Convert MKV to WAV — Free Online Tool
Extract and convert the audio from an MKV file into an uncompressed WAV file using the PCM 16-bit little-endian codec. Whether your MKV contains AAC, MP3, Opus, or any other compressed audio, this tool decodes it and outputs a lossless, broadcast-ready WAV that works with virtually every audio editor, DAW, and professional workflow.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your MKV file here
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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 typically store audio in a compressed format such as AAC or Opus. During this conversion, FFmpeg discards all video, subtitle, and chapter streams entirely and focuses solely on the first audio track. That compressed audio is fully decoded and then re-encoded as raw PCM signed 16-bit little-endian samples — the standard uncompressed format used inside WAV files. There is no audio quality ceiling imposed by a target bitrate; instead, the decoded audio is written sample-for-sample into the WAV container, making the output as faithful to the decoded source as 16-bit PCM allows. File size will increase substantially compared to the compressed MKV audio because WAV stores every sample without compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion both in the browser via WebAssembly and on your local desktop command line. |
-i input.mkv
|
Specifies the input file — in this case an MKV container, which may contain video, one or more audio tracks (AAC, Opus, FLAC, etc.), subtitles, and chapter markers. |
-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 (e.g., AAC or Opus) was stored in the MKV and writes raw PCM samples into the output. |
output.wav
|
Defines the output filename and format. The .wav extension tells FFmpeg to wrap the PCM audio in a WAV container — a format with near-universal compatibility across operating systems, audio editors, and broadcast equipment. All video and subtitle streams from the MKV are automatically excluded because WAV cannot hold them. |
Common Use Cases
- Import dialogue or music from an MKV video into a DAW like Audacity, Adobe Audition, or Pro Tools, which require uncompressed PCM audio for non-destructive editing
- Prepare a WAV master from an MKV-encoded film or TV episode for broadcast delivery, where WAV is mandated by the technical specification
- Extract the audio track from an MKV screen recording to clean up, normalize, and re-attach as a podcast episode or voiceover
- Convert an MKV music concert recording to WAV so it can be burned to an audio CD, which requires uncompressed 16-bit 44.1 kHz PCM
- Strip the audio from an MKV game capture to feed into a machine learning pipeline or speech recognition system that expects uncompressed WAV input
- Archive the audio component of an MKV documentary in a universally readable, non-proprietary format that will remain accessible without codec support decades from now
Frequently Asked Questions
The WAV output will be as accurate as a 16-bit PCM representation allows, but there is one unavoidable quality consideration: if the MKV's audio was already stored in a lossy format like AAC or Opus, those compression artifacts are baked in before the conversion begins. The WAV will not be 'better' than the compressed source — it will simply be a lossless container holding a decoded version of that already-lossy audio. If the MKV source audio was lossless (e.g., FLAC), the conversion to 16-bit PCM WAV is effectively transparent for most listeners, though a 32-bit floating-point WAV would preserve the full FLAC dynamic range.
WAV is a pure audio container with no support for video streams, subtitle tracks, chapter markers, or metadata beyond basic tags. FFmpeg automatically drops all non-audio streams during this conversion — no video is encoded, which is why the process is fast. If you need to keep the video and subtitles, you should instead extract only the audio track for editing and recombine it later.
By default, FFmpeg selects the first audio stream it finds in the MKV file, which is usually the primary language track. WAV only supports a single audio track, so only one stream is extracted. If you need a specific track — for example, the second language dub or a commentary track — you can modify the FFmpeg command by adding '-map 0:a:1' before the output filename to select the second audio stream (zero-indexed).
The MKV's audio was stored in a compressed codec like AAC or Opus, which typically achieves 10:1 to 20:1 compression ratios compared to raw PCM. WAV with pcm_s16le stores every audio sample uncompressed — a stereo 16-bit 44.1 kHz stream uses approximately 10 MB per minute. A 90-minute MKV with 192k AAC audio of roughly 130 MB could expand to a WAV of around 950 MB. This is expected and is the nature of uncompressed audio.
To change the bit depth, replace 'pcm_s16le' with another codec: 'pcm_s24le' for 24-bit (common in professional audio), 'pcm_s32le' for 32-bit integer, or 'pcm_f32le' for 32-bit floating point. To change the sample rate, add '-ar 44100' (or your desired rate) before the output filename — for example: 'ffmpeg -i input.mkv -c:a pcm_s24le -ar 48000 output.wav'. Audio CDs require 16-bit at 44.1 kHz, while professional broadcast typically uses 24-bit at 48 kHz.
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.mkv; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.mkv}.wav"; done'. On Windows Command Prompt, use: 'for %f in (*.mkv) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav"'. This processes each MKV in the current folder sequentially and saves a corresponding WAV alongside it. The in-browser tool handles one file at a time, but the FFmpeg command is ideal for bulk processing large collections on your local machine.
Technical Notes
The output codec pcm_s16le stands for PCM (Pulse-Code Modulation), signed, 16-bit, little-endian — which is the default and most universally compatible format inside a WAV container. It matches the standard used on audio CDs and is natively supported by every major operating system, DAW, and audio editor without requiring any additional codecs. The WAV format itself imposes a 4 GB file size limit due to its 32-bit chunk size header field; for very long recordings, consider using the RF64 or W64 extensions instead. FFmpeg preserves the source audio's channel layout (stereo, 5.1, etc.) and sample rate during conversion — it does not resample or downmix unless explicitly instructed. One notable limitation of WAV is that it does not support embedded album art, structured metadata, or lyrics tags in the way that MKV does; any rich metadata from the MKV will not survive the conversion. If the MKV contained a lossless FLAC audio track and you require full bit-depth preservation beyond 16 bits, consider using pcm_s24le or pcm_s32le to avoid any quantization rounding at the output stage.