ALL AUDIO TOOLS18
Change Audio Volume
Change Audio Volume Online — Free Gain Tool
Set a gain in decibels; the peak is measured on the decoded file and clipping is reported rather than hidden.
Nothing is enforced, but the file is decoded to 32-bit float in this tab — once to draw the waveform and once to scale it, at roughly 20 MB per minute of 44.1 kHz stereo — and the scaled copy is held alongside the original.
About Change Audio Volume
Sometimes the whole file is simply too quiet, or too loud, and what you want is not a loudness standard but a number of decibels. This tool applies exactly that: a gain, in dB, multiplied into every sample of the decoded file. +6 dB is roughly twice the amplitude, −6 dB roughly half, and the slider runs from −30 to +30 in half-decibel steps. There is no limiter and no compressor anywhere in the path, which is a deliberate choice and the reason the page reports what it does. A gain that pushes the loudest peak above full scale cannot be stored by any audio format; a tool that quietly flattened those peaks would be handing you a different sound from the one the number describes. So the peak is measured on the decoded samples before and after the multiply, both numbers are printed, and a run that goes over says by how much and how far to come back. The waveform on screen cannot answer that question on its own — it is normalised so the loudest bar fills the space, which is a picture of shape rather than level — so the clipping answer only appears once a run has really measured the file. On the MP3 and WAV arms nothing leaves the browser's own engines: the file is decoded with the Web Audio API, scaled in place, and encoded by a worker. FLAC output has no encoder in that worker, so that arm renders a lossless WAV and hands it to FFmpeg, which is also what decodes a container your browser refuses — WMA everywhere, and AAC or Ogg in some browsers. Length always comes out unchanged, and on the WAV and FLAC arms so do the channel count and sample rate. MP3 cannot: it carries at most two channels and 48 kHz, so a surround source is summed into stereo and anything above 48 kHz is resampled down. The run says which of those happened rather than leaving you to find out. Tags and cover art do not: decoding reduces the file to raw samples, where metadata stops existing.
Questions
How many decibels should I use?
+6 dB is roughly twice the amplitude and −6 dB roughly half, and the slider runs from −30 to +30 in half-decibel steps. If you want a specific loudness rather than a specific change, use the audio normalizer instead — it measures the file and works out the gain for you.
What happens if the gain makes it clip?
You are told. Nothing here limits or compresses, so a peak pushed above full scale cannot be stored — the tool measures the peak on the decoded samples before and after the multiply, prints both, and a run that goes over says by how much and how far to come back. It never quietly flattens the peaks and calls that success.
Why can't the waveform tell me whether it will clip?
Because the waveform is normalised: the loudest bar is scaled to fill the space, so it draws the shape of the file and not its absolute level. Two files with identical pictures can sit 20 dB apart. The clipping answer only appears after a run, because that is the first point at which anything has really measured the file.
Does it change anything besides the level?
The gain is one multiply per sample and nothing else, and the length never changes. The output format can, though: WAV and FLAC keep the channel count and sample rate exactly, while MP3 carries at most two channels and 48 kHz, so a surround source is summed to stereo and a 96 kHz source is written at 48 kHz. Both are reported on the page when they happen. What does not survive on any arm is tags and cover art: decoding reduces the file to raw samples, where metadata stops existing.
Does this download the FFmpeg engine?
Only if it has to. MP3 and WAV output run entirely on the Web Audio API and a worker encoder. FLAC output has no encoder in that worker, so that arm renders a lossless WAV and hands it to FFmpeg; so does a container your browser refuses to decode, which means WMA everywhere and AAC or Ogg in some browsers. If neither happens, the engine is never fetched.
Is my file uploaded to a server?
No. Transmute processes everything locally in your browser using JavaScript and WebAssembly. Your files never leave your device — there is no server, no upload, no cloud processing.