Convert AVI to FLAC Online
Extract audio from AVI files to lossless FLAC. Preserve original PCM audio or convert MP3/AC-3 to FLAC for permanent archiving. Free, in your browser.
.avi · up to 100 MB
What you can do
AVI to FLAC: preserve audio from the DivX era and old recordings
PCM to FLAC losslessly
Uncompressed PCM audio from AVI converts to FLAC keeping every original sample.
No additional loss
MP3/AC-3 is decoded and re-compressed in FLAC, avoiding a second generation of loss.
100% private
Your AVI never leaves your device. FFmpeg.wasm processes everything locally.
Historical DivX files
Compatible with AVI from the DivX/XviD era, VirtualDub captures, and Fraps recordings.
How it works
Three steps, no hassle
Upload your AVI file
Drag or select your .avi with PCM, MP3, AC-3, or DTS audio. Up to 500 MB, no signup.
Extraction to FLAC
FFmpeg.wasm analyzes the RIFF/AVI chunk to identify the audio codec. PCM and LPCM are compressed to FLAC losslessly. MP3/AC-3 are decoded and re-encoded to FLAC (no additional loss, even if the source was already lossy).
Download your FLAC
FLAC file ready for audiophile players, DAWs, or digital archiving. Download with one click.
FAQ
Got questions?
Yes. The AVI (Audio Video Interleave) format, defined by Microsoft in 1992, can encapsulate uncompressed PCM audio (wFormatTag=0x0001 in the WAVEFORMATEX header), which is strictly lossless. Many old video captures (analog capture cards from the 1990s and 2000s), DV camera recordings, and some screen recording software saved audio as PCM at 44.1 kHz/16-bit or 48 kHz/16-bit directly in AVI. Converting that PCM to FLAC is a lossless-to-lossless operation: the decoded audio is bit-for-bit identical to the original.
It's important to be honest here: MP3 (wFormatTag=0x0055) and AC-3 (wFormatTag=0x2000) are lossy codecs. If you convert MP3 or AC-3 to FLAC, you get a larger file than the original MP3, but with exactly the same quality (neither better nor worse). FLAC accurately preserves what the MP3 or AC-3 decoder produces, without adding further loss. This is useful if you plan future edits or transcodings: working from FLAC avoids accumulating a second generation of loss.
In the DivX era (1999–2008), DivX codec for video and MP3 for audio became the de facto standard for sharing movies on the internet. DivX 3.11 (a hack of Microsoft's MPEG-4 v3 codec) and later DivX 4 and 5 used the AVI container with 128 kbps MP3 audio as the most efficient combination for 700 MB CD-Rs. Billions of files from this era have this structure. MP3 quality was typically 128 kbps CBR, adequate for movies but not for high-quality archiving.
Significantly more. A 128 kbps MP3 for a 90-minute movie occupies approximately 86 MB. The FLAC resulting from decoding that MP3 to PCM and re-compressing with FLAC will occupy between 500 MB and 1 GB, because FLAC preserves all PCM samples (including MP3 artifacts). This is disadvantageous in terms of space but is the only way to avoid additional loss in future transcodings. If you only need to play or edit the audio once, keeping MP3 is perfectly valid.
There are several legitimate use cases: preserving personal recordings (weddings, family concerts) captured in AVI with PCM audio that you want to archive in a modern, well-supported format; audio restoration (importing the FLAC into a DAW for noise cleaning, normalization, or remastering); integration into music collections cataloged in foobar2000 or Roon, which manage FLAC better than MP3 embedded in AVI; and preparation for speech recognition or automatic subtitling with tools that prefer FLAC as input.
Yes. VirtualDub (released by Avery Lee in 1998 and maintained through 2013) saved captures with uncompressed PCM audio in AVI. Fraps, popular for gameplay capture in the 2000s, also used AVI with PCM audio at 48 kHz/16-bit stereo. These are exactly the cases where AVI to FLAC conversion is most beneficial: the original PCM audio is compressed to FLAC completely losslessly, reducing audio file size while maintaining every original sample.
Convert AVI to FLAC: preserve audio from DivX files, old captures, and historical recordings
Converting AVI to FLAC is the solution for preserving audio from AVI (Audio Video Interleave) video files, the most historically relevant multimedia container in Windows history from its introduction by Microsoft in November 1992 through its displacement by MKV and MP4 in the latter half of the 2000s. The AVI format uses the RIFF (Resource Interchange File Format) structure with chunks identified by FourCC codes: the LIST hdrl chunk contains the video and audio headers, and the movi chunks contain the interleaved multimedia data. The audio codec description is stored in the WAVEFORMATEX structure with the wFormatTag field identifying the codec: 0x0001 for uncompressed PCM, 0x0055 for MPEG Layer III which is MP3, 0x2000 for AC-3 or Dolby Digital, and 0x0162 or 0x0163 for Windows Media Audio Pro. PCM audio in AVI with wFormatTag=0x0001 is strictly lossless: each audio sample is stored as an 8, 16, or 24-bit integer with no psychoacoustic transformation and absolutely no spectral information discarding of any kind. Analog video captures from the 1990s via capture cards like Pinnacle, Matrox, and Canopus, DV-AVI recordings from miniDV cameras, and gameplay captures from tools like VirtualDub and Fraps all stored audio in exactly this uncompressed PCM format. Converting that PCM to FLAC is a completely lossless operation that simply repackages the samples into a modern container with efficient compression, reducing audio file size while preserving every single original bit of the recording with absolute mathematical certainty that can be verified at any time in the future. The FLAC STREAMINFO block includes an MD5 checksum of the decoded audio that can be re-verified years later to confirm the archive has not been silently corrupted, which is a capability that no lossy format like MP3 or AAC can offer.
The historical context of AVI files with MP3 audio deserves special attention and honest treatment. Between 1999 and 2008, the combination of DivX or XviD video with MP3 audio in AVI container was the de facto worldwide standard for movie distribution on the internet. The DivX 3.11 codec, known informally as DivX with a wink, was a reverse-engineering of Microsoft's MPEG-4 v3 codec published in 2000, and combined with 128 kbps CBR MP3 audio it allowed compressing a 90-minute movie onto a 700 MB CD-R with acceptable quality. Scene distribution groups like Centropy, TEN, and MoS popularized this format which later evolved with XviD, the open-source fork of OpenDivX released in 2001, maintaining the same AVI plus MP3 structure throughout that entire era. The tool converts these files to FLAC completely honestly: FFmpeg's MP3 decoder decodes the audio to float PCM and the FLAC encoder compresses it losslessly without adding any additional degradation whatsoever. The resulting FLAC has exactly the same perceptual quality as the original MP3, neither better nor worse, because FLAC preserves exactly what the decoder produces but cannot recover information that MP3 already discarded. However, FLAC is the cleanest possible starting point for any subsequent processing such as DAW editing, audio restoration, normalization, or conversion to another delivery format, because it avoids accumulating a second generation of lossy degradation on top of what already exists in the source material. This principle, sometimes called the one-generation rule in audio archiving, is what makes FLAC the recommended intermediate format even when the original source was not lossless to begin with, and it applies equally to historical AVI files from the DivX era and to modern content.
Convertir.ai runs AVI to FLAC conversion entirely in the browser with FFmpeg.wasm compiled with all necessary decoders. Processing begins by analyzing the AVI's RIFF structure: the parser reads the LIST hdrl chunk, within which the avih subchunk or AVI Main Header provides global parameters such as total file duration and frame rate per second, and the audio strl subchunk contains the strh or Stream Header with fccHandler indicating the codec, and the strf or Stream Format with the WAVEFORMATEX structure specifying wFormatTag, nChannels, nSamplesPerSec, nAvgBytesPerSec, and wBitsPerSample precisely. With this complete information, FFmpeg selects the appropriate decoder for each case: for wFormatTag=0x0001 which is pure PCM, there is no real decoding, just direct reading of integer samples with absolutely no mathematical transformation applied to the data. For 0x0055 which is MP3, the mp3float decoder in libavcodec is used, implementing the hybrid filter bank synthesis with MDCT transform described in the ISO/IEC 11172-3 standard published in 1993. For 0x2000 which is AC-3 or Dolby Digital, the ac3 decoder is used with support for automatic downmix from 5.1 or 7.1 configurations to stereo when needed. The decoder output in all cases is 32-bit float PCM that passes to the FLAC encoder in libavcodec, implementing adaptive linear prediction with automatic optimal LPC order selection per block and Rice coding with adaptive partition order based on spectral content. The default compression level is 5, providing the best balance between encoding speed in the WebAssembly environment and the resulting compression ratio of the FLAC output file.