Convert MP4 to WAV — Free Online Tool
Convert MP4 video files to WAV audio by extracting and decoding the audio stream to uncompressed PCM (16-bit little-endian) format. WAV is the go-to format for professional audio work, offering bit-perfect audio quality with no lossy compression — ideal when you need a lossless master from video source material.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
MP4 files typically carry audio encoded with a lossy codec such as AAC, MP3, or Opus. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio into raw PCM samples, then wraps them in a WAV container using the pcm_s16le codec — 16-bit signed integers in little-endian byte order, at whatever sample rate the source MP4 used (commonly 44100 Hz or 48000 Hz). Because AAC (the most common MP4 audio codec) is lossy, this conversion does not recover quality that was lost during the original encoding — it produces a lossless representation of the decoded audio as it currently exists. The resulting WAV file will be significantly larger than the MP4 source, as there is no compression applied to the audio samples.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. This is the same underlying engine that runs in your browser via WebAssembly — copying this command lets you run the identical conversion locally on any machine with FFmpeg installed. |
-i input.mp4
|
Specifies the input file — in this case an MP4 container that may contain video (typically H.264 or H.265), audio (typically AAC), and optionally subtitles or chapters. FFmpeg reads all available streams from this file and selects which to process based on the rest of the command. |
-c:a pcm_s16le
|
Sets the audio codec for the output to pcm_s16le — signed 16-bit little-endian PCM, the standard uncompressed audio format used in WAV files. FFmpeg decodes the compressed AAC audio from the MP4 and re-encodes it as raw PCM samples, producing bit-perfect uncompressed audio at the same sample rate as the source. |
output.wav
|
Specifies the output filename and container format. The .wav extension tells FFmpeg to wrap the PCM audio in a WAV container. Since WAV is audio-only, FFmpeg automatically discards the video stream from the source MP4 — no explicit video suppression flag is needed because WAV cannot contain video. |
Common Use Cases
- Extract a clean, uncompressed audio track from a recorded interview or lecture video for editing in a DAW like Pro Tools, Logic Pro, or Adobe Audition
- Prepare dialogue or voiceover audio from an MP4 screen recording for professional post-production, where WAV is required by the audio editor
- Extract audio from an MP4 music video to create a WAV master for archival purposes or CD authoring workflows
- Supply a broadcast-ready WAV file to a radio station or podcast distributor that requires uncompressed audio rather than AAC-encoded video files
- Pull audio from MP4 training or tutorial videos to feed into speech recognition or transcription software that works best with uncompressed WAV input
- Extract a game or film score from an MP4 trailer to create a WAV file for audio analysis or stem separation tools
Frequently Asked Questions
No — the output WAV will not have higher quality than the audio that was in the MP4. MP4 files almost always store audio as AAC, which is a lossy codec that permanently discards audio data during compression. Converting to WAV decodes that AAC audio to uncompressed PCM, which prevents any further quality loss, but it cannot restore information already removed by AAC encoding. Think of it as a lossless copy of a lossy original.
WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression applied. A single minute of stereo audio at 48000 Hz works out to roughly 11 MB in WAV format. MP4 uses AAC, which typically compresses audio to around 128–256 kbps, making the MP4 audio track many times smaller. The WAV file size is determined entirely by duration, sample rate, and bit depth — not by the loudness or complexity of the audio content.
The video stream is completely discarded. FFmpeg reads only the audio stream from the MP4 and writes it to the WAV container. WAV is a purely audio format and cannot store video data, so this is always an audio-only extraction. If you also need the video, you would need a separate conversion step.
No — WAV is a single-stream audio container and cannot hold multiple audio tracks. If your MP4 has several audio tracks (such as multiple language dubs or a stereo and surround mix), this conversion will extract only the default or first audio stream. If you need a specific track from a multi-track MP4, you would need to add a stream selector flag like -map 0:a:1 to the FFmpeg command to pick a particular track by index.
Yes. The command uses pcm_s16le for 16-bit audio, but you can change the codec to pcm_s24le for 24-bit or pcm_s32le for 32-bit by replacing -c:a pcm_s16le with your preferred codec. To resample the audio, add -ar 44100 or -ar 48000 to set the output sample rate explicitly. For example: ffmpeg -i input.mp4 -c:a pcm_s24le -ar 48000 output.wav produces a 24-bit WAV at 48 kHz, which is common in professional broadcast and music production workflows.
Yes, on the command line you can loop over files. In a bash shell on macOS or Linux, use: for f in *.mp4; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.mp4}.wav"; done. On Windows Command Prompt, use: for %f in (*.mp4) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This will process each MP4 in the current directory and output a matching WAV file. The browser-based tool processes one file at a time.
Technical Notes
The pcm_s16le codec used here is the most universally compatible WAV variant — it is what most hardware recorders, DAWs, broadcast systems, and speech processing tools expect by default. The 's16' means signed 16-bit integer, and 'le' means little-endian byte order, which is the native byte order on x86 systems and the WAV standard. The output sample rate is inherited from the source MP4 audio stream (typically 44100 Hz for music-sourced content or 48000 Hz for video-sourced content) unless explicitly overridden. WAV does not support subtitle tracks, chapter markers, or metadata fields beyond a limited set of INFO chunk tags, so all MP4 metadata including title, artist, album art, and chapter data is lost during conversion. If the MP4 source audio was encoded at a low bitrate (such as 96k AAC from a compressed web video), the WAV output will faithfully represent that lower-quality audio with no upsampling artifacts, but will not recover missing frequency content. For workflows requiring higher bit depth (e.g., 24-bit for audio mastering), changing the codec flag to pcm_s24le is recommended without any other change to the command.