Audio Normalizer is a free, in-browser audio tool. Normalize the loudness of any audio file to broadcast standard (EBU R128) targets. Choose between -14 LUFS (Spotify/YouTube), -16 LUFS (podcasts), or -23 LUFS (broadcast TV). Files are processed entirely in your browser with FFmpeg WebAssembly. The page exposes a small surface — input, controls, output — so a first-time visitor can complete the job without reading documentation.
Audio Normalizer fits naturally into the workflow of language learners reviewing speech and students preparing oral submissions, both of whom typically need a fast result inside the browser. There is no learning curve to budget for: anyone who has used a typical web upload form can complete a run on the first try.
Audio Normalizer is shaped for the gap between "I'll do it by hand" and "I'll script it." When the job is small enough that automating it would take longer than doing it, but annoying enough to want a focused tool — that is the situation this page is built for.
Audio Normalizer runs on standard browser APIs — an open-source, well-audited engine that performs the audio editing and conversion natively in the browser. It accepts MP3, WAV, M4A, AAC, OGG, Opus, FLAC, AIFF, and WMA and produces output that opens in any standard audio viewer. Per-run input is capped at 200 MB.
The architecture is local-first by design. Once the page is loaded, you can disconnect from the network and the tool still completes the job. The processing stack — standard browser APIs and the small UI shell wrapping it — ships with the page itself, so the tool keeps working in offline conditions, on a captive-portal Wi-Fi, or behind a corporate proxy that limits what the tab can reach.
Workflow tip: Audio Normalizer pairs well with Audio Volume Booster and Audio Loudness Meter. Other adjacent tools you may find useful are Audio Equalizer and Audio Noise Reducer. Because every tool is a separate page, you can mix and match the steps that match your job. Bookmark the ones you reach for the most.
The only practical limit is the 200 MB per-file ceiling, which keeps the tool responsive across a wide range of devices. Run the tool ten times in a row, run it ten thousand times — it behaves the same way and produces the same quality of result.
Audio Normalizer keeps the control set focused. Every option on the page is there because a real workflow needs it, and the defaults aim at the most common case so a first-time user can get the right output without changing any settings.
Output handling is intentionally boring: Audio Normalizer produces `{name}-edited.{ext}` and triggers your browser's standard "save" behaviour. If you have a default download folder configured, that is where it will land. There is no Favtoo-side history of jobs you have run.
From a product perspective, Audio Normalizer is one of the simplest possible expressions of "do one thing well." The catalog contains dozens of related tools that each handle a slightly different audio editing and conversion task, and every one is a separate page rather than a tab inside a larger app. That separation keeps each tool fast to load and easy to bookmark.
Audio Normalizer is built around the moment of need: a focused page you open when you have a specific task, complete the task, and close. The catalog contains many adjacent tools so the same model serves the surrounding parts of a typical audio editing and conversion workflow.
A few practical tips that experienced users of Audio Normalizer pick up over time. First, keep your default browser updated — the engine relies on standard web APIs and newer browser versions are noticeably faster than ones from a few years ago. Second, close other heavy tabs before processing a large input; the engine shares CPU and memory with whatever else is open. Third, if you re-run the same kind of job often, your last-used settings are remembered for the rest of the tab session, so subsequent runs are essentially one click.
Common gotchas worth flagging: Audio Normalizer only accepts MP3, WAV, M4A, AAC, OGG, Opus, FLAC, AIFF, and WMA, so if your file is in another format you will need to convert it first. The 200 MB ceiling is per-file, not per-session; you can run as many separate jobs as you like, but a single oversized input will be rejected on load.
If Audio Normalizer solved your problem, sharing the page link with someone who has the same problem is the most useful thing you can do. The catalog grows mostly through word of mouth; visitors arriving through a recommendation tend to be the ones the tool serves best.
Unlike peak normalization (which simply scales the audio so the loudest sample is at 0 dB), loudness normalization measures the perceived loudness in LUFS (Loudness Units Full Scale) and adjusts gain so the integrated loudness matches a target. This is the modern broadcast / streaming standard, used by Spotify, YouTube, Apple Music, and major broadcasters.
-14 LUFS for Spotify, YouTube, Tidal, Amazon Music. -16 LUFS for Apple Podcasts and most podcast directories. -23 LUFS for European broadcast TV (EBU R128). -24 LUFS for US broadcast TV (ATSC A/85). The default -14 LUFS is the safe modern choice for online music distribution.
Peak normalization just makes the loudest sample 0 dB but preserves the dynamic range. A quiet ballad and a loud rock track both end up at "0 dB peak" but feel very differently loud. Loudness normalization adjusts perceived loudness, so two normalized tracks feel equally loud regardless of dynamic content.
No — FFmpeg's loudnorm filter applies a true-peak limiter so peaks never exceed -1 dBTP, preventing inter-sample clipping that would distort on conversion to lower bit-depth or analog playback. This is why loudness normalization is preferred over simple gain adjustment.
Yes — that's the point. If your file measures -25 LUFS and your target is -14 LUFS, the tool applies +11 dB of gain (with safe limiting). If the file is already louder than the target, gain is reduced. The result is consistent perceived loudness.
Up to 200MB. Normalization is a single-pass loudness analysis followed by a gain adjustment, so it processes at several times real-time on a modern laptop.
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.
Audio Normalizer runs in any modern mobile browser — Safari, Chrome, Firefox and the in-app browsers in most messaging apps all support the underlying APIs. Performance depends on the device: a recent phone handles typical inputs nearly as fast as a laptop, while older devices may take a few seconds longer near the 200 MB ceiling. The interface lays out cleanly on small screens, so you do not need to pinch-zoom to see the controls.
Favtoo keeps no copy of your file because Favtoo never receives your file. Audio Normalizer runs entirely in your browser, the input is held only in your tab's memory, and closing the tab discards it. There is no opt-in cloud history, no "recent jobs" panel synced to an account, and no server-side retention to configure — the architecture simply has nowhere for your file to be stored.
Audio Normalizer accepts MP3, WAV, M4A, AAC, OGG, Opus, FLAC, AIFF, and WMA. If your input is in a format that is not directly supported, convert it first using one of Favtoo's converter tools — every Favtoo converter outputs a file that is a clean input to the next tool in the chain.
Audio Normalizer is a browser-only tool by design and does not expose a hosted API. The reason is the same as the privacy story: there is no Favtoo backend doing the work, so there is no service to call. If you need to script the same transformation, the underlying engine (standard browser APIs) is open-source and can be used directly from your own code.
Audio Normalizer is a static page running an open-source engine in your browser, so a typical corporate firewall does not get in the way as long as it allows JavaScript to load from Favtoo. For teams that need to host it themselves on an internal network, the underlying engine (standard browser APIs) is open-source and can be packaged into a private build with the same behaviour. Reach out via the Contact page if that is something you are exploring.
Audio Normalizer can be used for personal and commercial work alike — there is no separate "business" licence to purchase. The output you generate is yours to use however you want, including in client deliverables, internal documents, or commercial products. Favtoo's only ask is fair, individual use; the tool is not designed to be embedded as a backend service or wrapped behind an API for resale.
Inputs are capped at 200 MB per file, which keeps memory usage stable across phones, tablets and older laptops. You can run Audio Normalizer as often as you need; every run produces a full-quality result.
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.