Extract Audio from M4V to WAV — Free Online Tool
Extract audio from M4V video files and save it as uncompressed WAV audio. This tool decodes the AAC audio track commonly found in iTunes-format M4V files and outputs 16-bit PCM WAV — the raw, lossless waveform format used in professional audio workflows.
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FFmpeg Command
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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 typically contain AAC-encoded audio, a lossy compressed format optimized for iTunes and iOS playback. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio data from the M4V container, re-encoding it as uncompressed PCM (pulse-code modulation) audio at 16-bit depth stored in a WAV container. Because AAC is lossy and PCM is uncompressed, the output WAV will not recover any audio detail that was lost during the original AAC encoding — but it will be a faithful, uncompressed representation of what was in the M4V, with no further generation loss. The resulting WAV file will be significantly larger than the source audio track, since PCM stores every sample without compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing engine, which handles demuxing the M4V container, decoding the AAC audio stream, and encoding the output PCM WAV file. |
-i input.m4v
|
Specifies the input M4V file. FFmpeg reads the container, identifies the available streams (typically an H.264 video track and an AAC audio track), and makes them available for processing. |
-vn
|
Disables video output entirely, instructing FFmpeg to ignore the H.264 or H.265 video stream in the M4V and include only the audio stream in the output. This is essential for producing a clean audio-only WAV file. |
-c:a pcm_s16le
|
Sets the audio codec to 16-bit signed little-endian PCM, the standard uncompressed audio encoding used in WAV files. This decodes the lossy AAC audio from the M4V into raw, uncompressed waveform data compatible with virtually all audio software. |
output.wav
|
Defines the output file name and format. The .wav extension tells FFmpeg to use the WAV/RIFF container, which packages the uncompressed PCM audio stream in a format widely supported by DAWs, broadcast tools, and audio editors. |
Common Use Cases
- Extracting the audio from an iTunes movie or TV show download to import into a digital audio workstation (DAW) for editing or remixing
- Pulling spoken-word audio from an M4V lecture or educational video to transcribe it using software that requires WAV input
- Preparing extracted M4V audio for broadcast or podcast post-production workflows that require uncompressed WAV deliverables
- Converting M4V audiobook or music video audio to WAV for archiving purposes where uncompressed format is preferred for long-term storage
- Extracting audio from an iOS-recorded or iTunes-synced M4V file to use as a raw sample in music production software that ingests PCM WAV files
- Stripping the audio track from an M4V file before importing into video editing tools that handle audio and video as separate WAV and video assets
Frequently Asked Questions
The WAV file will be an uncompressed PCM representation of the audio that was in the M4V, but it is not truly lossless in the audiophile sense. M4V files almost always contain AAC audio, which is a lossy codec — meaning some audio data was discarded when the M4V was originally created. Converting that AAC audio to PCM WAV does not restore lost detail; it simply decompresses what remains into an uncompressed format. You will not experience any additional quality loss from this conversion, making the WAV output a faithful but bounded copy.
AAC, the codec used for audio in most M4V files, achieves roughly 10:1 compression compared to uncompressed audio. When FFmpeg decodes AAC and writes it as 16-bit PCM WAV, all that compression is removed and every audio sample is stored in full. A 128 kbps AAC audio track from a one-hour M4V might consume around 58 MB, whereas the equivalent 16-bit PCM WAV at 44.1 kHz stereo would be approximately 600–700 MB. This is expected and normal behavior.
No. WAV has very limited metadata support and does not support chapter markers, track listings, or the rich iTunes-style metadata tags (title, artist, album artwork) that M4V containers can carry. The extracted WAV file will contain basic audio data but none of the structural or descriptive metadata from the source M4V. If metadata preservation matters, consider extracting to a format like FLAC or M4A instead.
By default, FFmpeg selects the first audio track, which is typically the default language track as flagged in the M4V container. M4V supports multiple audio tracks (common in iTunes content with alternate language dubs), but the WAV format only supports a single audio stream. If you need a specific track other than the default, you can modify the FFmpeg command to add '-map 0:a:1' (for the second audio track) before the output filename to select a different stream by index.
The default output of this command is 16-bit PCM (pcm_s16le) at whatever sample rate was used in the source M4V, typically 44100 Hz or 48000 Hz. To change the bit depth to 24-bit, replace '-c:a pcm_s16le' with '-c:a pcm_s24le'. To force a specific sample rate, add '-ar 48000' (or another rate) before the output filename. For example: 'ffmpeg -i input.m4v -vn -c:a pcm_s24le -ar 48000 output.wav' produces a 24-bit, 48 kHz WAV suitable for professional video post-production.
The single-file command shown here processes one file at a time, but you can wrap it in a shell loop for batch processing. On Linux or macOS, run: 'for f in *.m4v; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.m4v}.wav"; done'. On Windows Command Prompt, use: 'for %f in (*.m4v) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav"'. This is particularly useful for the desktop FFmpeg workflow when processing large collections of iTunes-format video files that exceed the browser tool's 1 GB limit.
Technical Notes
M4V is structurally nearly identical to MP4 but includes Apple-specific extensions such as DRM (FairPlay), chapter markers, and AC3 audio for Apple TV content. Most non-DRM M4V files use AAC audio encoded at 128–256 kbps, which this tool decodes and writes as pcm_s16le WAV — 16-bit signed little-endian PCM, the most universally compatible PCM variant. The output sample rate is inherited from the source file; iTunes-originated M4V files are commonly 44100 Hz, while M4V files from video cameras or screen recordings may be 48000 Hz. WAV's practical file size limit is 4 GB due to its use of 32-bit chunk size headers in the standard RIFF format, which means very long high-sample-rate audio (hours of 24-bit 96 kHz content) could theoretically hit this ceiling — though for typical M4V audio at 16-bit 44.1 kHz stereo, you would need over 6.5 hours of audio to approach that limit. DRM-protected M4V files purchased from iTunes cannot be processed by FFmpeg or this tool; only DRM-free M4V content is compatible. Subtitle and chapter data in the M4V source is ignored entirely, as WAV has no mechanism to carry either.