Extract Audio from AVI to WAV — Free Online Tool
Extract audio from AVI video files and save it as a lossless WAV file using PCM encoding. This tool strips the video stream entirely and decodes whatever compressed audio (MP3, AAC, or Vorbis) is inside the AVI container into uncompressed 16-bit PCM — giving you a clean, universally compatible WAV file with no generation loss from re-compression.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AVI file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
Settings
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 in a compressed format such as MP3 (libmp3lame), AAC, or Vorbis. During this conversion, FFmpeg discards the video stream entirely using the -vn flag, then decodes the compressed audio stream and re-encodes it as uncompressed PCM signed 16-bit little-endian audio (pcm_s16le) inside a WAV container. This is a lossy-to-lossless pipeline in the sense that the compressed audio is fully decoded to raw PCM — however, the original quality ceiling is still bounded by whatever compression the AVI's audio track used. The output WAV will be significantly larger than the audio stored inside the AVI, because WAV at pcm_s16le stores every sample without any compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles decoding the AVI container, demuxing its audio stream, and encoding the output WAV file entirely within your browser via WebAssembly. |
-i input.avi
|
Specifies the input AVI file. FFmpeg reads the AVI container, identifies the interleaved audio and video streams inside it, and makes them available for processing. |
-vn
|
Disables video output entirely — this is what makes it an audio extraction rather than a full conversion. The AVI's video stream (typically H.264, MJPEG, or PNG) is read and then discarded, so only the audio stream flows through to the WAV output. |
-c:a pcm_s16le
|
Sets the audio codec to PCM signed 16-bit little-endian, which is the standard uncompressed format used inside WAV files. FFmpeg fully decodes the compressed audio from the AVI (whether MP3, AAC, or Vorbis) and re-encodes it as raw PCM samples — no compression, no quality tradeoffs beyond what the source already had. |
output.wav
|
Specifies the output file. The .wav extension tells FFmpeg to wrap the pcm_s16le audio stream in a WAV container, writing the standard RIFF/WAV header with the correct sample rate, bit depth, and channel count inherited from the AVI's audio track. |
Common Use Cases
- Archiving the audio track from legacy AVI home videos or camcorder footage as a lossless WAV master before further editing
- Importing AVI dialogue or voiceover audio into a DAW like Audacity or Adobe Audition, which handles WAV natively without needing codec plugins for AVI's compressed audio
- Preparing audio extracted from AVI recordings for broadcast or podcast post-production workflows that require uncompressed WAV deliverables
- Recovering a clean audio track from an old AVI screen recording or lecture capture when the video is no longer needed
- Converting AVI game cutscene audio to WAV for use as sound assets in a game engine or media project that requires PCM input
- Extracting audio from AVI files received from legacy editing software (e.g., old VirtualDub or Windows Movie Maker exports) for use in modern workflows that no longer handle AVI reliably
Frequently Asked Questions
The WAV output will be uncompressed PCM, so no additional compression artifacts are introduced by this conversion. However, if the audio inside your AVI was already stored as MP3 or AAC, those compression artifacts are baked in — decoding to PCM does not recover information that was discarded during the original lossy encoding. Think of WAV here as a perfect preservation of whatever quality existed in the AVI's audio track.
AVI files typically store audio in a compressed format like MP3 or AAC, which can achieve 10:1 compression ratios or better. WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression at all. A 5-minute stereo audio track at 44.1 kHz will occupy roughly 50 MB as uncompressed WAV, compared to maybe 5 MB as 128kbps MP3 — so a 10x size increase is normal and expected.
No — WAV is a single-stream audio container and cannot hold multiple audio tracks. If your AVI file contains more than one audio track, FFmpeg will select the default (usually the first) audio track by default. If you need to extract a specific track, you can modify the command to add -map 0:a:1 (for the second audio track) before the output filename.
FFmpeg can decode all common AVI audio codecs — including MP3 (the most common), AAC, and Vorbis — and convert them to pcm_s16le WAV. It also handles older AVI audio formats like PCM itself, ADPCM, and AC-3, so virtually any AVI audio track you encounter should convert successfully.
The command uses -c:a pcm_s16le which produces standard 16-bit WAV. If you need higher bit depth for professional audio work, you can swap the codec to pcm_s24le for 24-bit or pcm_s32le for 32-bit. For example: ffmpeg -i input.avi -vn -c:a pcm_s24le output.wav. Note that the quality ceiling is still limited by the source audio in the AVI — using 24-bit output won't recover detail that was lost in the original compression.
Yes — on Linux or macOS you can loop over files with: for f in *.avi; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.avi}.wav"; done. On Windows Command Prompt, use: for %f in (*.avi) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". This applies the same extraction and PCM encoding to every AVI in the current directory.
Technical Notes
WAV with pcm_s16le is the most universally compatible PCM format — it is supported by every DAW, audio editor, game engine, and broadcast tool in existence. The 16-bit signed little-endian format matches the native byte order of x86 systems, making it the standard choice for general-purpose audio interchange. AVI's audio track metadata (such as sample rate and channel count) is preserved during decoding and written into the WAV header automatically by FFmpeg — so stereo 44.1 kHz AVI audio will produce a stereo 44.1 kHz WAV without any resampling. One important limitation: WAV does not support embedded chapter markers, subtitles, or multiple audio tracks, and AVI's own metadata fields (title, author, etc.) are not mapped to WAV headers during this conversion. If the AVI contains no audio track at all, FFmpeg will return an error — the -vn flag removes video but requires at least one audio stream to be present for the output to be written.