Extract Audio from FLV to WAV — Free Online Tool
Extract the audio track from an FLV file and save it as an uncompressed WAV file. This tool decodes the AAC or MP3 audio embedded in the FLV container and re-encodes it to 16-bit PCM (pcm_s16le) — the standard lossless format used in broadcast, audio editing, and archival workflows.
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 of which are lossy compressed formats. Because WAV with pcm_s16le is uncompressed PCM, the audio cannot simply be remuxed — it must be fully decoded from its compressed form and then written out as raw 16-bit signed little-endian PCM samples. This means you do incur one additional decode step from the original lossy codec, but no further lossy compression is applied; the result is a lossless representation of exactly what was audible in the FLV. The video stream is discarded entirely using the -vn flag. The output WAV file will be significantly larger than the source audio in the FLV, because uncompressed audio at CD quality (44.1 kHz stereo, 16-bit) requires roughly 10 MB per minute compared to under 1 MB per minute for 128k AAC.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no data is sent to a server. On your desktop, this calls the locally installed FFmpeg binary. |
-i input.flv
|
Specifies the input FLV file. FFmpeg will detect the container format and identify the contained streams — typically a video stream (H.264 or Sorenson Spark) and an audio stream (AAC or MP3) — before processing begins. |
-vn
|
Disables all video output, discarding the video stream from the FLV entirely. Since WAV is a pure audio container, this flag ensures FFmpeg does not attempt to include or transcode the video track, which would result in an error. |
-c:a pcm_s16le
|
Decodes the FLV's compressed audio (AAC or MP3) and re-encodes it as 16-bit signed little-endian PCM — the standard uncompressed audio format used in WAV files for broadcast, archival, and DAW compatibility. |
output.wav
|
Sets the output filename and tells FFmpeg to write a WAV container. The .wav extension triggers FFmpeg's WAV muxer, which wraps the pcm_s16le audio data in the standard RIFF/WAV file structure recognized by all major operating systems and audio tools. |
Common Use Cases
- Extracting narration or commentary audio from a Flash-era tutorial or webinar recording so it can be imported into a DAW like Audacity or Adobe Audition for editing
- Archiving the audio from legacy FLV broadcast clips in an uncompressed format before the FLV files are deleted or migrated
- Pulling background music or sound effects out of an FLV promotional video to use in a video editor that requires uncompressed audio on the timeline
- Preparing audio extracted from an FLV source for transcription services that require WAV input (many speech-to-text APIs such as Google Cloud Speech prefer or require PCM WAV)
- Recovering the audio from a corrupted or partially downloaded FLV file where the video stream is unplayable but the audio packets are intact
- Creating a broadcast-ready WAV master from an FLV recording of a live event or stream, free of additional lossy encoding stages
Frequently Asked Questions
The FLV's audio is already lossy-compressed (typically AAC or MP3), so some quality was lost when the FLV was originally encoded. Converting to WAV does not compress the audio any further — it decodes the AAC or MP3 and writes every sample out as uncompressed 16-bit PCM. You will not get better quality than was in the original FLV, but you will not lose any additional quality. The WAV output is a lossless snapshot of what the FLV sounded like.
FLV stores audio in a compressed codec like AAC (commonly at 128 kbps), which discards perceptually redundant data to shrink file size. WAV with pcm_s16le stores every audio sample as a raw 16-bit integer with no compression. At CD-quality stereo (44.1 kHz), that works out to roughly 10 MB per minute — compared to under 1 MB per minute for 128k AAC. A 50 MB FLV might produce a 70–150 MB WAV depending on the video-to-audio ratio and the original audio sample rate.
The output uses pcm_s16le, meaning 16-bit signed PCM stored in little-endian byte order. This is the same format used by Audio CDs and is universally compatible with DAWs, video editors, transcription tools, and broadcast playout systems. If you need higher bit depth for professional audio mastering, you could modify the FFmpeg command to use pcm_s24le or pcm_s32le instead of pcm_s16le. For most extraction and editing purposes, pcm_s16le is the right choice.
Yes. To resample the audio, add -ar 48000 before the output filename (replacing 48000 with your target rate such as 22050 or 44100). To convert to mono, add -ac 1. For example: ffmpeg -i input.flv -vn -c:a pcm_s16le -ar 44100 -ac 1 output.wav. These flags are useful when delivering to transcription APIs that require mono 16 kHz audio or to broadcast systems with a specific sample rate requirement.
The command as shown processes one file at a time, but you can batch it easily in a shell. On Linux or macOS, run: for f in *.flv; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.flv}.wav"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". This is especially practical for large collections of archived FLV recordings where the 1 GB browser limit might apply to individual files.
FLV files can carry limited metadata in their onMetaData event block, but this metadata is not standardized and is often absent or minimal. WAV supports basic metadata via INFO chunks, and FFmpeg will attempt to copy any recognized tags from the FLV into the WAV output. In practice, FLV files rarely contain rich metadata, so do not rely on tag preservation — verify the output in your audio editor and add tags manually if needed.
Technical Notes
FLV is a legacy streaming container whose audio track is almost always AAC (in modern encoders) or MP3 (in older Flash-era files). Both codecs are lossy, so the extraction process involves a full decode stage before writing PCM samples to the WAV container. The default codec in this tool is pcm_s16le (16-bit signed little-endian PCM), which matches the WAV specification's most common and broadly compatible profile. WAV files do not natively support chapters, embedded subtitles, or multiple audio tracks, and neither does FLV, so no feature loss occurs on those dimensions. One known limitation: very old FLV files encoded with Nellymoser or Speex audio codecs (used by some early Flash applications) may decode with degraded fidelity or require FFmpeg builds that include those decoders. Sample rate is preserved from the source — if the FLV audio is 22050 Hz mono, the WAV will also be 22050 Hz mono unless explicitly resampled via -ar. The resulting WAV has no upper limit on playback compatibility and is accepted by virtually every audio and video tool in existence.