Audio Effects Online is an audio effects online alternative that decodes one browser-supported audio file, applies a single deterministic transformation in the current browser tab, and exports the result as a PCM16 WAV that stays on your device throughout the process. Every step — file reading, decoding through the browser's Web Audio implementation, sample processing, WAV encoding, and download creation — happens in the same tab, and audio is not uploaded to Lizely or stored in an account. There is no server render and no hidden AI pass to wait for. The tool is deliberately narrow: you pick one file, choose one of six bounded effects — echo, lightweight reverb, reverse, fade in, fade out, or peak normalization — set a whole-number amount from 0 to 100 percent, and download the complete result. The original stays untouched on your device, so you can retry with different settings without re-exporting from a DAW or re-recording the source. Each run uses a fresh decode context, and temporary object URLs are revoked when the file, effect, or amount changes, which prevents stale results from replacing newer ones.
Why people look for an audio effects online alternative
Most public audio tools fall into one of three buckets, and each one tends to push users toward something more local. The first bucket requires you to upload the file to a remote server, where it is processed and then sent back — that upload is a privacy concern for voice memos, unreleased recordings, client work, and any audio that has not yet been cleared for distribution. The second bucket hides the actual effect behind an account, a sign-up wall, or a subscription, even for a single echo. The third bucket installs a heavyweight desktop application or a browser extension that updates itself in the background. None of those models is wrong for every user, but each one has a cost the user is forced to pay: bandwidth and trust, friction and money, or disk space and update noise. An audio effects online alternative that runs the entire pipeline inside the browser tab sidesteps all three at once.
Local processing also changes how predictable the result is. A cloud tool may re-encode your audio at a chosen bitrate before sending it back, so the WAV you download is not a lossless transformation of the file you uploaded. A browser tool that decodes through the W3C Web Audio API and writes its own PCM16 stream can apply exactly one disclosed operation to every sample, and you can read the numbers back to confirm what happened. That kind of transparency is hard to find in a black-box web service.
What Audio Effects Online does differently
Audio Effects Online is built around a small, fixed contract: one file in, one effect applied, one WAV out. It accepts filenames associated with MP3, WAV, M4A, AAC, Ogg, WebM, and FLAC, but the actual codec that the browser can decode is what counts — a recognized extension does not guarantee that a particular encoded file will decode, and if decoding fails, no result is created. Decoded audio must fit 1–8 equal-length channels, a sample rate between 8,000 Hz and 192,000 Hz, and no more than 30 million channel samples in either the input or the output. The encoded file itself must stay under 50 MiB. Echo and reverb add frames to the output, so they can pass the input budget yet fail the output budget; in that case nothing is truncated and no file is written. Sample processing is constrained to the legal −1 to +1 floating range before quantization, which keeps the resulting PCM16 stream from wrapping on overflow.
Because the tool applies one effect per run, it is easy to reason about what changed in any given file. You always know which single operation touched the samples, and you can compare the original against the output without wondering whether two effects fought each other behind the scenes.
The six effects and how each one behaves
The six effects are deliberately simple, disclosed, and reproducible. The table below lists what each one actually does to the sample stream. None of them models a concert hall, plate, spring, impulse response, or branded processor — they are explicit Lizely product choices, and the coefficients are not measurements of acoustic spaces.
| Effect | What it does to the samples | How the amount slider (0–100%) changes it |
|---|---|---|
| Echo | Retains every input sample and adds a half-strength copy 200 ms later; the complete delayed tail is allocated up front so the last echo is not cut off. | Scales the delayed copy linearly — at 0% you hear only the dry signal, at 100% the delayed copy is half the input amplitude. |
| Lightweight reverb | Adds three discrete early reflections at 29.7 ms, 37.1 ms, and 41.1 ms. | Multiplies the reflection weights 0.50, 0.35, and 0.25 by the selected amount — a quick space or draft sound-design pass, not a room simulator. |
| Reverse | Reads every decoded sample frame in the opposite order and blends with the original. | Amount controls how much reversed signal mixes against the dry signal. |
| Fade in | Linear gain ramp from the dry proportion at the start to full level at the end. | Dry start percentage scales with amount, giving shorter or longer audible build-ups for the same 0–100% range. |
| Fade out | Inverse of fade in — full level at the start, scaled down to the dry end level. | Same whole-number range controls how much of the tail fades versus how much stays at the dry proportion. |
| Normalization | Finds the largest absolute decoded sample and scales the whole file toward a 0.95 sample peak. | Not loudness matching: it does not target LUFS, perceived volume, broadcast compliance, dynamics, true peak between samples, or equal loudness across recordings. Silent input stays silent. |
The reverb design in particular is worth understanding before you use it. Three reflections spaced between 29.7 and 41.1 ms add a small amount of perceived space, but they do not model room dimensions, frequency-dependent decay, stereo diffusion, damping, pre-delay, convolution, or a professional reverberation tail. For a deeper look at how those specific delay times were chosen, see Reverb Delay Times Explained: 29.7, 37.1, and 41.1 ms.
Apply one effect and download the PCM16 WAV
- Choose one browser-decodable audio file no larger than 50 MiB. Filenames that end in MP3, WAV, M4A, AAC, OGG, WEBM, or FLAC are accepted, but the actual decode path follows what the browser can read.
- Select echo, lightweight reverb, reverse, fade in, fade out, or normalization.
- Set a whole-number amount from 0% through 100%. The amount blends the wet signal, scales the delay weights, or controls the fade endpoints depending on the effect.
- Apply the effect locally. Decoding, sample processing, and PCM16 encoding happen in your browser tab; the page does not upload the audio.
- Inspect the reported channel count, sample rate, input frame count, output frame count, output peak, and exact WAV byte size to confirm the run did what you expected.
- Download the PCM16 WAV while keeping the untouched original so you can re-run the tool with different settings.
Reading the reported numbers after processing
The page prints a small block of statistics after each run, and those numbers are the most reliable way to confirm the tool actually applied the chosen operation. Channel count and sample rate come straight from the decoded AudioBuffer defined by the W3C Web Audio specification. The input and output frame counts let you see, for example, that an echo at 100% amount produced an output that is longer than the input by the exact number of samples a 200 ms delay adds at the file's sample rate — at 44,100 Hz that is 8,820 extra frames regardless of channel count — 17,640 is the extra channel samples for stereo, not extra frames. The output peak shows how close the loudest sample came to the legal ±1 range after clipping was applied. The exact WAV byte size lets you cross-check against the documented PCM block alignment: at 16 bits, mono frames take 2 bytes and stereo frames take 4 bytes, in line with Microsoft's WAVEFORMATEX structure.
For a 5-second mono clip at 44,100 Hz, the dry frame count is 220,500 (5 × 44,100). With echo at 100%, the output frame count becomes 220,500 + 8,820 = 229,320, because 0.2 × 44,100 = 8,820 extra mono frames are appended. The WAV byte size for that mono output is 229,320 × 2 = 458,640 bytes. The same 5 seconds in stereo becomes 220,500 + 8,820 = 229,320 frames (458,640 channel samples), or 229,320 × 4 = 917,280 bytes. These figures are illustrative — confirm the exact bytes on the page after running your own file through Audio Effects Online.
Stacking more than one effect
Because the tool applies one effect per run, a chain is built manually. Download the result, inspect the reported numbers, then feed that WAV back in as the next input and pick a different effect. This pattern makes every transformation auditable, but it does have a cost. Repeated lossy decode and PCM processing can reveal source artifacts from the original encoder, and repeated clipping or normalization can change the dynamics of a file in ways that compound run after run. Treat the original as the source of truth, keep it untouched, and work from copies.
Honest limits before you load a file
Three limits matter in practice. First, the encoded input cannot exceed 50 MiB, and the decoded input and output each cannot exceed 30 million channel samples — so a very long stereo file at a high sample rate can hit the channel-sample budget even when the file size is modest. Second, echo and reverb add frames, which means a file that decodes within budget can still produce an output that exceeds the output budget; in that case nothing is truncated and no file is created. Third, changing the file, effect, or amount invalidates older work — decode contexts are closed, stale jobs cannot replace newer state, and temporary object URLs are revoked on replacement and unmount to prevent leaks. The reliable contract stays narrow: decode one supported file, apply the exact disclosed operation to every frame, preserve the complete calculated tail within budget, encode a standards-based PCM16 WAV, and state clearly what the lightweight effect does not claim to emulate.
If you're weighing options, Plan the Steps Needed to Join Audio Files Locally covers this in detail.