Audio Converter changes a sound file from one format to another. The reasons people need this are familiar: an iPhone produces M4A files that some Windows tools refuse to import; a podcast app exports WAV when your CMS only accepts MP3; an old recording is in some obscure format and you need a modern equivalent; a video has audio you want to extract as a standalone file. The tool handles all of these with one interface and the same set of supported formats: MP3, WAV, AAC, OGG/Opus, FLAC, M4A, and audio extracted from video files.
Format choice matters more than people sometimes realise. MP3 is universal — every device plays it, every editor accepts it, every CMS understands it. AAC produces noticeably better quality than MP3 at the same bitrate, especially below 128 kbps; it is the default for iTunes/Apple Music and standard inside MP4 video. WAV is uncompressed and lossless but takes 10× the disk space of compressed equivalents. FLAC is lossless but compressed (about half the size of WAV) — perfect for archival storage where every sample matters. OGG/Opus is the open-source choice and produces excellent quality at very low bitrates.
The bitrate slider controls the trade-off between file size and audio quality for lossy formats (MP3, AAC, OGG). The defaults pick sensible values: 192 kbps for MP3 (transparent for most music), 128 kbps for AAC (matches the original Apple iTunes default), 96 kbps for Opus (genuinely transparent for speech). For lossless formats (WAV, FLAC) bitrate is determined entirely by the source’s bit depth and sample rate — you control those independently if you want to downsample.
Audio extraction from video is built in: drop an MP4, MOV, MKV, or WebM file and the tool pulls the audio track out and re-encodes it to your chosen format. Multi-track audio (multiple language dubs, commentary tracks) gets a track-picker so you choose which one to extract. The whole conversion runs in your browser tab using FFmpeg-WASM and the Web Audio API; nothing is uploaded.
Input: MP3, WAV, OGG (Vorbis and Opus), FLAC, M4A (AAC), AAC, Opus, AIFF, WMA, and most other common formats FFmpeg can decode. Output: MP3, WAV, OGG (Vorbis), FLAC, M4A (AAC), Opus. The picker shows compatible options based on what you upload.
It depends on the conversion. Going lossless → lossless (e.g., WAV to FLAC) is byte-perfect. Lossy → lossless (MP3 to WAV) wraps the already-compressed audio in a bigger container — no quality is regained but no extra is lost. Lossy → lossy (MP3 to AAC) means re-encoding which adds a small amount of generation loss.
MP3 at 192 kbps is the universal choice — plays everywhere, reasonable file size, good quality. M4A/AAC sounds slightly better at the same bitrate and is preferred by Apple devices. Opus is the best modern codec but has limited hardware support. WAV/FLAC are for lossless archival.
For MP3: 128 kbps is OK, 192 kbps is good, 256 kbps is great, 320 kbps is the highest quality. For AAC and Opus: subtract about 30% — Opus 96 kbps sounds like MP3 128 kbps. For voice-only (podcasts), 64–96 kbps mono is enough.
Up to 200MB. Conversion happens in your browser using FFmpeg WebAssembly. Lossy → lossless conversions (which expand the file) may approach the limit faster than other directions.
Server-side tools run native FFmpeg with multi-threading and hardware acceleration. We run FFmpeg compiled to WebAssembly in a single browser thread, typically 2–5× slower. The trade-off is total privacy — your audio never leaves your device.
Server-side tools use multi-threaded native FFmpeg running on dedicated CPUs with fast disks and parallel pipelines. Our engine is FFmpeg compiled to WebAssembly, which runs single-threaded inside your browser tab and has no access to native hardware acceleration. That makes browser-based jobs typically 3–8× slower than a server. The trade-off is total privacy: your audio file is never uploaded, never logged, and never stored — closing the tab erases everything from memory immediately. For most clips up to a few minutes the wait is small, and for sensitive recordings (voice memos, drafts, confidential meetings) the privacy gain is well worth it.
No. Everything runs entirely inside your browser tab using FFmpeg compiled to WebAssembly. The file is read into local memory only, processed in the same tab, and the result is offered as a direct download. Nothing is transmitted to any server, no account is required, no analytics are tied to your file, and closing the tab discards every byte from memory.
The file picker accepts audio inputs up to about 1 GB, which is well above what mainstream "free tier" online converters allow. The real ceiling is your device — everything runs inside your browser tab, which shares memory with the rest of the page. Most podcasts, songs, and voice memos sit comfortably under that limit even on a phone. If a very large lossless WAV or FLAC ever fails, trim it first or transcode to MP3 / Opus to bring the size down before re-running the tool.
MP3, WAV, OGG (Vorbis and Opus), FLAC, M4A (AAC), AAC, Opus, AIFF, and WMA all decode reliably via FFmpeg WASM. Output formats depend on the specific tool — most editing tools default to MP3 (universal) or WAV (lossless) but expose a format picker so you can pick the one that fits your downstream player or DAW.
Recent Chrome, Edge, Firefox, Safari, and other Chromium-based browsers all work. The tool relies on WebAssembly and SharedArrayBuffer, which require the page to be served over HTTPS with the right cross-origin headers — this site is configured correctly by default. On phones the same code runs but is slower than on a desktop because mobile CPUs are weaker.
No. The tool is completely free, requires no account, attaches no watermark, applies no usage caps, and shows no popup ads on your output. Because the work happens on your own device, there is no per-user quota for us to enforce — your hardware and browser memory are the only limits. The download is the file you would get from running FFmpeg locally, nothing more, nothing less.
Lossless to lossless (WAV → FLAC, FLAC → WAV) preserves every sample exactly — no quality change. Lossless to lossy (WAV → MP3, FLAC → AAC) adds one generation of compression loss; at sensible bitrates (192+ kbps for MP3, 128+ for AAC) this is essentially inaudible. Lossy to lossy (MP3 → AAC, AAC → OGG) accumulates loss because the new codec re-compresses what is already lossy data — for one conversion the loss is small; repeated conversions degrade the file noticeably.
For most music, 192 kbps is genuinely transparent — listeners cannot reliably distinguish it from the lossless source in blind tests. 128 kbps is the lowest bitrate that still sounds acceptable for music. For speech (podcasts, audiobooks, voice memos), 96 kbps is plenty and produces files about half the size. The default 192 kbps balances quality and size for all-purpose use.
Yes — drop an MP4, MOV, MKV, or WebM file and the tool extracts the audio track and converts it to your chosen format. Multi-track videos (with multiple audio languages or a commentary track) show a track picker so you can choose which one to extract.
MP3 is lossy compression — it discards detail that human ears are bad at noticing. At 192+ kbps the loss is essentially inaudible to most listeners on most equipment. At lower bitrates (96 kbps and below) you may hear specific artefacts: a slight "swooshing" on cymbals, a flatness in piano sustains, or a metallic edge on female vocals. Bumping the bitrate to 256 kbps usually eliminates any audible difference; 320 kbps is overkill for almost all uses.
AAC is a newer codec designed to fix MP3’s known weaknesses. At the same bitrate, AAC sounds noticeably better than MP3, especially below 128 kbps and for high-frequency content like cymbals. The trade-off is compatibility — every device plays MP3, but some very old devices do not play AAC. For modern uses (iPhones, modern Android, web playback, modern streaming) AAC is strictly better.
Yes, if you want lossless compression. FLAC stores audio with no quality loss whatsoever and is typically 40–60% smaller than the equivalent WAV. Modern tools and music players almost universally support FLAC; the one notable exception is iTunes, which prefers ALAC (Apple’s lossless format). For pure archival, FLAC is the right answer.
Yes — the advanced settings let you set output sample rate (44.1, 48, 96, 192 kHz) and bit depth (16, 24, 32 bits) independently of the input. Downsampling (e.g. 96kHz → 44.1kHz) is sometimes useful when a destination requires CD-quality input. Upsampling does not improve audio quality but can be required by certain tools.
Yes by default — ID3 tags (for MP3), iTunes-style tags (for AAC/M4A), Vorbis comments (for OGG/FLAC) all carry through. The tool reads metadata from the source format and writes it to the equivalent fields in the output format. If a tag does not exist in the destination format, it is dropped.
Audio Recorder
Record from your microphone directly in the browser. Pick quality (high, medium, low), toggle echo cancellation, noise suppression and auto-gain, then save to WebM/Opus or M4A/AAC. Audio is captured locally — nothing is uploaded.
Text to Speech
Type or paste text, pick a system voice, and listen instantly. Adjust speaking rate (0.5×–2×), pitch, and volume in real time. Uses your browser's built-in Web Speech API — no cloud TTS, no API keys, no costs.
Tone Generator
Generate a pure tone at any frequency from 20 Hz to 20 kHz. Pick a sine, square, triangle, or sawtooth waveform, choose duration, amplitude, and mono/stereo. Exports a 16-bit PCM WAV file at 44.1 kHz with built-in click-preventing fades.
Silence Generator
Generate a perfectly silent WAV file of any length from 1 second up to 1 hour. Pick mono or stereo, get a 16-bit PCM WAV at 44.1 kHz. Useful as padding between clips, intro silence, leader audio for video timing, or test material.
White Noise Generator
Generate white, pink, or brown noise as a 16-bit PCM WAV file. Pick noise type, duration up to 1 hour, amplitude, and mono/stereo. Useful for sleep, focus, masking distractions, audio testing, and as a backing layer for ambient music.
Metronome
A precise browser-based metronome powered by the Web Audio API. Set BPM from 30 to 300, choose a time signature, accent the first beat, and use tap-tempo to sync. Click timing is sample-accurate using lookahead scheduling — much steadier than typical JavaScript setInterval beats.
Audio Trimmer
Trim any audio file to a precise start and end time. Outputs a lossless stream-copy by default (no quality loss, very fast) or re-encodes to MP3, WAV, OGG, or M4A. Files are processed entirely in your browser with FFmpeg WebAssembly.
Audio Splitter
Split a long audio file into N equal-length parts and download them as a ZIP. Each part is named sequentially. Great for chapterizing audiobooks, podcasts, or long DJ mixes. Runs entirely in your browser with FFmpeg WebAssembly.