Convert WebM to FLAC Online
Convert audio from WebM files to lossless FLAC. Ideal for archiving browser recordings, preprocessing audio for speech recognition, and preserving maximum quality. Free, in your browser.
.webm · up to 100 MB
What you can do
WebM to FLAC: archive web recordings at maximum quality with no additional loss
No additional loss
FLAC preserves exactly the Opus/Vorbis decoder output without adding any degradation.
Ideal for Whisper and ASR
FLAC is the optimal input format for speech recognition APIs like OpenAI Whisper.
100% private
Your WebM never leaves your device. FFmpeg.wasm processes everything locally in the browser.
Professional DAW compatible
Pro Tools, Logic Pro, Ableton Live, and Reaper import FLAC directly for editing.
How it works
Three steps, no hassle
Upload your WebM file
Drag or select your .webm with Opus or Vorbis audio. Up to 500 MB, no signup.
Conversion to FLAC
FFmpeg.wasm decodes Opus or Vorbis audio from the WebM to PCM and compresses it to lossless FLAC. Note: if WebM audio is Opus (lossy), the FLAC wraps exactly what Opus produces, with no additional loss but without recovering what Opus already discarded.
Download your FLAC
FLAC file ready for audiophile players, DAWs, speech recognition APIs, or permanent archiving. Download with one click.
FAQ
Got questions?
This question deserves an honest and precise answer. FLAC is a lossless compression codec, meaning the produced FLAC is bit-for-bit identical to the PCM that the Opus decoder generates. However, Opus itself is a lossy codec: when the WebM was recorded, Opus discarded audio information from the original that cannot be recovered. The resulting FLAC perfectly preserves the Opus decoder's output, without adding any additional degradation. It's analogous to high-resolution scanning a photocopy: the scan is perfect, but the photocopy already had less detail than the original. If the WebM has Vorbis (also lossy), the same reasoning applies. No WebM with truly lossless audio exists because neither Opus nor Vorbis are lossless codecs.
There are several valid use cases. First, avoiding additional loss: if you plan to edit the audio in a DAW and then export it again (to MP3, AAC, or another format), starting from FLAC avoids accumulating a second generation of codec loss. Second, interoperability: FLAC is accepted by virtually all professional DAWs (Pro Tools, Logic Pro, Ableton Live, FL Studio), speech recognition tools (Whisper, Google Speech-to-Text), and archiving systems where WebM is not supported. Third, integrity verification: FLAC stores an MD5 hash of the decoded audio in the STREAMINFO block, allowing future verification that the file has not been corrupted. Fourth, library uniformity: keeping the entire collection in FLAC simplifies management in players like foobar2000 and Roon.
Yes, and it's one of the most valuable use cases. OpenAI Whisper accepts multiple audio formats including FLAC, WAV, MP3, M4A, and OGG. However, FLAC and WAV are preferable because they don't add compression artifacts that could confuse the model. Video conference recordings in WebM (Google Meet meetings, browser microphone recordings) converted to FLAC before passing to Whisper guarantee that the model receives exactly the audio the Opus decoder produces, without double compression. For Spanish voice transcription, Whisper large-v2 and large-v3 models have WER (Word Error Rate) below 5% with clean audio at 16 kHz mono, and FLAC at 16 kHz mono is the input format recommended by Whisper's documentation.
FLAC doesn't have a fixed bitrate like MP3 or Opus: it's lossless compression whose size depends on the audio's spectral complexity. For voice audio recorded with Opus at 128 kbps in WebM (48 kHz sample rate, 16-bit equivalent), the resulting FLAC will typically occupy between 300–600 kbps effective (depending on how much the voice spectrum varies). This is 2–5 times larger than the original Opus, but much smaller than uncompressed PCM (1,536 kbps for 16-bit/48 kHz stereo). For musical WebM audio at 192 kbps Opus, the FLAC can occupy 700–1,200 kbps effective.
Yes, fully. Pro Tools, Logic Pro, Ableton Live, FL Studio, Cubase, Studio One, and Reaper import FLAC without issues. The FLAC produced by this tool uses 16 or 24-bit depth depending on the resolution of the PCM produced by the decoder (Opus internally works at 16-bit/48 kHz for most profiles), and 48 kHz sample rate. Some DAWs may show a warning when importing 48 kHz FLAC if the project is configured at 44.1 kHz, and will ask for sample rate conversion confirmation. This is normal and does not affect quality.
Technically no. The WebM specification (based on Matroska with WebM profile) restricts audio codecs to Vorbis and Opus, both lossy codecs. Unlike the MKV container, which can encapsulate native FLAC (CodecID A_FLAC) or uncompressed PCM (A_PCM/INT/LIT), the WebM profile does not include these codecs. Thus, all .webm files in existence contain lossy audio. If you need truly lossless audio from a browser recording, the option would be to use the MediaRecorder API with the 'audio/wav' container (uncompressed PCM) instead of WebM, although WAV support in MediaRecorder is limited to Safari.
Convert WebM to FLAC: archive browser recordings at maximum quality with no additional loss
Converting WebM to FLAC is the quality preservation operation for those who need to work with browser recording audio in environments where FLAC is the standard including DAWs, transcription APIs, and archiving systems, without introducing additional degradation beyond what the original WebM's Opus or Vorbis codec already introduced at the time of recording. It is essential to understand the honest limitation of this conversion: WebM with Opus or Vorbis is always lossy audio, and FLAC will encapsulate the decoded PCM completely losslessly, but cannot recover the information that Opus or Vorbis discarded during the original encoding process. The Opus codec, standardized as RFC 6716 by the IETF in September 2012, is the technical result of successfully merging two previous audio projects: SILK (originally developed by Skype for low-latency voice communications) and CELT (developed by Xiph.Org for high-quality musical audio transmission). Opus uses MDCT for CELT mode and a combination of LPC (Linear Predictive Coding) with MDCT for SILK mode based on automatically detected content type. At 128 kbps, Opus produces quality equivalent to AAC at 256 kbps and Vorbis at 192 kbps according to multiple blind comparison studies from Hydrogenaudio and independent comparisons published in the official project forum since 2012. However, at those bitrates, it has already discarded spectral components that the human ear generally does not perceive in everyday casual listening. Converting Opus decoder output to FLAC preserves every PCM sample produced by the decoder, guaranteeing that no future transcoding adds additional loss on top of what already inevitably exists in the original source material recorded in the browser.
Use cases for WebM to FLAC concentrate in three main practical application areas. The first is preprocessing for automatic speech recognition (ASR). Meeting, interview, and conference recordings in WebM generated by browser-based video conferencing tools such as Google Meet, Jitsi Meet, BigBlueButton, and Microsoft Teams in web mode constitute a vast and growing source of audio for transcription in corporate and academic contexts in 2025. OpenAI Whisper, the most widely adopted speech recognition model for offline transcription, accepts FLAC, WAV, MP3, MP4, and OGG as direct input formats without additional conversion needed. FLAC is preferable to MP3 or OGG for this specific use because it adds no compression artifacts of its own to the audio: Whisper receives exactly the audio that the WebM's Opus decoder produced, without double compression that could introduce confusing artifacts for the recognition model. For corporate meeting transcription in Spanish, Whisper large-v3 models achieve WER below 8 percent with typical video conference recording quality at 128 kbps Opus. The second area is audio editing and post-production in a professional DAW. Podcasters and content producers who record remote interviews via browser and receive WebM files from their participants convert them to FLAC before importing into DAWs for editing. Starting from FLAC in the DAW and exporting to MP3 or AAC produces only one generation of loss in the final published file, instead of two generations if starting from the lossy WebM directly, which matters for maintaining the highest possible quality in the final product. A third application area involves digital forensics and legal transcription contexts, where FLAC's built-in MD5 integrity check provides a defensible chain of custody for audio evidence extracted from browser-recorded WebM files that would otherwise be difficult to authenticate.
Convertir.ai runs WebM to FLAC conversion entirely in the browser using FFmpeg.wasm compiled with all necessary audio decoders. Processing begins with EBML header analysis to confirm DocType webm and map the Segment blocks via the SeekHead index for efficient access without linear scanning. The audio track is identified in the Tracks block by TrackType=2 in the TrackEntry element, and the CodecID stored in that element determines the processing path to follow. For A_OPUS, FFmpeg's libopus decoder processes Opus packets identified by the first two bytes of the packet encoding the Table of Contents (ToC) with config, stereo flag, and frame count as specified in RFC 6716. Opus operates internally always at 48 kHz regardless of the original recording's sample rate, producing 32-bit float PCM at 48 kHz as the decoder output. For A_VORBIS, the libvorbis decoder processes packets following the complete Vorbis I specification, producing float PCM at the native sample rate of the stream declared in the Vorbis identification header. The float PCM produced by the decoder is converted to 16 or 24-bit integers via triangular TPDF (Triangular Probability Density Function) dithering to minimize quantization error and avoid non-random deterministic distortion patterns, and the FLAC encoder in libavcodec compresses it in 4096-sample blocks with adaptive-order linear prediction selected automatically by spectral analysis of each block. The output FLAC includes STREAMINFO with sample rate, bit depth, channels, total sample count, and MD5 hash of decoded audio for long-term integrity verification. Because the entire conversion runs in WebAssembly inside the browser with no server communication, the user's audio recordings remain completely private regardless of their content, which is particularly important for confidential business meetings and interviews recorded via browser-based conferencing tools.