Audio Sample Rate Converter
Change the sample rate of audio without losing quality. Convert 48 kHz video audio to 44.1 kHz for music platforms, 44.1 kHz to 48 kHz for video projects, or reduce speech recordings to 16 kHz for transcription services. A high-quality resampler with a steep anti-aliasing filter is used, and you can save as WAV or FLAC with 16 or 24 bits, or as MP3 or M4A.
- Files stay on your device
- No sign-up
- Free to use
Sample rate
High-quality resampling with a steep anti-aliasing filter. Raising the sample rate does not add detail; lowering it removes frequencies above half the new rate.
Processing happens on your device; your audio is not uploaded.
How to use Audio Sample Rate Converter
- Drop an audio file onto the page.
- Choose the new sample rate.
- Choose the bit depth and output format.
- Convert and download.
Audio Sample Rate Converter features
All common rates
8 to 96 kHz.
High-quality resampling
Long filter, steep cutoff.
16 or 24 bit
For WAV and FLAC.
Any output
Original, MP3, M4A, WAV, FLAC.
Tags kept
Title and artist preserved.
Private
Processed in your browser; nothing is uploaded.
When to use Audio Sample Rate Converter
- Matching audio to a 48 kHz video project.
- Preparing 44.1 kHz masters for streaming.
- Downsampling speech for recognition services.
- Meeting a broadcaster’s delivery spec.
Audio Sample Rate Converter FAQ
Does a higher sample rate improve quality?
No. Upsampling cannot add information that was not recorded; it only increases file size.
What is lost when lowering the rate?
Frequencies above half the new rate. 44.1 kHz keeps everything up to 22 kHz, beyond human hearing; 16 kHz keeps up to 8 kHz, enough for speech.
Which rate does video use?
Video and broadcast use 48 kHz; music CDs and most streaming use 44.1 kHz.
Why can bit depth be chosen only for WAV and FLAC?
MP3 and AAC do not store a fixed bit depth.
Why does the first run take longer?
Re-encoding uses FFmpeg compiled to WebAssembly (about 32 MB), which is downloaded once and then cached.
Which formats can I use?
MP3, WAV, M4A/AAC, OGG, Opus, FLAC, WMA and AIFF, and the audio of MP4, MOV, WebM and MKV videos. Save in the original format or as MP3, M4A, WAV or FLAC.
Sample rates explained
Digital audio is a sequence of measurements – samples – taken many thousands of times per second. The sample rate decides the highest frequency that can be represented: half the rate, known as the Nyquist frequency. CD audio at 44,100 samples per second covers up to 22,050 Hz, slightly above the limit of human hearing.
Different fields settled on different rates. Music uses 44.1 kHz, video and broadcast 48 kHz, telephony 8 kHz, speech recognition often 16 kHz, and studios record at 88.2 or 96 kHz for headroom during processing. Moving audio between these worlds requires resampling.
Resampling calculates new sample values between the existing ones. Done carelessly, it creates aliasing – audible artefacts from frequencies that do not fit the new rate. The converter uses FFmpeg’s resampler with a long filter and a steep cutoff just below the new Nyquist frequency, removing those frequencies cleanly before reducing the rate.
For further editing, save as WAV or FLAC with 24 bits; for distribution, MP3 or M4A at the new rate.
Your audio is processed on your device.