Convert FLV to AIF — Free Online Tool

Extract and convert audio from FLV video files into AIF format, producing uncompressed PCM audio stored in Apple's Audio Interchange File Format. This tool decodes the AAC or MP3 audio stream embedded in your FLV container and re-encodes it as 16-bit big-endian PCM — the lossless, uncompressed format natively supported on macOS.

FFmpeg Command

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Estimated output:

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How It Works

FLV files typically carry audio encoded in a lossy codec — either AAC or MP3 — inside Adobe's Flash Video container. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio into raw PCM samples, then writes them out as 16-bit big-endian PCM (pcm_s16be) wrapped in an AIFF container. Because the source audio is lossy and the output is lossless uncompressed, the conversion does not recover any audio detail lost during the original FLV encoding — but it does produce a fully uncompressed file with no further generation loss, which is ideal for archiving or importing into audio editing software on Mac.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg program. In this browser-based tool, it runs via FFmpeg.wasm (WebAssembly) entirely in your browser — no server is involved. The same command works identically in a local desktop FFmpeg installation.
-i input.flv Specifies the input file, an FLV (Flash Video) container. FFmpeg will read both the video and audio streams from this file, though only the audio will be used in the output since AIF is an audio-only format.
-c:a pcm_s16be Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding used inside AIF files. This decodes the lossy AAC or MP3 audio from the FLV and writes it as raw, uncompressed samples in the byte order required by the AIFF specification.
output.aif Defines the output filename with the .aif extension, which tells FFmpeg to wrap the PCM audio in an AIFF container. AIFF is Apple's lossless uncompressed audio format, natively compatible with macOS and professional audio software including Logic Pro and GarageBand.

Common Use Cases

  • Extracting a recorded Flash-era webcast or lecture from an FLV archive to edit and remaster the audio in Logic Pro or GarageBand on macOS
  • Pulling audio from a downloaded FLV music video for import into an Apple-native DAW that prefers AIF over compressed formats
  • Archiving audio from legacy FLV screencasts or tutorials before Flash Player support ends on systems, preserving content in a durable uncompressed format
  • Preparing audio from an FLV interview recording for use in a professional audio post-production pipeline that requires uncompressed source files
  • Converting FLV game stream recordings to AIF to isolate and edit commentary audio without introducing additional lossy compression artifacts

Frequently Asked Questions

No — the AIF output will not be higher quality than the audio that was encoded inside the FLV. FLV files store audio in lossy formats like AAC or MP3, and any quality loss from that original encoding is permanent. What this conversion does do is decode the audio to uncompressed PCM, meaning no further quality degradation occurs from this point forward. The result is a lossless container holding the best possible representation of your FLV's audio.
This is expected and is a direct consequence of switching from lossy compressed audio to uncompressed PCM. For example, a 128 kbps AAC stream in an FLV might expand to roughly 1.4 MB per minute, while 16-bit stereo PCM at 44.1 kHz in AIF uses approximately 10 MB per minute. The AIF file contains every audio sample as a raw number rather than a compressed approximation, which is why the file size increases dramatically.
No. AIF is a pure audio format and cannot contain video streams. FFmpeg automatically drops the video when writing to an audio-only output container like AIF. If you need to preserve the video, you should convert to a video format such as MP4 or MOV instead.
Replace pcm_s16be with pcm_s24be in the command: ffmpeg -i input.flv -c:a pcm_s24be output.aif. This writes 24-bit big-endian PCM into the AIF container, which is useful if your downstream workflow (such as a professional DAW session) expects 24-bit source files. Other options include pcm_s32be for 32-bit integer or pcm_f32be for 32-bit float, depending on your editing environment's requirements.
Yes. AIF (also written as AIFF) is an Apple-native format with deep support across macOS, Logic Pro, GarageBand, Pro Tools, Ableton Live, and Final Cut Pro. It is one of the original professional audio formats on the Mac platform and is treated as a first-class citizen in virtually every Apple audio and video application. Windows DAWs like Reaper and Cubase also support AIF without issue.
The single-file command shown is the basis for batch processing. On macOS or Linux you can loop over files in a shell: for f in *.flv; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.flv}.aif"; done. On Windows Command Prompt you can use: for %f in (*.flv) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". This applies the same pcm_s16be conversion to every FLV in the directory.

Technical Notes

The AIF format uses big-endian byte ordering, which is why the codec is pcm_s16be (signed 16-bit big-endian PCM) rather than the little-endian pcm_s16le used by WAV. FFmpeg handles the byte order automatically when you specify the .aif output extension. Because FLV does not support lossless audio codecs, all FLV audio arrives at this conversion already lossy-encoded — typically AAC at 96–192 kbps or MP3. The decoded PCM output will faithfully represent whatever audio quality exists in the source, including any lossy compression artifacts. Metadata support in AIF is limited compared to formats like FLAC or MP3; standard ID3-style tags embedded in the FLV are not reliably transferred to the AIFF container by FFmpeg, so if metadata preservation is critical you may need to add tags manually after conversion using a tool like Kid3 or exiftool. The default 16-bit depth is broadly compatible, but professional workflows often prefer 24-bit (pcm_s24be) to maintain headroom for further processing.

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