Extract Audio from MKV to WAV — Free Online Tool
Extract audio from MKV video files and save it as a WAV file with uncompressed PCM audio. This tool demuxes the MKV container and re-encodes the audio stream to 16-bit signed PCM (pcm_s16le), producing a universally compatible lossless WAV file ready for audio editing, broadcast, or archiving.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your MKV 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
MKV files commonly carry audio encoded in lossy formats like AAC, MP3, Opus, or Vorbis — or lossless formats like FLAC. Because WAV's default codec is uncompressed 16-bit signed PCM (pcm_s16le), the audio stream must always be decoded and re-encoded regardless of the source codec. The video stream is discarded entirely using the -vn flag, so no video processing occurs. The result is a WAV file containing raw, uncompressed PCM audio — which is larger than the original audio track but free of any additional compression artifacts and compatible with virtually every audio tool and device in existence.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. All processing in this conversion — demuxing the MKV container, decoding the audio stream, and encoding to PCM WAV — is handled by FFmpeg running locally in your browser via WebAssembly. |
-i input.mkv
|
Specifies the input Matroska file. FFmpeg reads the MKV container and identifies all available streams — video, audio, subtitles, and chapters — before applying the output options that follow. |
-vn
|
Disables video output entirely. Since the goal is to extract only the audio, this flag tells FFmpeg to ignore all video streams in the MKV, which avoids any unnecessary video processing and ensures the output WAV contains only audio data. |
-c:a pcm_s16le
|
Sets the audio codec to 16-bit signed little-endian PCM, which is the standard uncompressed audio format used in WAV files. Regardless of whether the MKV's audio is AAC, Opus, FLAC, or any other codec, this decodes the source audio and re-encodes it as raw PCM samples — the native, uncompressed format that makes WAV universally compatible. |
output.wav
|
Defines the output filename and format. The .wav extension tells FFmpeg to wrap the PCM audio stream in a RIFF WAV container, which is the format expected by DAWs, broadcast tools, and audio editors that require uncompressed audio input. |
Common Use Cases
- Preparing audio from an MKV screen recording or lecture capture for import into a DAW like Audacity, Adobe Audition, or Pro Tools, which natively favor uncompressed WAV over compressed MKV-embedded audio
- Extracting a lossless audio master from an MKV film or TV episode for archiving or manual re-encoding at a later stage without any generation loss
- Delivering broadcast-ready audio extracted from an MKV source file to a client or broadcaster that requires WAV format for compliance reasons
- Stripping the audio from a large MKV video game recording or cutscene to use as raw material for sound design, sampling, or remixing
- Extracting dialogue or music from an MKV file to run through automated transcription or audio analysis tools that require uncompressed WAV input
- Converting an MKV podcast recording that was captured with a video component into a clean WAV audio file for distribution or editing
Frequently Asked Questions
It depends on the audio codec inside the MKV. If the source MKV contains lossless audio such as FLAC, the conversion to PCM WAV is fully lossless — the audio is decoded from FLAC and stored as uncompressed PCM with no quality degradation. However, if the source MKV contains lossy audio like AAC, Opus, Vorbis, or MP3, the existing compression artifacts are already baked in. Decoding a lossy stream to PCM WAV does not recover that lost information, but it also does not introduce any new artifacts — the WAV will be a transparent representation of whatever the lossy source sounded like.
The audio inside an MKV is almost always stored in a compressed format — AAC at 128k, for example, stores audio at a fraction of its uncompressed size. WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression, so a stereo 44.1kHz stream consumes roughly 10 MB per minute. A one-hour MKV with 128k AAC audio might have an audio track of around 55 MB, while the equivalent WAV would be approximately 600 MB. This size increase is expected and is the trade-off for having uncompressed, universally readable audio.
WAV has very limited metadata support compared to MKV. The WAV format can carry some basic metadata via its INFO chunk (fields like title, artist, and comment), but FFmpeg does not automatically map MKV metadata tags into WAV INFO chunks in all cases. Chapter markers, subtitle tracks, and multiple audio track information from the MKV are all lost, as WAV supports none of those features. If preserving rich metadata is important, consider extracting to FLAC instead, which supports extensive tagging while still being lossless.
By default, FFmpeg selects the first audio stream in the MKV, which is typically the default-flagged track. If your MKV has multiple audio tracks — for example, a director's commentary track alongside the main audio — only one will be extracted. To extract a specific track, you can modify the command to include a stream selector such as -map 0:a:1 to pick the second audio stream. The browser tool extracts the default track automatically, but the displayed FFmpeg command can be modified locally for multi-track scenarios.
The default output is 16-bit signed PCM (pcm_s16le), which is CD-quality and the most widely compatible WAV variant. If you need higher bit depth for professional audio work — such as 24-bit or 32-bit — you can replace pcm_s16le in the command with pcm_s24le for 24-bit or pcm_s32le for 32-bit integer PCM. For floating-point PCM, use pcm_f32le. The full modified command for 24-bit output would be: ffmpeg -i input.mkv -vn -c:a pcm_s24le output.wav. Note that these higher bit-depth formats are still lossless but produce proportionally larger files.
The single-file command shown is the foundation for batch processing. On Linux or macOS, you can loop over all MKV files in a directory with: for f in *.mkv; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.mkv}.wav"; done. On Windows Command Prompt, the equivalent is: for %f in (*.mkv) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". The browser-based tool processes one file at a time, so the local FFmpeg command is particularly valuable when you need to convert an entire folder of MKV recordings at once.
Technical Notes
WAV with pcm_s16le is the most broadly compatible audio format in existence — supported by every major DAW, video editor, broadcast system, and consumer device. The default 16-bit depth matches CD standard and is appropriate for virtually all source material originating from consumer or prosumer MKV recordings. MKV's audio codec diversity means the input could be anything from lossy AAC at 64k to lossless FLAC at 24-bit, and FFmpeg handles the decode-to-PCM step transparently in all cases. One important limitation: WAV files cannot store more than approximately 4 GB of audio data due to the 32-bit file size field in its RIFF header — for very long recordings, FFmpeg will either warn or produce a malformed header; in such cases, splitting the source MKV first or using a format like W64 or RF64 is advisable. Subtitle tracks, chapter markers, and secondary audio streams present in the MKV are not carried over, as WAV is a single-stream audio-only container with no support for these features.