Convert WEBA to AIFF — Free Online Tool
Convert WEBA audio files (WebM audio containers with Opus or Vorbis encoding) to AIFF, Apple's uncompressed PCM format. This tool decodes the lossy compressed web audio and writes it as 16-bit big-endian PCM — ideal for importing into macOS audio software like Logic Pro or GarageBand.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your WEBA file here
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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
WEBA files store audio using lossy codecs — typically Opus or Vorbis — compressed for efficient web delivery. Converting to AIFF requires fully decoding that compressed audio back to raw PCM samples, then writing them into the AIFF container as uncompressed 16-bit big-endian PCM (pcm_s16be). Unlike remuxing operations where streams are copied directly, this conversion involves actual decoding and re-encoding: FFmpeg decodes the Opus or Vorbis stream, reconstructs the PCM waveform, and writes it without any further compression. The resulting AIFF file will be substantially larger than the source WEBA file, and because the original source was lossy, the AIFF will represent the best-possible reconstruction of the audio — not a restoration of the original lossless recording.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In this browser-based tool, FFmpeg runs entirely via WebAssembly (FFmpeg.wasm) — no data leaves your device. The same command works identically on a desktop FFmpeg installation for files over 1 GB. |
-i input.weba
|
Specifies the input file — a WEBA audio file, which is a WebM container holding a compressed Opus or Vorbis audio stream. FFmpeg automatically detects the container and codec from the file's binary structure. |
-c:a pcm_s16be
|
Sets the audio codec to signed 16-bit big-endian PCM, which is the standard uncompressed audio encoding for AIFF files. This instructs FFmpeg to fully decode the lossy Opus or Vorbis stream and write raw PCM samples in the byte order required by the AIFF specification. |
output.aiff
|
Defines the output filename and format. The .aiff extension tells FFmpeg to wrap the pcm_s16be audio stream in an Audio Interchange File Format container, which is the uncompressed lossless format natively supported by macOS applications like Logic Pro, GarageBand, and QuickTime. |
Common Use Cases
- Importing a web-ripped audio clip into Logic Pro or GarageBand, which natively read AIFF but cannot open WEBA files
- Archiving a podcast or speech recording originally streamed in WebM/Opus format into an uncompressed format suitable for long-term macOS storage
- Preparing a WEBA audio asset for editing in Final Cut Pro or Audacity on macOS, where AIFF is a preferred interchange format
- Converting a browser-captured audio recording (saved as WEBA) into AIFF so it can be used as a sample in a digital audio workstation
- Delivering audio to a client or collaborator on macOS who requires uncompressed AIFF files rather than web-optimized formats
Frequently Asked Questions
No — and this is an important distinction. The AIFF container is lossless and uncompressed, but because the original WEBA audio was encoded with a lossy codec (Opus or Vorbis), some audio quality was permanently discarded at the time of that encoding. FFmpeg decodes the lossy stream as accurately as possible and writes the result as uncompressed PCM in the AIFF file, so there is no additional quality loss during this conversion. However, the AIFF will not sound better than the WEBA source; it simply stores what remains at full resolution.
WEBA files use Opus or Vorbis compression, which typically achieves 5:1 to 10:1 or greater reduction in file size compared to uncompressed audio. AIFF stores raw PCM samples with no compression at all — a 3-minute stereo audio file at 16-bit, 44.1 kHz takes roughly 30 MB as AIFF but may only be 2–4 MB as WEBA at 128k bitrate. This size increase is expected and unavoidable when converting from any compressed format to AIFF.
Yes — AIFF supports multiple PCM bit depths including pcm_s24be (24-bit), pcm_s32be (32-bit), pcm_f32be (32-bit float), and pcm_f64be (64-bit float). To use 24-bit in the FFmpeg command, you would change -c:a pcm_s16be to -c:a pcm_s24be. However, since the source WEBA audio is lossy, choosing a higher bit depth will not recover detail that was lost during the original Opus or Vorbis encoding — it simply stores the decoded samples in a larger container. For most use cases, 16-bit is sufficient.
WEBA files stored in the WebM container support a limited set of metadata tags. AIFF has its own metadata structure using TEXT chunks. FFmpeg will attempt to map compatible tags during the conversion, but WebM metadata support is not as rich as formats like MP3 or FLAC, so some tags may not transfer or may be silently dropped. If preserving specific metadata is important, you should verify the output AIFF tags after conversion using a tool like MediaInfo or exiftool.
You can use a shell loop to process multiple files. On macOS or Linux: for f in *.weba; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.weba}.aiff"; done. On Windows Command Prompt: for %f in (*.weba) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This runs the same conversion shown on this page for every WEBA file in the current directory, outputting a corresponding AIFF file for each.
AIFF is a big-endian format — a design choice Apple made when it was developed for Motorola 68k-based Macs. The 'be' in pcm_s16be stands for big-endian, meaning the most significant byte comes first in each audio sample. This is required for a valid AIFF file. The related format pcm_s16le (little-endian) is used in WAV files, which follow the Windows/Intel convention. FFmpeg correctly writes big-endian PCM when targeting AIFF, ensuring the output plays correctly in all AIFF-compatible applications.
Technical Notes
WEBA is an audio-only WebM container, and the audio inside is almost always Opus (standardized in RFC 6716), though Vorbis is also possible in older files. Opus is a modern codec optimized for speech and music at low bitrates, but its lossy nature means frequency information above certain thresholds is discarded during encoding. When FFmpeg decodes Opus or Vorbis to produce PCM output, it uses its internal libopus or libvorbis decoder to reconstruct sample values — these are accurate representations of the decoded signal but cannot restore pre-encoding content. The output codec pcm_s16be is signed 16-bit big-endian PCM, which supports a dynamic range of 96 dB and sample rates up to whatever the source provides (commonly 48 kHz for Opus, 44.1 kHz for Vorbis). If the WEBA source was encoded at 48 kHz (standard for Opus), the AIFF output will be at 48 kHz — FFmpeg does not resample unless explicitly instructed to. AIFF does not support transparency, subtitles, or multiple audio tracks, which is consistent with WEBA's own limitations, so no stream selection issues arise in this conversion. The -vn flag is not needed here since WEBA is audio-only, and FFmpeg handles the single audio stream automatically.