Extract Audio from FLV to ALAC — Free Online Tool
Extract audio from FLV files and save it as ALAC (Apple Lossless Audio Codec) in an M4A container — perfect for archiving Flash video soundtracks with zero quality loss. Because FLV's AAC audio is transcoded into ALAC's lossless encoding, the resulting file preserves every bit of audio fidelity the original FLV contained.
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FFmpeg Command
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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 AAC or MP3 audio streams alongside H.264 or legacy FLV video. This tool discards the video stream entirely and re-encodes the audio — whether AAC or MP3 — into ALAC, Apple's lossless codec stored in an MPEG-4 (.m4a) container. Because ALAC is lossless, the output captures the full decoded audio signal from the FLV source without introducing any additional compression artifacts. However, since FLV audio is itself lossy (AAC or MP3), ALAC will preserve that decoded signal perfectly but cannot recover detail that was already lost during the original FLV encoding. The output file will be noticeably larger than the FLV's audio track, but it becomes a stable, artifact-free archival copy.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles all demuxing, decoding, encoding, and muxing for this conversion. |
-i input.flv
|
Specifies the input FLV file. FFmpeg reads the Flash Video container and identifies the available streams — typically an H.264 or FLV video stream and an AAC or MP3 audio stream. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the FLV's video stream. Since the goal is audio extraction, this ensures no video data is processed or written to the output M4A file. |
-c:a alac
|
Sets the audio codec to ALAC (Apple Lossless Audio Codec). FFmpeg decodes the FLV's lossy AAC or MP3 audio to PCM and then re-encodes it using ALAC's lossless compression algorithm for the output. |
-c:a alac
|
This flag appears twice in the resolved command, which is redundant but harmless — FFmpeg simply applies the last valid value, confirming ALAC as the audio encoder. In practice, a single -c:a alac is sufficient. |
output.m4a
|
Defines the output filename with an .m4a extension, which tells FFmpeg to wrap the ALAC audio stream in an MPEG-4 container — the standard and Apple-recommended container for ALAC files. |
Common Use Cases
- Archiving the audio from old Flash-era video content — tutorials, lectures, or music performances — before the FLV files degrade or become unplayable
- Importing a FLV video's soundtrack into GarageBand, Logic Pro, or Final Cut Pro, which natively support ALAC/M4A but not FLV
- Preserving a recorded live stream or webinar audio in lossless format for long-term storage without the risk of further generational quality loss from re-encoding
- Extracting music or audio commentary from an FLV game recording to create an iTunes-compatible track with proper metadata tagging
- Preparing audio from Flash video interviews or podcasts saved as FLV for distribution in an Apple ecosystem-friendly lossless format
- Creating a lossless reference copy of FLV audio to use as a master when producing multiple derivative formats (MP3, AAC) without repeated lossy transcoding
Frequently Asked Questions
No — ALAC is lossless, meaning it perfectly preserves whatever audio data it receives, but it cannot reconstruct detail that was already discarded by the FLV's original lossy encoder (typically AAC or MP3). Think of it as making a perfect photocopy of an already-compressed image: the copy is exact, but the original compression artifacts remain. The benefit of ALAC here is preventing any further quality degradation if you ever need to re-encode the audio again later.
FLV stores audio using lossy codecs like AAC (typically at 128k bitrate) that discard audio data to reduce file size. ALAC stores the fully decoded audio signal using lossless compression, which is far less aggressive. A 128kbps AAC stream might expand to 5–10x its compressed size when stored as ALAC, because lossless formats don't throw away any of the PCM audio data. This is expected and is the trade-off for having a stable, re-encodable master copy.
Yes — ALAC in an M4A container is natively supported across the entire Apple ecosystem, including iTunes, Apple Music, iPhone, iPad, macOS, and Apple TV. You can import the output M4A file directly into your iTunes or Music library and it will be recognized as a lossless audio file. This makes the FLV-to-ALAC conversion particularly useful for archiving audio you intend to sync to Apple devices.
FLV files have limited metadata support, and most FLV files in the wild carry little to no embedded tag information beyond basic streaming parameters. The M4A container used by ALAC supports rich metadata tags (title, artist, album, artwork), but FFmpeg will only carry over whatever metadata actually exists in the source FLV. You will likely need to add or edit metadata manually after conversion using a tool like MusicBrainz Picard or iTunes.
ALAC does not use bitrate-based quality settings — it is always lossless, so there is no -b:a or -q:a flag that applies here. However, you can resample the audio to a different sample rate by adding -ar 44100 (for CD quality) or -ar 48000 before the output filename, for example: ffmpeg -i input.flv -vn -c:a alac -ar 44100 output.m4a. Keep in mind the source FLV audio is already lossy, so downsampling may reduce file size slightly but upsampling will not add any real audio information.
On Linux or macOS, you can use a shell loop: for f in *.flv; do ffmpeg -i "$f" -vn -c:a alac "${f%.flv}.m4a"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a alac "%~nf.m4a". This processes every FLV in the current directory and outputs a matching M4A file for each, making it practical for large archives of Flash video content.
Technical Notes
FLV (Flash Video) containers most commonly carry AAC audio at bitrates between 64k and 192k, though older FLV files from the early Flash era may use MP3 audio instead. In either case, FFmpeg decodes the lossy audio stream to raw PCM before re-encoding it as ALAC — meaning this is a full transcode, not a stream copy. The ALAC encoder in FFmpeg is mature and well-tested; it produces files fully compliant with Apple's ALAC specification and compatible with all ALAC-capable decoders. One notable limitation: FLV does not support chapters or multiple audio tracks, so there is nothing of that kind to carry over, even though the M4A container supports chapters. The output M4A file will have a single stereo (or mono, matching the source) ALAC stream. If the FLV source was encoded with a non-standard sample rate, FFmpeg will preserve it in the ALAC output, though some Apple devices may resample unusual rates on playback. For files originally encoded at 44.1kHz or 48kHz — by far the most common cases — compatibility is universal.