Convert FLV to AIFF — Free Online Tool
Extract and convert audio from FLV video files into AIFF, Apple's uncompressed lossless audio format. This tool decodes the AAC or MP3 audio track from your FLV container and re-encodes it as 16-bit big-endian PCM — the native AIFF codec — delivering a full-fidelity, uncompressed audio file ready for professional use on macOS.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your FLV file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
Settings
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
FLV files typically carry audio encoded in a lossy format — most commonly AAC or MP3 — wrapped inside Adobe's Flash Video container. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio back to raw PCM samples. Those samples are then written into an AIFF container using the pcm_s16be codec (16-bit signed big-endian PCM), which is the standard uncompressed format AIFF uses. Because the audio was originally lossy (AAC or MP3 in the FLV), the output AIFF will be lossless in the sense that no further compression is applied — but it cannot recover detail that was lost when the FLV was first encoded. The result is a large, uncompressed file that is ideal for editing in macOS-native audio software like Logic Pro or GarageBand.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In this browser-based tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no file is sent to a server. You can also run this exact command in a terminal on any desktop that has FFmpeg installed. |
-i input.flv
|
Specifies the input file — your FLV video. FFmpeg reads the Flash Video container and identifies both the video stream (typically H.264 or Sorenson Spark) and the audio stream (typically AAC or MP3) inside it. |
-c:a pcm_s16be
|
Sets the audio codec for the output to pcm_s16be — 16-bit signed big-endian PCM. This is the standard uncompressed audio encoding used by the AIFF format and is required for full compatibility with Apple's Core Audio system and macOS-native audio software. |
output.aiff
|
Defines the output filename and format. The .aiff extension tells FFmpeg to write an Audio Interchange File Format container. Because AIFF is a pure audio format, FFmpeg automatically drops the video stream from the FLV without any additional flags needed. |
Common Use Cases
- Extracting the audio from a legacy Flash-era video lecture or webinar to import into Logic Pro or Pro Tools for professional editing on macOS
- Archiving audio commentary or narration from old FLV screencasts into an uncompressed format before the original source is lost
- Preparing audio from FLV game recordings or streams for use in macOS video editing timelines that require AIFF assets
- Pulling the audio track from an FLV music performance or live recording to deliver a high-quality, uncompressed file to a mastering engineer
- Converting FLV audio to AIFF as an intermediate step before applying lossless processing or effects in a macOS DAW workflow
- Recovering usable audio from archived FLV files downloaded before Flash Player was deprecated, for storage in a long-term lossless archive
Frequently Asked Questions
Not in the purest sense. The AIFF file itself is uncompressed and introduces no new quality loss, but the audio in your FLV was already encoded in a lossy format (AAC or MP3) when the FLV was created. FFmpeg decodes that lossy audio to raw PCM and writes it uncompressed into AIFF — so you are preserving exactly what was in the FLV, without any additional degradation. You cannot reconstruct audio detail that AAC or MP3 compression discarded before you ever had the file.
FLV stores audio in a compressed format like AAC or MP3, which can reduce audio data by 90% or more compared to uncompressed PCM. AIFF stores raw, uncompressed 16-bit PCM samples — every sample takes its full space in the file. A one-minute FLV with 128 kbps AAC audio might produce an AIFF of roughly 10 MB, whereas the audio data in the FLV was only about 1 MB. This size increase is expected and is the nature of lossless, uncompressed storage.
FLV files can carry some metadata in their header, but AIFF uses a different metadata structure (ID3 tags or AIFF ANNO/NAME chunks), and FFmpeg does not always map FLV metadata fields cleanly to AIFF. In practice, most FLV files contain minimal audio metadata anyway. You should plan to add or verify tags manually in an audio editor like Foobar2000 or Kid3 after conversion if accurate metadata is important for your workflow.
Yes, by modifying the FFmpeg command. The default command uses pcm_s16be (16-bit), but AIFF also supports 24-bit (pcm_s24be), 32-bit integer (pcm_s32be), and 32- or 64-bit float (pcm_f32be, pcm_f64be). To get 24-bit output, for example, you would change the command to: ffmpeg -i input.flv -c:a pcm_s24be output.aiff. Keep in mind that since the FLV source was lossy, using a higher bit depth will not recover lost detail — it simply stores the decoded samples in a higher-precision container.
You can use a shell loop to process multiple files at once. On Linux or macOS, run: for f in *.flv; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.flv}.aiff"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This processes every FLV in the current directory and writes a matching AIFF file alongside it. The browser-based tool handles one file at a time, so the FFmpeg command is the recommended approach for bulk conversions.
AIFF has broad compatibility beyond the Apple ecosystem. Professional DAWs on Windows such as Adobe Audition, Reaper, and Steinberg Cubase all read AIFF natively. However, AIFF is not supported by most web browsers or consumer media players on Windows. If you need a lossless format with wider compatibility outside of professional audio tools, WAV (which uses the same PCM audio data but a different container) is often a better choice for cross-platform use.
Technical Notes
The FLV container was designed specifically for streaming over the web via Adobe Flash Player and supports only a limited set of audio codecs — primarily AAC and MP3 (libmp3lame), both of which are lossy. AIFF, by contrast, is a strict lossless container that stores raw PCM audio with no compression. The conversion command uses pcm_s16be — 16-bit signed integer samples stored in big-endian byte order — which is the AIFF specification's foundational codec and ensures maximum compatibility with macOS Core Audio APIs, Logic Pro, GarageBand, and other Apple-native tools. The video stream in the FLV is automatically dropped because AIFF is a pure audio format with no video track support. There is no need to pass an explicit -vn flag because FFmpeg detects that the output container cannot carry video and handles the stream mapping accordingly. One important limitation: FLV does not support subtitles or chapter markers, and AIFF supports neither either, so no metadata of those types is ever at risk of being lost. Sample rate is preserved from the source FLV during conversion — if the FLV audio was 44100 Hz stereo, the output AIFF will also be 44100 Hz stereo.