Change an audio file's sample rate, from 8 kHz to 96 kHz

Pick a target rate and the tool recalculates every sample with a band-limited filter, so a 48 kHz camera recording can become 44.1 kHz, or a voice file can drop to 16 kHz for speech software.

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Processed on your device — nothing is uploaded.

How to use the sample rate converter

  1. Drop in the audio file, or the video whose sound you want to resample.
  2. Choose the Target sample rate: 8, 11.025, 16 (speech recognition), 22.05, 32, 44.1 (CD), 48 (video), 88.2 or 96 kHz.
  3. Choose the WAV type: 16-bit PCM, 24-bit PCM or 32-bit float.
  4. Press Resample and download the WAV. The result lists New rate, Original rate, Length and File size.

If the file is already at the rate you picked, you get “The file is already 44,100 Hz.” (or whichever rate) instead of a new copy.

What it does and when to use it

Software and hardware are often strict about sample rates. Video projects usually run at 48 kHz, so 44.1 kHz music may need converting. Speech-to-text engines and voice-bot platforms often ask for 16 kHz mono. Phone systems and some embedded players need 8 kHz. A CD-burning program wants 44.1 kHz. This tool gives you the exact rate those systems expect, in a WAV file.

It always writes WAV. For a compressed file at a new rate, the audio converter has a smaller choice of rates (44.1, 48, 22.05, 16 kHz) plus M4A and Opus output. To make the file mono as well, run mono to stereo in mono mode. To check a file’s current rate first, open it in the audio metadata viewer.

How it works

The file is decoded in a Web Worker on your device. The tool then uses its own polyphase windowed-sinc resampler: each new sample is a weighted sum of nearby original samples, with weights from a sinc function shaped by a Blackman window and 512 fractional positions. When downsampling, the filter cut-off is placed at 94% of the new Nyquist frequency (half the new rate), which removes content that would otherwise fold back as aliasing distortion. When upsampling, the cut-off sits at 94% of the original Nyquist. The result is written as WAV.

Worked examples

From the tool’s output on 10-second test files:

  • 48 kHz stereo → 44.1 kHz: New rate 44,100 Hz, Original rate 48,000 Hz, Length 0:10.00, File size 1.7 MB (16-bit). A note says frequencies above 22.05 kHz are filtered out.
  • 44.1 kHz mono → 16 kHz: File size 312.5 KB instead of about 861 KB, with a note that content above 8 kHz is removed.
  • 44.1 kHz mono → 48 kHz: File size 937.5 KB, with the note “Upsampling does not add detail; it only changes the rate.”

Limits and tips

  • Resample once, from the best original you have. Each extra conversion filters the top end slightly.
  • The filter works on each channel separately; channel count does not change.
  • Inputs other than WAV are decoded with your browser’s WebCodecs. Rare formats such as AIFF, AMR, WMA or CAF are not covered by this tool’s options; convert them with audio to WAV and its ffmpeg fallback first.

Frequently asked questions

What sample rate should I use?
44.1 kHz is the CD and most music-streaming rate. 48 kHz is standard for video. 16 kHz is common for speech recognition and voice assistants, and 8 kHz for old telephone systems. If you have no requirement, keep the original.
Does upsampling to 96 kHz improve quality?
No. It only adds samples between the existing ones; the tool even says so. It can help when a program refuses a lower rate, but the sound holds no new detail and the file gets bigger.
What is lost when I go down to 16 kHz?
Everything above roughly 8 kHz, half the new rate. Speech stays clear but loses some crispness on s and t sounds; music sounds dull. That is unavoidable when downsampling.
Why does the length stay the same?
Resampling changes how many samples describe each second, not how many seconds there are. Pitch and speed stay the same. If you want a speed change, use the speed changer.

Written by the ToolsRift team · Last updated

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