MyMediaTools

Extract Audio from FLV to AIF — Free Online Tool

Extract audio from FLV video files and save it as a lossless AIF file using PCM encoding. This tool discards the H.264 or FLV video stream entirely and decodes the compressed AAC or MP3 audio from the FLV container into uncompressed 16-bit big-endian PCM — the native format Apple's AIF standard uses — giving you archival-quality audio with no further quality loss.

FFmpeg Command

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Estimated output:

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How It Works

FLV files typically carry audio encoded in a lossy compressed format — most commonly AAC or MP3. During this conversion, FFmpeg reads the FLV container, ignores the video stream entirely, and decodes the compressed audio stream into raw PCM samples. Those samples are then written into an AIF container using the pcm_s16be codec — signed 16-bit big-endian PCM — which is the uncompressed audio encoding Apple's AIF format is built around. Because the source audio was already lossy (AAC or MP3), this process does not recover audio information that was lost during the original FLV encoding; it simply stops compressing it further. The resulting AIF file will be significantly larger than the FLV's audio track, and every audio sample will be preserved exactly as decoded, with no additional compression artifacts introduced.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion both in the browser via WebAssembly and on your local desktop installation.
-i input.flv Specifies the input file — in this case an FLV container, which typically holds an H.264 or native FLV video stream alongside a lossy AAC or MP3 audio stream.
-vn Disables video output entirely, instructing FFmpeg to ignore the video stream in the FLV file. Since the goal is to extract only the audio track into an AIF file — which is an audio-only format — no video data is needed or carried forward.
-c:a pcm_s16be Sets the audio codec to pcm_s16be (signed 16-bit big-endian PCM), which is the standard uncompressed audio encoding used in AIF files. FFmpeg decodes the lossy AAC or MP3 audio from the FLV and re-encodes it as raw PCM samples in big-endian byte order, matching the AIF specification.
output.aif Defines the output filename with the .aif extension, which tells FFmpeg to wrap the uncompressed PCM audio in an Audio Interchange File Format container — the lossless, uncompressed audio format developed by Apple and natively supported across all Apple audio and video applications.

Common Use Cases

  • Recovering dialogue or voiceover audio from old Flash-era tutorial or lecture recordings stored as FLV files, for re-editing in a DAW like Logic Pro or GarageBand that handles AIF natively
  • Extracting background music or sound effects from a downloaded FLV video clip to use in a Mac-based video or audio production project where AIF is the preferred interchange format
  • Archiving the audio track of legacy Flash video content — such as old webinars or conference recordings — in an uncompressed format before the FLV files become unplayable on modern systems
  • Pulling a recorded live performance or DJ set from an FLV stream capture and converting it to AIF for delivery to a mastering engineer who requires uncompressed source files
  • Preparing audio extracted from FLV screen recordings for transcription or accessibility workflows where lossless quality is required for accurate speech recognition
  • Stripping the audio from an FLV video file to produce an uncompressed AIF stem that can be imported into Pro Tools or Final Cut Pro X without any additional decoding overhead

Frequently Asked Questions

No — the AIF output will not improve on the audio quality present in the FLV source. FLV files use lossy audio codecs like AAC or MP3, which permanently discard some audio data during encoding. Converting to AIF decodes that compressed audio into uncompressed PCM, preventing any further quality loss, but the artifacts introduced during the original FLV compression are baked in. You are preserving what is there, not restoring what was removed.
AIF stores audio as uncompressed PCM samples, meaning every audio sample is written at full resolution with no compression applied. A typical stereo audio track at 44.1 kHz in 16-bit PCM consumes roughly 10 MB per minute, whereas the same audio compressed as AAC at 128k inside an FLV would be closer to 1 MB per minute. The dramatic size increase is expected and reflects the shift from lossy compression to lossless uncompressed storage.
The output uses pcm_s16be — signed 16-bit big-endian PCM — which is the standard codec for AIF files and is natively supported by all Apple software including GarageBand, Logic Pro, Final Cut Pro, and QuickTime. It is also widely supported in professional DAWs on other platforms. If your workflow specifically requires a higher bit depth such as 24-bit or 32-bit PCM, you would need to modify the FFmpeg command to use pcm_s24be or pcm_s32be, though those bit depths cannot recover detail that was lost in the original FLV's lossy compression.
FLV files can carry basic metadata such as duration, title, and author in their onMetaData event, but the AIF format has limited metadata support compared to modern containers. FFmpeg will attempt to map compatible tags, but many FLV metadata fields may not transfer cleanly or at all. For archival use, you should verify and add metadata manually using a tool like MP3Tag or your DAW's file properties after conversion.
The command uses '-c:a pcm_s16be' for standard 16-bit AIF audio. To produce a 24-bit AIF — which some professional workflows prefer — replace that flag with '-c:a pcm_s24be', giving you the command: ffmpeg -i input.flv -vn -c:a pcm_s24be output.aif. Similarly, you can use pcm_s32be for 32-bit integer or pcm_f32be for 32-bit float PCM. Keep in mind that since the FLV source audio was lossy, the additional bit depth will not recover lost detail but may be required by a specific delivery specification.
Yes. On macOS or Linux, you can use a shell loop to process an entire folder: for f in *.flv; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.flv}.aif"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aif". The browser-based tool on this page processes one file at a time, so the FFmpeg command is particularly useful when you need to handle a large batch or files exceeding 1 GB.

Technical Notes

FLV (Flash Video) was Adobe's dominant streaming container through the 2000s and early 2010s, and the vast majority of FLV files encountered today carry audio in AAC (Advanced Audio Coding) or MP3 format, both lossy codecs. The pcm_s16be codec used in the AIF output encodes audio as signed 16-bit integers in big-endian byte order, consistent with the AIF specification originally developed by Apple in the late 1980s for Macintosh systems. Because AIF uses big-endian byte order while most modern processors are little-endian, AIF files are slightly less efficient to read on non-Apple hardware but are natively optimal on Apple silicon and historical PowerPC systems. The absence of any compression in the output means frame-accurate editing is trivially supported in any AIF-compatible audio editor. One important limitation: if the FLV file contains multiple audio tracks (rare but possible in some capture tools), only the first audio stream will be extracted. Additionally, FLV does not support chapters or subtitle streams, so no data of that kind will be present in the source to begin with. The output AIF file will match the sample rate and channel count of the source audio stream — typically 44.1 kHz stereo — without any resampling unless you add explicit FFmpeg flags to change them.

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