Audio Tools New Tool

Free Online Audio Sample Rate Converter

Change an audio sample rate free without uploading anything. Downsampling is filtered first so nothing aliases, every option shows the highest frequency that will survive and the size change, and the page is honest that converting upwards gains nothing.

Free Forever Nothing Uploaded Filtered, Not Decimated Shows What You Lose

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Drop audio files here to change their sample rate Down to make them smaller, or up to match what something else expects. Select Audio Files MP3, WAV, M4A, OGG, FLAC and more • best under 25 minutes • nothing is uploaded

The shape of your sound will appear here.

0:00.000 / 0:00

The new sample rate

Convert to

This file is-

Highest sound kept-

Size change-

Output

Format

Channels

WAV detail Only used when the output is WAV. The compressed formats have their own.

Encoder

File name

.wav

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How to Change an Audio Sample Rate

Pick a rate, see what it costs, run it.

Drop the files in Each one's current sample rate is read as it opens, which is what the comparison is made against.

Choose the new rate Every option says what it is for. The panel shows the highest sound that will survive and how the size will change.

Check the direction Going down loses the top end and makes the file smaller. Going up costs size and gains nothing, and the page says so rather than letting you assume otherwise.

Run it and take the files The conversion is filtered properly, so nothing aliases. WAV out needs no encoder at all.

What Sample Rate Actually Controls

One number, and what it decides.

The highest sound a file can hold is half its sample rate. At 44,100 that is 22,050 Hz, above anything most adults can hear. At 16,000 it is 8,000 Hz, which is plenty for speech and removes the sparkle from music. That one rule explains every choice on this page.

Converting upwards adds nothing. A 22,050 Hz file made into a 48,000 Hz file is the same recording in a file twice the size. It is worth doing only when something else insists on a particular rate, such as a video editor that wants everything at 48,000.

Going down has to be filtered, and it is. Simply throwing samples away makes high sounds reappear as low ones, which is called aliasing and sounds metallic and wrong. Your browser's audio engine filters properly before it resamples, which is why this is done that way rather than by hand.

For WAV, the size follows the rate exactly. Half the rate is half the file. For MP3 and the other compressed formats it makes much less difference, because there the bitrate is what decides the size.

The work happens on your own machine. Your file is opened by this page, not sent anywhere. That is why there is no upload bar, no queue and no waiting for a server. It is also why a long recording leans on your own memory: nothing is being done for you somewhere else.

Most of what this page does needs no download at all. Your browser already has a full audio engine in it, and that is what does the work. A thirty megabyte encoder is only fetched if you ask for MP3, M4A, OGG or FLAC at the end. Ask for WAV and nothing is downloaded, ever.

Length matters more than file size here. A browser opens sound at full detail, which works out at about twenty one megabytes of memory for every minute of stereo. A three minute song is nothing. A two hour recording is a different matter, and this page tells you where you stand the moment you pick a file.

MP3, M4A, OGG and Opus throw information away, every time. Opening one and saving it again is a second pass of that, so the result is always slightly worse than the original even if you changed nothing. Where it matters, take WAV or FLAC out of this page instead.

Key Features & Capabilities

What this tool does, and what it deliberately does not.

Nothing uploaded Converted in the page on your own machine, one file or a folder.

Filtered, not decimated Downsampling is filtered first, so high sounds do not fold back as low ones. That is what makes the difference between clean and metallic.

Says what you lose The highest frequency that survives and the size change, shown for each rate before you commit.

Honest about going up It will convert upwards if you need to match something, and it tells you plainly that nothing is gained by it.

A whole folder at once Files already at the target rate are passed through unchanged rather than needlessly reprocessed.

WAV downloads nothing The resampling itself uses what your browser already has.

About the Bulk Audio Sample Rate Converter

Sample rate is the number of times a second the shape of a sound was measured when it was recorded, and it decides exactly one thing: the highest frequency the file can contain. That limit is half the sample rate, which is a mathematical result rather than a convention, and everything else about this page follows from it. A CD at 44,100 samples a second can hold sound up to 22,050 Hz, above the top of adult hearing. A speech recording at 16,000 can hold up to 8,000 Hz, which covers every consonant that matters and none of the sparkle that music needs.

Going down is therefore a real choice with a real cost, and the cost is entirely at the top. Speech at 22,050 Hz is indistinguishable from speech at 44,100 for most listeners, and the file is half the size. Music at 22,050 loses its air and its cymbals sound dull. The page shows the surviving frequency limit for every option so that the trade is visible rather than implied.

Going down also has to be done correctly, and this is where tools differ. If you simply keep every second sample and discard the rest, frequencies above the new limit do not disappear; they fold back down into the audible range as different, wrong frequencies. That is aliasing, and it sounds metallic and unpleasant. The correct method filters everything above the new limit out first and then resamples. That filtering is built into your browser's audio engine, and using it is the reason this page does not do the arithmetic by hand.

Going up is the case people misunderstand most often. Converting a 22,050 Hz file to 48,000 does not recover anything, because there is nothing to recover: the original recording never contained sound above 11,025 Hz, and interpolating between samples cannot invent it. What you get is a file twice the size with identical content. There is still one good reason to do it, which is that other software may insist on a particular rate, and video editors in particular want everything at 48,000. So the option is here, and the page tells you plainly what it does rather than letting the larger number imply an improvement.

For a batch, each file is measured on its own, and any file already at the target rate is passed through untouched rather than needlessly rebuilt. If you take the results out as WAV, nothing is downloaded at any point.

Frequently Asked Questions

Rates, quality, file size and matching other software.

For music, 44,100 Hz, which is what CDs and most streaming use. For anything going into video, 48,000 Hz, because that is what video expects and a mismatch causes drift in some editors. For speech that needs to be small, 16,000 or 22,050 Hz is perfectly clear.

No. The detail above half the original rate was never recorded, and no conversion can invent it. You get a larger file containing exactly the same sound. The only good reason to convert upwards is that something else requires a particular rate.

Everything above half the new rate. At 16,000 Hz that means nothing above 8,000 Hz survives, which speech does not need but music does. At 22,050 it is 11,025 Hz, which is noticeable on cymbals and not much else.

For WAV, because the size is the rate multiplied by the channels, the bit depth and the length, so halving the rate halves the file exactly. For MP3 and other compressed formats the bitrate decides the size, so the change is much smaller.

Not when it is done properly, which means filtering out everything above half the new rate before resampling. Skipping that step makes high frequencies fold back as low ones, a metallic effect called aliasing. Your browser's audio engine does the filtering, which is why this page uses it.

Yes. Convert everything to whatever the project uses, usually 48,000 Hz for video, and the complaint goes away. Dropping files at different rates into one timeline is the usual cause of audio slowly drifting out of sync.