MyMediaTools

Extract Audio from FLV to AIFF — Free Online Tool

Extract audio from FLV (Flash Video) files and save it as a high-quality, uncompressed AIFF file using PCM audio encoding. This conversion trades the lossy AAC or MP3 audio compressed inside the FLV container for a lossless PCM representation, making it ideal for professional audio workflows on macOS.

FFmpeg Command

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Estimated output:

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How It Works

FLV files typically carry audio encoded with lossy codecs — most commonly AAC or MP3 — compressed for efficient streaming delivery via Adobe Flash Player. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio track, then re-encodes it as 16-bit big-endian PCM (pcm_s16be), which is the native uncompressed format inside an AIFF container. Because the source audio is lossy, this process does not recover detail that was lost during the original FLV encoding — the output is a lossless, uncompressed representation of what the FLV's audio track currently contains. The resulting AIFF file will be significantly larger than the original FLV but fully compatible with macOS audio tools, DAWs, and professional editing software.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg tool, which handles all decoding, stream processing, and encoding steps entirely within the browser via WebAssembly in this tool, or on your local machine when run from the command line.
-i input.flv Specifies the input FLV file. FFmpeg reads the Flash Video container, identifying both the video stream (typically H.264 or Sorenson Spark) and the audio stream (AAC or MP3) embedded inside it.
-vn Disables all video output, dropping the FLV's video stream entirely. Since AIFF is a pure audio format, this flag ensures FFmpeg doesn't attempt to process the video track at all, speeding up conversion and preventing errors.
-c:a pcm_s16be Encodes the output audio as 16-bit signed big-endian PCM, the standard uncompressed codec for AIFF files. This decodes the FLV's lossy AAC or MP3 audio and writes it as raw PCM samples — the native, editable format expected by macOS audio applications and DAWs.
output.aiff Specifies the output filename and tells FFmpeg to wrap the PCM audio in an AIFF container, Apple's uncompressed audio format. The .aiff extension triggers FFmpeg to apply the correct big-endian AIFF container structure compatible with macOS, Logic Pro, and Final Cut Pro.

Common Use Cases

  • Recovering audio from archived Flash-era web videos or recorded livestreams saved as FLV files before importing into a macOS DAW like Logic Pro or GarageBand
  • Extracting voice-over or narration audio from FLV screen recordings or tutorials to re-edit or repurpose in a professional audio editing workflow
  • Pulling the audio track from FLV-format video files captured by older streaming software (e.g., OBS older versions, Camtasia) for use in Final Cut Pro, which handles AIFF natively
  • Archiving the audio from FLV media collections to a more future-proof, uncompressed format before FLV players and decoders become even less supported
  • Extracting background music or sound effects stored in FLV format for use in video production projects requiring uncompressed audio sources
  • Converting FLV audio for use in Apple's hardware ecosystem — AirPlay, Logic Remote, or professional audio interfaces — which favor AIFF over web-streaming formats

Frequently Asked Questions

No — the AIFF output will not have higher quality than the audio already encoded in the FLV. FLV files store audio as lossy AAC or MP3, meaning some audio detail was permanently discarded during the original encoding. This conversion decodes that lossy audio and writes it as uncompressed PCM inside AIFF, which preserves every bit of what remains but cannot restore what was already lost. The benefit of AIFF is that no further quality degradation occurs in subsequent editing or re-exporting steps.
FLV files use lossy audio compression (AAC typically at 128kbps, or MP3) specifically to reduce file size for web streaming. AIFF with pcm_s16be stores raw, uncompressed audio samples — a 16-bit stereo signal at 44.1kHz consumes roughly 10MB per minute, compared to about 1MB per minute for 128kbps AAC. A 10-minute FLV might produce an AIFF file 8–10 times larger. This size increase is expected and is the cost of having an uncompressed, editable audio file.
pcm_s16be stands for 16-bit signed PCM with big-endian byte order, which is the standard bit depth for CD-quality audio and the AIFF default. Since FLV audio is typically encoded at or below 44.1kHz/16-bit quality, pcm_s16be accurately captures the full resolution of what the FLV contains. If you know your FLV source was encoded from a 24-bit original and you want to future-proof the file, you could change the codec flag in the FFmpeg command to -c:a pcm_s24be, but for most FLV sources this provides no practical benefit.
FLV files have limited and non-standardized metadata support, and metadata is often absent or minimal. FFmpeg will attempt to copy over any ID3-style or AMF metadata tags present in the FLV to the AIFF output, but AIFF's metadata support (via MARK and NAME chunks) is also limited compared to formats like MP3 or FLAC. In practice, you should expect little to no metadata in the output AIFF and may need to add tags manually using a tool like Meta or mp3tag after conversion.
To use a higher bit depth in your AIFF output, replace pcm_s16be in the command with another supported AIFF codec: pcm_s24be for 24-bit, pcm_s32be for 32-bit integer, or pcm_f32be / pcm_f64be for 32/64-bit floating point. For example: ffmpeg -i input.flv -vn -c:a pcm_s24be output.aiff. Note that the -b:a (bitrate) flag has no effect on PCM codecs since they are uncompressed — the file size is determined entirely by sample rate, bit depth, and duration.
Yes — on macOS or Linux, you can use a shell loop: for f in *.flv; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.flv}.aiff"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aiff". This is especially useful for archiving large FLV collections, since the browser-based tool handles files one at a time and is capped at 1GB.

Technical Notes

FLV is a legacy Adobe container that encodes audio as either AAC (MPEG-4 Audio) or MP3 via libmp3lame, both lossy codecs with a default bitrate around 128kbps in most web-streaming contexts. When extracting to AIFF, FFmpeg fully decodes the compressed audio stream before writing it as pcm_s16be — this means two codec stages occur (decode + encode), but since the output is uncompressed PCM, the re-encode stage is lossless by definition. The primary quality ceiling is therefore set by the FLV's original lossy encoding. AIFF uses big-endian byte order (inherited from Motorola 68k architecture, the basis of classic Macs), which is why the codec is pcm_s16be rather than the little-endian pcm_s16le used in WAV files — both represent identical audio data in different byte arrangements. AIFF has no support for multi-track audio, subtitles, or chapters, and neither does FLV, so no stream selection complexity is involved. One known limitation: if the FLV contains variable-frame-rate audio or non-standard sample rates, FFmpeg may resample or adjust timing during extraction, which could introduce very minor artifacts at transition points.

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