Convert AVI to WAV — Free Online Tool
Extract and convert audio from AVI video files into WAV format using PCM encoding — producing an uncompressed, broadcast-ready audio file with no quality loss. This tool strips the video stream entirely and decodes whatever audio codec was stored in the AVI (typically MP3 or AAC) into raw 16-bit PCM WAV, the universal lossless audio format.
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
AVI files commonly store audio as compressed streams — often MP3 (libmp3lame) or sometimes AAC or Vorbis. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio track from the AVI container, then re-encodes it into uncompressed PCM signed 16-bit little-endian audio (pcm_s16le) wrapped in a WAV container. This means the compressed audio is fully decoded and stored as raw waveform data — there is no video processing overhead, and the resulting WAV file will typically be significantly larger than the audio portion of the original AVI because compression is removed. The 16-bit / little-endian PCM format is the standard used by audio CDs and is universally compatible with audio editing software, DAWs, and broadcast systems.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg application. When running in-browser, this is executed via FFmpeg.wasm, a WebAssembly port that runs the full FFmpeg engine locally in your browser without any server involvement. |
-i input.avi
|
Specifies the input file — an AVI container that may hold a video stream alongside a compressed audio track (commonly MP3, AAC, or Vorbis). FFmpeg will demux both streams and use only the audio for this conversion. |
-c:a pcm_s16le
|
Sets the audio codec to PCM signed 16-bit little-endian, which is the standard uncompressed audio encoding used in WAV files. This decodes whatever compressed audio format was stored in the AVI and writes it as raw waveform data — the result is fully lossless from this point forward, containing no compression artifacts of its own. |
output.wav
|
Defines the output file as a WAV container. FFmpeg infers the container format from the .wav extension and wraps the pcm_s16le audio stream in a standard RIFF WAV file, which is immediately compatible with audio editors, DAWs, broadcast systems, and virtually every platform that handles audio. |
Common Use Cases
- Extracting a clean, uncompressed audio track from an old AVI home video to import into a DAW like Audacity or Adobe Audition for editing or restoration
- Pulling dialogue or narration from an AVI screen recording or lecture capture to use as source material for transcription or subtitling workflows
- Converting AVI audio to WAV for use in game engines like Unity or Unreal Engine, which prefer uncompressed PCM audio assets
- Archiving the audio from legacy AVI files in a lossless format before the original files degrade or become unplayable due to codec obsolescence
- Preparing audio from AVI broadcast recordings for ingestion into a professional audio system or playout server that requires WAV/PCM input
- Extracting a music or sound effect track from an AVI video clip to use in a video editor's audio timeline without recompression artifacts
Frequently Asked Questions
Not exactly — the WAV file will not recover quality that was already lost when the audio was originally compressed inside the AVI. If the AVI stored MP3 audio at 128kbps, that lossy compression happened at encode time and cannot be reversed. What you get is a lossless, uncompressed copy of whatever audio data exists in the AVI — no further quality degradation occurs during this conversion, and no additional compression artifacts are introduced.
AVI files typically store audio as compressed MP3 or AAC streams, which use lossy compression to reduce file size. WAV with pcm_s16le encoding stores every audio sample as raw, uncompressed 16-bit integers — roughly 10MB per minute for stereo audio at 44.1kHz. A 5-minute AVI with 128kbps MP3 audio might have about 4.8MB of compressed audio, but the equivalent WAV would be around 50MB. This is expected and is the nature of uncompressed audio.
By default, FFmpeg selects the first audio stream it finds in the AVI file. AVI does technically support multiple audio tracks, though this is uncommon in practice. If your AVI has multiple audio streams and you need a specific one, you can add a stream selector to the command, such as -map 0:a:1 to select the second audio track before the output filename.
AVI metadata is generally minimal and the WAV container has very limited metadata support compared to formats like FLAC or MP3. Most AVI files carry little meaningful audio metadata beyond technical stream properties. FFmpeg will attempt to copy any available metadata tags, but in practice you should not expect artist, title, or album information to carry over reliably when going from AVI to WAV.
Yes. The default command uses pcm_s16le which produces 16-bit WAV, but you can substitute a different PCM codec for higher bit depth. Replace -c:a pcm_s16le with -c:a pcm_s24le for 24-bit audio or -c:a pcm_s32le for 32-bit, both of which are supported by the WAV format. This can be useful when the source AVI contains high-quality audio and you're delivering to a professional audio workflow that requires 24-bit masters.
On Linux or macOS, you can run a shell loop: for f in *.avi; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.avi}.wav"; done. On Windows Command Prompt, use: for %f in (*.avi) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This applies the same pcm_s16le conversion to every AVI in the current directory and names each output WAV after the original file.
Technical Notes
The pcm_s16le codec selected here — PCM signed 16-bit little-endian — is the most universally compatible WAV encoding and matches the CD audio standard (16-bit, typically 44.1kHz stereo). The sample rate of the output WAV will match whatever sample rate was present in the AVI's audio stream; FFmpeg does not resample unless explicitly instructed. If the AVI contains audio at 48kHz (common in video production), the WAV will also be 48kHz, which is perfectly valid but worth noting if your target system expects 44.1kHz. AVI does not support subtitles or chapters, so there is nothing to lose on that front. One known limitation: AVI's audio interleaving means that in malformed or very old AVI files, audio sync may already be imperfect — FFmpeg will extract exactly what is in the file, so any pre-existing sync issues are preserved rather than corrected. Since WAV does not support multiple audio tracks, only the primary selected audio stream is written to the output file.