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Convert MKV to WAV Online

Extract audio from MKV files to uncompressed WAV. No servers.

Drag your file here

.mkv · up to 100 MB

Processed in your browser — file never uploadedFree
Note: The first conversion loads the FFmpeg engine (~25MB). Subsequent conversions will be faster.

MKV movie and anime audio to professional WAV

Music production sampling

Extract soundtracks, dialogue, and sound effects from movies and anime as uncompressed WAV for your beats.

100% private

Your MKV file never leaves your device. FFmpeg.wasm runs in WebAssembly in your browser.

DTS / AC3 / AAC → PCM

Decode any audio format inside the MKV — Dolby Digital, DTS, AAC, and more — to clean PCM WAV.

DAW-ready

Resulting WAV is compatible with Audacity, Pro Tools, Logic Pro, Ableton Live, and any professional DAW.

Three steps, no hassle

1

Upload your MKV file

Drag or select your .mkv file — movie, anime, series, or documentary. Up to 500 MB, no signup.

2

Audio decoding

FFmpeg.wasm extracts the first audio track from the MKV — whether DTS, AC3, AAC, or PCM — and decodes it to uncompressed WAV in your browser.

3

Download your WAV

Get the WAV file ready to import into your DAW (Audacity, Pro Tools, Logic Pro, Ableton Live) or for sampling.

Got questions?

MKV (Matroska Video) is an open-source multimedia container created in 2002 by Steve Lhomme and the Matroska community. Unlike MP4 or AVI, MKV can hold an unlimited number of video, audio, and subtitle tracks in multiple languages without modifying the file. That is why it is the preferred format for anime releases (with subtitles in multiple languages, Japanese and English audio tracks, and chapters) and high-quality movies with DTS-HD or Dolby TrueHD. MKVToolNix and FFmpeg are the reference tools for working with this format.

FFmpeg can decode any audio track found in the MKV container: PCM (already uncompressed), FLAC (lossless), AAC-LC and HE-AAC, MP3, AC3/Dolby Digital (5.1), E-AC3/Dolby Digital Plus, DTS, DTS-HD Master Audio (lossless), Dolby TrueHD, and Opus. In all cases the result is a stereo or multichannel WAV file with linear PCM at the original sample rate and bit depth.

The current version automatically extracts the first audio track detected in the MKV file. In files with multiple tracks (e.g., Japanese audio as track 0 and English as track 1 in an anime), track 0 is extracted. If you need a specific track other than the first, you can pre-remux the MKV with MKVToolNix to reorder the tracks, or use FFmpeg from the command line with -map 0:a:1 to select the second audio track.

Yes. FFmpeg respects the source sample rate (typically 44.1 kHz for standard video content, 48 kHz for cinema and broadcast, up to 96 kHz or 192 kHz for high-resolution audio) and bit depth (16-bit for most content, 24-bit for cinema audio like DTS-HD or Dolby TrueHD). The resulting WAV is a faithful PCM representation of the original audio without any re-encoding that would introduce additional quality loss.

The most common use cases are: music production (sampling phrases, sound effects, or movie and anime soundtracks for beats or compositions), professional audio editing (importing dialogue or effects into Pro Tools, Reaper, or Nuendo to sync with other productions), audio transcription (speech recognition systems like OpenAI's Whisper work best with uncompressed WAV), forensic audio analysis, and preserving high-quality audio tracks like DTS-HD extracted from Blu-ray rips.

WebAssembly processing inside the browser has available RAM limitations (typically 2-4 GB depending on the device and browser). For 20-50 GB MKV files (Blu-ray rips in H.265 with DTS-HD audio), conversion may fail due to memory exhaustion. For these cases, the recommended alternative is using FFmpeg directly from the command line: ffmpeg -i input.mkv -vn -acodec pcm_s16le -ar 48000 output.wav, which processes the file sequentially without loading everything into RAM.

Convert MKV to WAV: extract Matroska audio uncompressed for production and editing

The Matroska Video format (MKV) is the most versatile open-source multimedia container available in 2025. Created in 2002 by Lasse Kärkkäinen and Steve Lhomme as a free alternative to proprietary formats AVI, MOV, and MP4, MKV can hold an unlimited number of video, audio, subtitle, and chapter tracks in a single file with no restrictions on codec, channel count, or language. This flexibility has made it the de facto standard for anime distribution (with Japanese and English audio tracks, multilingual subtitles, and fansubs), Blu-ray rips in H.264 or H.265 with DTS-HD Master Audio or Dolby TrueHD, and any high-quality multimedia content requiring multiple simultaneous tracks. The MKV container specification is maintained by the Matroska Foundation and documented in IETF RFC 8794, published in 2021.

Extracting audio from an MKV file to uncompressed WAV format is an essential operation for music producers, audio editors, sound engineers, and any professional who needs to work with audiovisual content audio in a professional production environment. The WAV format (Waveform Audio File Format), specified by Microsoft and IBM in 1991 as an extension of the RIFF format, is the most universal uncompressed audio container in the industry: 100% compatible with all professional DAWs — Pro Tools, Logic Pro, Ableton Live, Cubase, Reaper, Nuendo, FL Studio — with all speech recognition systems, with audio analysis tools like Adobe Audition, iZotope RX, and Sonic Visualiser, and with any production pipeline requiring linear PCM audio without codec artifacts. When audio inside the MKV is encoded in DTS, DTS-HD, Dolby Digital (AC3), E-AC3, AAC, or FLAC, converting to PCM WAV eliminates any dependency on proprietary decoders at the destination.

The technical process of extracting MKV audio to WAV involves three phases: demultiplexing the Matroska container to identify and separate audio tracks (using libmatroska, the reference library integrated in FFmpeg), decoding the compressed audio codec (libavcodec supports all relevant codecs: libdca for DTS and DTS-HD, liba52 for AC3 and E-AC3, the native FFmpeg implementation for AAC, libflac for FLAC), and writing the resulting PCM data into a RIFF/WAV container with the fmt chunk specifying the sample rate, bit depth, and channel count of the original audio. FFmpeg preserves the original sample rate (48 kHz is standard for cinema and broadcast content, 44.1 kHz for CD-derived content, up to 96 kHz for Blu-ray high-resolution audio) and bit depth (16-bit or 24-bit depending on the source codec). Convertir.ai runs this entire process in WebAssembly with FFmpeg.wasm directly in the user's browser, without transmitting any fragment of the MKV file to external servers.