Convert FLV to WAV — Free Online Tool
Extract and convert the audio track from an FLV (Flash Video) file into a WAV file using uncompressed PCM audio. This tool strips the lossy AAC or MP3 audio from the FLV container and decodes it into 16-bit PCM WAV — ideal for archiving, audio editing, or broadcast workflows that require uncompressed source material.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your FLV file here
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Settings
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 a lossy format — most commonly AAC or MP3 — packaged alongside a video stream encoded with H.264 or the legacy Sorenson Spark codec. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio into raw PCM samples, then writes those samples into a WAV container using the pcm_s16le codec (signed 16-bit little-endian PCM). This is a full decode-and-re-encode of the audio: the compressed audio data is decompressed to lossless PCM, which means any quality loss that occurred during the original FLV encoding is preserved, but no additional lossy compression is applied. The resulting WAV file will be significantly larger than the FLV source because PCM audio is uncompressed — a stereo 128kbps AAC stream at 44.1kHz will expand to roughly 10× its compressed size in WAV form.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In this browser-based tool, this runs via FFmpeg.wasm (WebAssembly) entirely within your browser — no files are sent to a server. The same command works identically in a local FFmpeg installation on Windows, macOS, or Linux. |
-i input.flv
|
Specifies the input file — your FLV (Flash Video) source. FFmpeg reads the FLV container, identifies its video stream (H.264 or legacy FLV codec) and audio stream (typically AAC or MP3), and makes both available for processing. |
-c:a pcm_s16le
|
Sets the audio codec for the output to pcm_s16le — signed 16-bit little-endian PCM. This decodes the lossy AAC or MP3 audio from the FLV into fully uncompressed audio samples, the standard format for WAV files and compatible with virtually all audio software. |
output.wav
|
Specifies the output file and its container format. The .wav extension tells FFmpeg to write a Waveform Audio File Format container. Since WAV cannot hold video, FFmpeg automatically omits the FLV's video stream — only the decoded PCM audio is written to this file. |
Common Use Cases
- Recovering audio from old Flash-era video files (e.g., downloaded YouTube videos from pre-2010, course recordings, or archived webcasts) for use in a modern audio editor like Audacity or Adobe Audition
- Extracting a voice-over or interview audio track from an FLV screen recording to import into a podcast production workflow that requires WAV source files
- Preparing audio from legacy Flash video content for broadcast or radio use, where WAV/PCM is the standard deliverable format
- Pulling a music performance or live event recording stored in FLV format into a DAW (Digital Audio Workstation) for mastering or mixing
- Converting FLV audio to WAV as an intermediate step before further processing — for example, transcoding to FLAC or applying noise reduction — since WAV preserves full fidelity after the initial lossy decode
- Archiving the audio component of Flash video content before the original FLV files become unplayable due to the deprecation of Adobe Flash Player
Frequently Asked Questions
No — the WAV file will not recover quality that was lost when the FLV was originally encoded. FLV files store audio in lossy formats like AAC or MP3, and that compression is irreversible. What this conversion does is decode the compressed audio to uncompressed PCM, meaning no further quality degradation is introduced. The WAV will be a bit-perfect representation of what the FLV's audio sounded like, just in an uncompressed container.
This is expected behavior. FLV files store audio in compressed formats (typically AAC at 128kbps or MP3), which achieve 8–10× compression over raw audio. WAV with pcm_s16le stores every audio sample uncompressed, so a 5-minute stereo FLV audio track at 128kbps AAC (about 5MB) will become roughly 50MB as a 16-bit 44.1kHz WAV. The larger file size is the cost of having fully uncompressed, edit-ready audio.
No. WAV is a pure audio container and cannot store video streams. FFmpeg automatically drops the video track when writing to a WAV output. Only the audio stream from the FLV is decoded and written to the WAV file. If you need to keep the video, you would need to convert to a video container format like MP4 or MKV instead.
pcm_s16le stands for signed 16-bit little-endian PCM — this is the most universally compatible form of uncompressed audio and the same encoding used by standard audio CDs. It supports a dynamic range of 96dB and is compatible with virtually every audio editor, DAW, and media player. If your source FLV has high-quality audio and you need more bit depth for professional mastering, you could modify the FFmpeg command to use pcm_s24le (24-bit) or pcm_s32le (32-bit) instead, though for most FLV content the original audio quality won't justify going beyond 16-bit.
Replace pcm_s16le in the command with your preferred WAV-compatible codec. For 24-bit audio use '-c:a pcm_s24le', for 32-bit floating point use '-c:a pcm_f32le', or for 8-bit audio (low-quality, small files) use '-c:a pcm_u8'. The full modified command would look like: ffmpeg -i input.flv -c:a pcm_s24le output.wav. Note that WAV does not support codecs like AAC or MP3 in broad practice, so PCM variants are the standard choice.
Yes, with a small shell script. On Linux or macOS, run: for f in *.flv; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.flv}.wav"; done — this loops over every FLV in the current directory and produces a matching WAV file. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This is especially useful for archiving large collections of legacy Flash video files.
Technical Notes
FLV files carry audio almost exclusively in AAC (MPEG-4 Audio) or MP3 format, both of which are lossy codecs. When FFmpeg reads the FLV and writes WAV, it fully decodes the compressed audio bitstream to raw PCM samples — this means the process involves actual audio decoding, not a simple remux or stream copy. The output uses pcm_s16le at whatever sample rate and channel count the source FLV audio was encoded with (commonly 44100Hz stereo, but older Flash content may use 22050Hz or even 11025Hz mono). FFmpeg preserves the original sample rate and channel layout by default, so no resampling occurs unless explicitly requested. One notable limitation: FLV has minimal metadata support and rarely carries embedded tags like artist, title, or album info, so WAV output will typically have no ID3-style metadata. WAV itself has limited metadata support anyway (basic BWF chunks aside), so this is rarely a concern. If the FLV contains multiple audio tracks (uncommon but possible), only the first audio stream is extracted by default. The resulting WAV file will have no chapter markers, no subtitles, and no video — it is a flat, linear audio file, which is precisely what makes it maximally compatible with downstream tools.