Convert WebM to FLAC — Free Online Tool

Extract and convert the Opus or Vorbis audio track from a WebM file into a FLAC archive — a lossless format that preserves every detail of the decoded audio. Ideal for archiving high-quality audio from WebM video without any further quality degradation.

FFmpeg Command

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How It Works

WebM files typically carry audio encoded in Opus or Vorbis — both lossy codecs. During this conversion, FFmpeg decodes the compressed audio stream back to raw PCM, then re-encodes it using the FLAC codec, which applies lossless compression. Because the source audio is already lossy, the FLAC output is a mathematically perfect capture of the decoded audio — no additional quality is lost in this step, but the original lossy compression artifacts from Opus or Vorbis remain. The video stream is dropped entirely since FLAC is an audio-only format. The result is a losslessly compressed audio file that is larger than the WebM's audio track but bitwise identical to the decoded signal, making it suitable for archiving, editing, or further processing without generational loss.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg command-line tool. In the browser version, this runs via FFmpeg.wasm compiled to WebAssembly, executing the same logic as the desktop binary without any server involvement.
-i input.webm Specifies the input WebM file. FFmpeg reads the Matroska-based container and identifies the available streams — typically a VP9 video track and an Opus or Vorbis audio track — before applying the output mapping.
-c:a flac Instructs FFmpeg to decode the WebM audio stream (Opus or Vorbis) to raw PCM and then re-encode it using the FLAC lossless codec. No audio data is lost in this encoding step — FLAC compression is mathematically reversible.
-compression_level 5 Sets the FLAC encoder's compression effort to level 5 on a scale of 0–8. This controls file size and encoding speed only; audio quality is identical at every level because FLAC is always lossless. Level 5 is the widely accepted default that balances speed and file size.
output.flac Defines the output filename and tells FFmpeg to use the FLAC container format. Since FLAC is audio-only, the VP9 video track and any subtitles or chapters present in the WebM file are automatically dropped.

Common Use Cases

  • Archive the audio from a WebM recording of a live concert or performance in a lossless container for long-term preservation before re-encoding to distribution formats
  • Extract dialogue or voiceover audio from a WebM screen recording to use as a clean, uncompressed source for podcast editing in a DAW
  • Pull the audio track from a WebM lecture or educational video into FLAC so it can be imported into audio editing software like Audacity or Adobe Audition without introducing additional codec roundtrips
  • Convert Opus-encoded WebM audio to FLAC to satisfy the requirements of a digital distribution platform or archival system that mandates lossless source files
  • Prepare audio from a WebM video clip for sample-accurate editing or time-stretching in a music production workflow where lossy intermediate formats would degrade quality further
  • Strip the audio from a large WebM file and store it as FLAC to save space compared to keeping the full video while retaining all decoded audio fidelity

Frequently Asked Questions

No — FLAC is a lossless format, but it cannot recover information that was already discarded by the Opus or Vorbis encoder when the WebM file was originally created. What FLAC guarantees is that no further quality is lost during this conversion. The output is a bit-perfect snapshot of the decoded audio, so it is ideal as an editing or archiving source but will not sound better than the original WebM audio track.
Opus and Vorbis are highly efficient lossy codecs that achieve very small file sizes by permanently discarding audio data. FLAC uses lossless compression, which preserves every sample of the decoded audio and therefore requires significantly more space — often several times larger than the Opus or Vorbis stream. The increase in size is expected and reflects the shift from lossy to lossless compression, not a problem with the conversion.
FFmpeg will attempt to copy compatible metadata from the WebM container to the FLAC file's Vorbis comment tags, which is the native metadata system for FLAC. Common fields such as title, artist, and date typically transfer correctly. However, WebM-specific metadata fields or embedded cover art may not carry over, so you should verify tags in a tool like MusicBrainz Picard or foobar2000 after conversion if accurate tagging is critical.
Both are discarded. FLAC is a pure audio format with no support for video streams, subtitles, or chapter markers. FFmpeg automatically drops non-audio streams when the output format cannot contain them. If you need to keep the video or subtitles, you would need to use a different output container such as MKV or MP4.
Replace the value after -compression_level in the command. Valid values are 0 through 8, where 0 applies no compression (fastest, largest files) and 8 applies maximum compression (slowest, smallest files). The default of 5 is a good balance for most use cases. Crucially, compression level does not affect audio quality in any way — FLAC is always lossless regardless of this setting; it only controls how hard the encoder works to reduce file size.
Yes. On Linux or macOS you can use a shell loop: for f in *.webm; do ffmpeg -i "$f" -c:a flac -compression_level 5 "${f%.webm}.flac"; done. On Windows Command Prompt, use: for %f in (*.webm) do ffmpeg -i "%f" -c:a flac -compression_level 5 "%~nf.flac". This is especially useful for large collections since the browser-based tool processes one file at a time.

Technical Notes

The FFmpeg FLAC encoder (-c:a flac) supports sample rates up to 655,350 Hz and bit depths of 16 or 24 bits. If the decoded Opus or Vorbis audio from the WebM is 48 kHz / 16-bit — the most common configuration for Opus — the FLAC output will faithfully represent that signal. Vorbis audio is also typically decoded to floating-point PCM and then quantized to 16- or 24-bit integer samples for FLAC storage; FFmpeg handles this automatically. The -compression_level parameter controls only the LPC order and search effort of the lossless encoder and has zero effect on decoded audio fidelity. Because WebM can contain multiple audio tracks, FFmpeg will by default map only the first audio stream to the output; if you need a specific track, add -map 0:a:1 (or the appropriate index) to the command. Replay Gain tags are not automatically calculated during conversion but can be added afterward using a tool like loudgain or the metaflac utility. FLAC cue sheet support is not applicable here since the source is a single-track media file rather than a disc image.

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