Extract Audio from FLV to AIFC — Free Online Tool
Extract audio from FLV (Flash Video) files and save it as AIFC using a lossless PCM S16BE codec — ideal for archiving or professional audio workflows. The conversion strips the video stream entirely and re-encodes the AAC or MP3 audio from the FLV container into a 16-bit big-endian PCM AIFC file, preserving audio fidelity without lossy compression.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
FLV files typically carry lossy audio encoded as AAC or MP3 alongside a video stream. During this conversion, FFmpeg discards the video stream entirely (-vn) and decodes the compressed audio from the FLV container, then re-encodes it as PCM S16BE (signed 16-bit big-endian pulse-code modulation) — the default uncompressed audio codec for AIFC. This means the lossy AAC or MP3 audio is decoded to raw PCM samples and written into the AIFC container, which is an Apple-defined extension of AIFF that supports both compressed and uncompressed audio. The result is an uncompressed audio file suitable for professional editing, archival, or use in Apple-ecosystem tools. Note that because the source audio in FLV is lossy, the AIFC output will not recover any quality lost during the original FLV encoding — it simply stores the decoded audio losslessly from that point forward.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool — the underlying engine that powers this browser-based converter via WebAssembly. The same command can be run locally on your desktop if you have FFmpeg installed. |
-i input.flv
|
Specifies the input Flash Video file. FFmpeg will detect the FLV container and identify the audio stream (typically AAC or MP3) and video stream inside it. |
-vn
|
Stands for 'video none' — this flag tells FFmpeg to completely ignore and discard the video stream from the FLV file. Since we are extracting audio only into AIFC, no video data is written to the output. |
-c:a pcm_s16be
|
Sets the audio codec to PCM S16BE (signed 16-bit big-endian pulse-code modulation), which is the standard uncompressed audio codec for the AIFC format. This decodes the lossy AAC or MP3 audio from the FLV and stores it as raw, uncompressed samples in the Apple big-endian PCM format. |
-b:a 128k
|
Specifies a target audio bitrate of 128kbps. For the lossless pcm_s16be codec, this flag has minimal practical effect since PCM bitrate is fixed by sample rate and bit depth — it is included for command consistency but the actual output bitrate will reflect the source audio's sample rate and channel configuration. |
output.aifc
|
Defines the output filename and tells FFmpeg to write the result as an AIFC file. The .aifc extension signals FFmpeg to wrap the PCM S16BE audio data in the AIFC container format, which is Apple's extension of AIFF that supports both compressed and uncompressed audio codecs. |
Common Use Cases
- Extracting the audio track from a Flash-era video file (e.g., an old web lecture or archived stream) for use in a professional DAW on macOS, which natively reads AIFC files
- Archiving the audio content of FLV recordings to a stable, uncompressed format before FLV playback support disappears from modern browsers and players
- Preparing audio from an FLV screen recording or webinar for editing in Final Cut Pro or Logic Pro, both of which handle AIFC natively
- Converting Flash video audio tracks to PCM AIFC for use in broadcast or post-production workflows that require uncompressed, big-endian audio files
- Extracting a music or voice-over track embedded in an FLV file and storing it in a format compatible with classic Mac and professional audio archival standards
- Stripping video from an FLV file to produce a clean AIFC audio deliverable for a client whose workflow requires Apple-format uncompressed audio
Frequently Asked Questions
No — the audio quality cannot improve beyond what was captured in the original FLV file. FLV containers store audio as AAC or MP3, both of which are lossy formats. When FFmpeg decodes this audio and re-encodes it as uncompressed PCM S16BE in the AIFC container, it preserves the decoded signal exactly, but any quality lost during the original lossy encoding is permanent. The AIFC output will be uncompressed and will not introduce any additional loss, but it will not restore frequencies or detail that were discarded when the FLV was first created.
FLV files store audio in compressed formats like AAC or MP3, which can reduce audio data size by 10x or more compared to uncompressed audio. AIFC with the PCM S16BE codec stores raw, uncompressed audio samples — no compression is applied. For example, a one-hour FLV with 128kbps AAC audio might have around 57MB of audio data, while the equivalent PCM S16BE AIFC at CD quality (44.1kHz stereo) would be approximately 635MB. This is expected behavior and is the cost of moving from lossy compressed audio to uncompressed PCM.
AIFC supports several codecs including PCM S16BE, PCM S24BE, PCM S32BE, floating-point PCM (F32BE and F64BE), and compressed formats like A-law and mu-law. This tool defaults to PCM S16BE because it is the standard baseline codec for AIFC and offers broad compatibility with Apple software and professional audio tools. If you need higher bit depth for audio editing or mastering, you can modify the FFmpeg command to use pcm_s24be or pcm_s32be instead.
To change the bit depth, replace pcm_s16be in the command with another supported PCM codec such as pcm_s24be (24-bit) or pcm_s32be (32-bit). To change the sample rate, add the -ar flag followed by your desired rate — for example, -ar 48000 for 48kHz or -ar 44100 for CD-quality output. A full example command would be: ffmpeg -i input.flv -vn -c:a pcm_s24be -ar 48000 output.aifc. The -b:a flag in the base command has minimal effect on lossless PCM codecs since bitrate is determined by sample rate, bit depth, and channel count.
Yes — on Linux or macOS, you can use a shell loop to process multiple files: for f in *.flv; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.flv}.aifc"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aifc". This is especially useful for archiving a folder of old Flash video files, and running FFmpeg locally this way is the recommended approach for large batches or files over 1GB.
AIFC is primarily an Apple format — it was developed by Apple and is natively supported in macOS and iOS audio tools including Logic Pro, GarageBand, Final Cut Pro, and QuickTime. On Windows, native support is limited, but many professional DAWs such as Adobe Audition, Reaper, and Steinberg Cubase can open AIFC files. If you need broader Windows or cross-platform compatibility, consider using the same FFmpeg command but outputting to WAV instead, which uses similar PCM encoding without the AIFF/AIFC container.
Technical Notes
The AIFC format uses big-endian byte ordering for its PCM data (hence 'BE' in codec names like pcm_s16be), which reflects its Apple and professional broadcast heritage — most modern consumer formats use little-endian ordering. The -b:a 128k flag in the base command technically targets a bitrate, but for lossless PCM codecs this value is largely informational; actual bitrate is determined by the sample rate, bit depth, and channel count of the source audio. If the FLV file contains stereo AAC at 44.1kHz, the AIFC output at pcm_s16be will have a true bitrate of approximately 1411kbps regardless of the -b:a setting. No video metadata or subtitle data is preserved because FLV does not support subtitles and the video stream is explicitly dropped. Audio timing and duration are preserved faithfully. One notable limitation: if the source FLV was encoded with a non-standard sample rate, FFmpeg will pass it through without resampling unless you explicitly add -ar to the command, so the output sample rate will match whatever was in the FLV source.