Audio Tools

Sample Rate Converter

Resample audio to any standard rate from 8 kHz to 96 kHz using the browser's offline rendering engine.

Free · no account Runs in your browser Audio

Convert with this tool

Drag & Drop File Here

or choose a file from your device — it is processed in this tab and never uploaded

Accepted: WAV, MP3, FLAC, M4A/AAC, OGG, Opus

    Ready.

    Files stay on your device. There is no upload, no queue and no watermark, so a conversion works even with the network switched off after the page has loaded.

    Supported formats & processing

    Accepted inputWAV, MP3, FLAC, M4A/AAC, OGG, Opus
    Produced outputWAV or MP3 at 8, 16, 22.05, 32, 44.1, 48 or 96 kHz
    ProcessingWeb Audio API + canvas + native JavaScript encoders
    Upload requiredNo — the file is read locally and never transmitted

    How to use this tool

    1. Add the audio fileThe source rate is reported from the decoded buffer.
    2. Choose the target rate48 kHz for video work, 44.1 kHz for music delivery, 16 kHz for speech models, 8 kHz for telephony.
    3. Pick an output formatWAV for further processing, MP3 for distribution.
    4. ConvertResampling is fast: the offline render runs much quicker than real time.
    5. Check the reported rateThe result card confirms the new rate, the channel count and the new duration.

    About this tool

    Sample rate is how many times per second the amplitude of the waveform is measured. 44.1 kHz covers frequencies up to 22.05 kHz, just past the limit of human hearing; 48 kHz is the video and broadcast standard; 96 kHz gives editing headroom; 8 and 16 kHz are for telephony and speech recognition.

    Resampling is done by the browser's OfflineAudioContext, which interpolates the waveform properly rather than dropping or repeating samples. When you downsample, the audio content above the new Nyquist frequency has to go — that is physics, not a limitation of the tool, and it is why a 48 kHz recording converted to 8 kHz sounds like a telephone.

    Features

    • Seven standard rates covering telephony through studio work
    • Proper band-limited resampling via OfflineAudioContext, not naive decimation
    • Nyquist consequences explained before you commit
    • Channel count preserved, or changed deliberately with mono/stereo conversion
    • Duration and level reported after resampling so drift is visible
    • Combine with the audio compressor for voice pipelines in one pass

    Frequently asked questions

    Should I upsample 44.1 kHz audio to 96 kHz?

    Only if your editing software requires a uniform rate across a session. Upsampling does not add detail; it just makes the file larger and gives plug-ins more headroom to work in.

    Why does audio above 11 kHz disappear at 22.05 kHz?

    The Nyquist theorem sets the limit at half the sample rate. To store 15 kHz content faithfully you need at least a 30 kHz sample rate, which is why 22.05 kHz sounds dull for music.

    What rate do I need for video?

    48 kHz. Video editors and delivery specifications expect it, and mixing 44.1 kHz audio into a 48 kHz timeline causes the editor to resample anyway.

    Will the duration change?

    It stays the same. Sample rate changes the number of samples, not the length of time they represent — the duration is reported after conversion so you can confirm it.