Convert M4V to WAV — Free Online Tool
Extract audio from M4V video files and save it as uncompressed WAV, converting the AAC or MP3 audio track from Apple's iTunes-compatible container into a broadcast-ready PCM waveform. This is ideal for audio editors and post-production workflows that require lossless, uncompressed audio rather than the lossy AAC encoding typical of M4V files.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
M4V files store audio most commonly as AAC (Advanced Audio Codec), a lossy compressed format optimized for iTunes and iOS playback. Converting to WAV involves decoding that AAC stream entirely — decompressing it from its lossy encoded state back into raw PCM samples — and then writing those samples into a WAV container using the pcm_s16le codec (16-bit signed little-endian PCM). The video stream is discarded completely, since WAV is an audio-only format. Because AAC is lossy, the WAV output is not a perfect reconstruction of the original pre-compression audio; it is a decoded representation of what survived the AAC encoding step. The resulting WAV file is uncompressed and immediately compatible with virtually any DAW, audio editor, or broadcast system without further decoding.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, the open-source multimedia processing engine that powers this conversion both in the browser via WebAssembly and on the desktop via a local installation. |
-i input.m4v
|
Specifies the input file — an M4V container, which typically holds H.264 or H.265 video alongside AAC or MP3 audio in an Apple-compatible MPEG-4 wrapper. FFmpeg reads all streams from this file and makes them available for processing. |
-c:a pcm_s16le
|
Decodes the M4V's AAC audio stream and re-encodes it as 16-bit signed little-endian PCM — the standard uncompressed audio encoding used in WAV files. This is the step that transforms compressed, lossy AAC data into a flat, uncompressed waveform ready for any audio editor or broadcast system. |
output.wav
|
Defines the output filename and format. The .wav extension tells FFmpeg to write a Waveform Audio File container, and since WAV is audio-only, the video stream from the M4V is automatically discarded without needing an explicit -vn flag. |
Common Use Cases
- Import dialogue or music from an iTunes-purchased M4V movie into a DAW like Pro Tools or Adobe Audition for audio restoration or remixing
- Extract the audio commentary track from an M4V video file for transcription or accessibility purposes, where WAV is required by the transcription service
- Convert iTunes video lesson audio to WAV for import into audio fingerprinting or music analysis software that does not support AAC
- Prepare audio from an M4V screen recording for inclusion in a broadcast or podcast production chain that mandates uncompressed WAV deliverables
- Strip and archive the audio from an M4V file collection in WAV format for long-term archival storage where open, uncompressed formats are preferred over Apple's proprietary container
- Extract a music track embedded in an M4V music video file for use in a video editing project that requires WAV for frame-accurate audio syncing
Frequently Asked Questions
Not in the strictest sense. AAC is a lossy codec, meaning some audio information was permanently discarded when the M4V was originally encoded. The WAV file produced here faithfully represents the decoded AAC audio — it is uncompressed PCM — but it cannot recover what was lost during the original AAC compression. Think of it as a lossless container holding a lossy source: no further quality degradation occurs during this conversion, but the audio quality ceiling is set by the original AAC encode.
WAV with pcm_s16le stores audio as raw, uncompressed 16-bit samples at the full sample rate — typically 44,100 or 48,000 samples per second per channel. AAC in an M4V file achieves compression ratios of roughly 10:1 or higher. A one-hour M4V with a 128 kbps AAC audio track might have only about 56 MB of audio data, while the equivalent WAV could be close to 600 MB. This is expected and normal: you are trading file size for universal compatibility and zero decoding overhead.
No. WAV is a simple audio-only container with no support for chapters, subtitles, or multiple tracks. This conversion extracts only the first (default) audio stream from the M4V and discards all video, subtitle, and secondary audio data. If your M4V contains multiple audio tracks — for example, a director's commentary alongside the main mix — and you need both, you would need to run separate FFmpeg commands using the -map flag to select each stream individually.
Yes. The command uses pcm_s16le by default, which produces 16-bit WAV suitable for most purposes. To get 24-bit WAV — preferred for professional audio production and mastering — replace pcm_s16le with pcm_s24le in the command: ffmpeg -i input.m4v -c:a pcm_s24le output.wav. For 32-bit floating-point output, use pcm_f32le. Note that if the original AAC was encoded from a 16-bit source, increasing the bit depth will not add real information, but 24-bit is a safe choice for downstream editing flexibility.
No. M4V files purchased from iTunes that are protected by Apple FairPlay DRM cannot be decoded by FFmpeg, including the WebAssembly version used here. Attempting to convert a DRM-protected M4V will result in an error or a silent/empty audio output. DRM-free M4V files — such as those from ripped Blu-rays, personal screen recordings, or iTunes content that has been matched and re-downloaded as DRM-free — will convert without any issue.
On macOS or Linux, you can loop over all M4V files in a directory with a shell one-liner: for f in *.m4v; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.m4v}.wav"; done. On Windows Command Prompt, use: for %f in (*.m4v) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This is particularly useful for files over 1 GB, which exceed the browser tool's limit but work fine on the desktop with FFmpeg installed locally.
Technical Notes
The pcm_s16le codec chosen here — 16-bit signed little-endian PCM — is the WAV format's most universally compatible variant and the default used by Windows, most DAWs, and broadcast playout systems. The sample rate of the output WAV is inherited directly from the source M4V audio stream (commonly 44,100 Hz for music content or 48,000 Hz for video production audio) without any resampling, preserving timing accuracy. M4V's AAC audio is typically encoded at 128–256 kbps; once decoded to pcm_s16le at 48,000 Hz stereo, the bitrate balloons to approximately 1,536 kbps, which is why file sizes increase dramatically. Metadata such as iTunes tags (title, artist, album art) and chapter markers are not written into the WAV output, as the WAV specification has only limited and non-standardized metadata support. If metadata preservation matters, consider WAV's INFO chunk support — but most tools ignore it. One known limitation: if the M4V contains an HE-AAC (AAC-LC + SBR) audio track, FFmpeg will correctly decode and upsample it to the full bandwidth frequency, so the output will reflect the actual perceived audio quality rather than just the core AAC layer.