Audio Converter
Change sample rate and channel count, normalize, and re-export
100% In-Browser · Private & Instant · Free Forever
Your file is decoded, processed, and re-encoded in this browser tab. Nothing is uploaded to an OfflineTTS processing API.
Interactive Workspace
Audio Converter
Convert a file to the sample rate and channel layout another tool or platform expects. Everything is decoded to PCM first, so the input format is irrelevant — MP3, WAV, FLAC, OGG, M4A, and the audio track of an MP4 or WebM file all work.
Sample rate conversion is done by linear interpolation. That is fast and adequate for speech, for preparing audio for a voice model, and for reducing file size. It is not a mastering-grade resampler with an anti-aliasing filter, so downsampling material with significant high-frequency content can introduce aliasing. For music destined for release, downsample in a DAW instead.
Channel conversion folds stereo to mono by averaging, or duplicates a mono source across two channels. Averaging can cancel content that is out of phase between the channels — the classic symptom is a stereo synth pad that partly disappears in mono. Check the mono result if your material will be played on a mono speaker.
Peak normalization brings the file to a chosen ceiling before export, which is useful when several converted files need to match.
Formats you get: WAV at 16-bit PCM, and MP3 at 128, 192, or 320 kbps. FLAC, OGG, and M4A export are deliberately not offered: encoding them in the browser would require a multi-megabyte WASM codec, and a page that promises a format it cannot reliably produce wastes your time.
Files are decoded, measured, and re-encoded inside this browser tab. Nothing is uploaded and no result is retained after you close the page.
How It Works
Load a file
Drop in the audio or video file you want to convert.
Choose the target
Set the sample rate, channel layout, and normalization.
Pick an export format
WAV for lossless, MP3 with a bitrate for delivery.
Convert
The file is processed and downloaded from this tab.
Features
Sample Rate
8 kHz to 48 kHz, or keep the original rate.
Mono / Stereo
Fold stereo to mono or widen a mono source.
Peak Normalize
Bring the export to -3, -1, or -0.3 dBFS.
Wide Input Support
MP3, WAV, FLAC, OGG, M4A, AAC, MP4 and WebM inputs.
Choosing a Sample Rate and Channel Layout
Sample rate determines the highest frequency that can be represented: roughly half the sample rate, a relationship known as the Nyquist limit. A 16 kHz file can carry content up to 8 kHz, which covers the range that matters for speech intelligibility but cuts the brightness of music and cymbals. A 44.1 kHz file reaches about 22 kHz, which is beyond most adult hearing.
The practical question is what the destination needs. Voice models and speech recognition typically want 16 kHz mono. Video editors want 48 kHz at the project's channel count. Streaming platforms accept 44.1 or 48 kHz. Converting to a higher rate than the source ever contained adds file size without adding information, so it is almost never worth doing.
Channel count is a separate decision from sample rate, and the two interact. Converting a 48 kHz stereo file to 16 kHz mono is a common preparation step for speech tooling; converting a 16 kHz mono file to 48 kHz stereo produces a large file with no more content than the original.
What Conversion Does Not Fix
Conversion changes the container the audio lives in, not the audio itself. It will not repair clipping, remove noise, restore lost high frequencies, or improve a badly recorded source. If the goal is a better-sounding file rather than a compatible one, the other tools on this site are the right starting point — noise removal, equalization, and level correction change the signal rather than reformatting it.
For lossy sources there is a second consideration: transcoding an MP3 to a different lossy format adds a generation of compression artefacts. An MP3 decoded and re-encoded at a lower bitrate will sound worse than the original, and the loss is cumulative. Where you have the choice, keep a WAV master and encode to lossy formats only at the final delivery step.
Files are decoded, measured, and re-encoded inside this browser tab. Nothing is uploaded and no result is retained after you close the page.
Related Tools
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Audio Joiner
Merge files in order, with optional normalization, crossfades, and gaps
BPM & Key Finder
Tempo, key with Camelot code, tuning reference, and loudness measurements
Audio Converter — FAQ
Why can't I export FLAC, OGG, or M4A?
Producing those formats in the browser needs a large WASM encoder. Shipping one would add several megabytes to every page load and slow down the common case — converting a file to WAV or MP3. Since the platform does not do it well, the option is not offered at all rather than presented and then failing.
Is sample rate conversion lossy?
Changing the sample rate is mathematically a resampling operation, and this implementation uses linear interpolation. Upsampling adds no information, and downsampling without an anti-aliasing filter can fold high frequencies back into the audible range. For critical music work, do the conversion in a DAW with a proper resampler.
What sample rate should I choose?
16000 Hz is standard for speech recognition and voice models. 22050 Hz suits speech playback and small files. 44100 Hz matches CD audio and most consumer playback. 48000 Hz is the video and broadcast standard. If you do not have a specific requirement, keep the original rate.
Will converting to mono break my audio?
It will change it. Mono conversion averages the left and right channels, so any content that is identical in both survives while content that is opposite in phase cancels. Stereo widening effects and some synth pads are vulnerable. Check the mono result on headphones before committing.
Can I convert a video file to audio?
Yes. MP4 and WebM files are decoded by the browser and their audio track can be converted and exported. Video is discarded, and no video output is produced.
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