Convert FLV to AIFC — Free Online Tool
Convert FLV video files to AIFC audio format, extracting the AAC or MP3 audio stream and re-encoding it to PCM S16BE — a 16-bit big-endian uncompressed format used in professional audio workflows on Apple platforms. Ideal for archiving or further processing legacy Flash video audio content in a lossless PCM container.
to
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 audio encoded in AAC or MP3, both lossy compressed formats. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio from the FLV container, then re-encodes it to PCM signed 16-bit big-endian (pcm_s16be) — the default lossless PCM codec for the AIFC format. Because the audio is first decoded from a lossy codec and then written to an uncompressed PCM container, the output will not recover any quality lost in the original FLV encoding, but it will be a fully uncompressed, bit-for-bit stable representation of whatever audio fidelity was present in the source file. The resulting AIFC file is significantly larger than the FLV source and is ready for import into professional audio applications like Logic Pro or Pro Tools.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser-based version of this tool, this runs via FFmpeg.wasm compiled to WebAssembly, so no files leave your device. |
-i input.flv
|
Specifies the input file — an FLV container that typically holds video (H.264 or FLV1) and audio (AAC or MP3). FFmpeg reads and demuxes this file to access the raw compressed audio stream for decoding. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, which is the standard lossless uncompressed audio codec for AIFC files. The compressed AAC or MP3 audio from the FLV is fully decoded and then written as raw 16-bit samples in big-endian byte order, as required by the AIFC specification. |
-b:a 128k
|
Specifies an audio bitrate target of 128 kilobits per second. For the uncompressed pcm_s16be codec used here, this flag has no practical effect on output quality — PCM audio is written at a fixed bitrate determined by sample rate and bit depth — but it is included for compatibility with the tool's quality settings interface. |
output.aifc
|
Defines the output filename and tells FFmpeg to write the result as an AIFC file. FFmpeg uses the .aifc extension to select the AIFF-C muxer, which wraps the pcm_s16be audio data in the big-endian chunk structure that AIFC-compatible applications like Logic Pro and Pro Tools expect. |
Common Use Cases
- Archiving audio from old Flash-based e-learning recordings or webinar captures into a lossless PCM container for long-term preservation
- Importing audio from FLV screen recordings into Logic Pro or GarageBand, which natively support AIFC files, for music production or voiceover editing
- Extracting a spokesperson's voiceover from a legacy Flash promotional video into AIFC for re-use in a new production without lossy re-compression stacking
- Preparing audio extracted from FLV livestream recordings for mastering workflows that require uncompressed big-endian PCM source files
- Converting Flash video tutorial audio to AIFC so it can be imported into Final Cut Pro or other Apple-centric post-production pipelines without format compatibility issues
- Recovering audio from FLV files downloaded from archival video platforms before the original Flash content is taken offline permanently
Frequently Asked Questions
No — converting from FLV to AIFC will not recover quality lost during the original FLV encoding. The audio in FLV files is stored as AAC or MP3, both lossy formats that permanently discard audio data during compression. When FFmpeg decodes that audio and writes it to PCM S16BE inside an AIFC container, you get an uncompressed representation of the already-lossy source — the AIFC file is larger and uncompressed, but its fidelity ceiling is still set by the original FLV bitrate. Think of it as a lossless snapshot of a lossy recording.
FLV stores audio in compressed formats like AAC or MP3, which can reduce audio data by 10x or more compared to raw PCM. AIFC with pcm_s16be stores every audio sample as an uncompressed 16-bit value, so a 10-minute FLV with 128 kbps AAC audio might produce an AIFC file that is 50–80 MB or more. This size increase is expected and is the nature of uncompressed PCM audio containers — you are trading storage space for full compatibility with professional audio tools and zero generational loss in future exports.
No. FLV carries lossy compressed audio codecs like AAC and MP3, while AIFC was designed primarily for PCM and a small set of telephony codecs like A-law and mu-law. The tool uses pcm_s16be (16-bit big-endian PCM), which is the standard lossless codec for AIFC and ensures the broadest compatibility with Apple and professional audio software. AIFC cannot contain AAC or MP3 streams the way a container like M4A or MP3 file can.
FLV files store metadata in a format-specific structure that is not directly compatible with AIFC's chunk-based metadata system. FFmpeg will attempt to map common metadata fields, but FLV files often carry minimal or no standard metadata tags, and AIFC has limited metadata support compared to formats like MP4 or FLAC. In practice, most FLV-to-AIFC conversions will produce an AIFC file with little or no embedded metadata, so you may need to add tags manually in an audio editor after conversion.
Because the output is pcm_s16be — an uncompressed PCM codec — the -b:a bitrate flag has no meaningful effect on the audio fidelity; quality is determined by the bit depth and sample rate of the decoded source. If you want a higher bit depth, you can replace -c:a pcm_s16be with -c:a pcm_s24be or -c:a pcm_s32be for 24-bit or 32-bit output respectively, which AIFC also supports. To control the sample rate, add -ar 44100 or -ar 48000 before the output filename. For example: ffmpeg -i input.flv -c:a pcm_s24be -ar 48000 output.aifc
Yes. On Linux or macOS, you can use a shell loop: for f in *.flv; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.flv}.aifc"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc". Each file is processed sequentially and saved as a separate AIFC file. Note that the browser-based tool processes one file at a time; the FFmpeg command is the recommended approach for batch jobs, especially for large collections or files over 1 GB.
Technical Notes
AIFC (Audio Interchange File Format Compressed) is a big-endian format that originated in the Apple and SGI professional audio ecosystem. The pcm_s16be codec writes 16-bit signed integers in big-endian byte order, which means the most significant byte comes first — this is the native byte order for AIFC and is expected by Apple-platform tools. The source FLV audio is almost always 44.1 kHz or 48 kHz stereo; FFmpeg preserves the source sample rate and channel layout by default, so no resampling occurs unless explicitly requested. One known limitation is that FLV does not support subtitles or chapter markers, and neither does AIFC, so no metadata structure is lost in that regard. If the FLV source has multiple audio tracks (uncommon but possible in some non-standard FLV implementations), only the first audio track will be extracted by default. The -b:a flag is included in the command for consistency with the tool framework but does not affect the uncompressed PCM output quality in any practical way — pcm_s16be output quality is entirely determined by the bit depth (16-bit) and the fidelity of the decoded source audio.