Audio Sample Rate Converter

Resample audio files between 44.1 kHz (CD), 48 kHz (Video & Broadcast), 88.2 kHz, and 96 kHz studio standards with anti-aliasing interpolation and lossless WAV export.

🎵 Audio 🔒 100% Client-Side ⚡ OfflineAudioContext Sinc Engine ✓ Zero Drift Audio
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Click to Upload Audio (WAV, MP3, OGG, FLAC) or Drag & Drop
Convert between 44.1 kHz music standard and 48 kHz YouTube / Premiere video standard
Quick Overview & Direct Answer

Audio Sample Rate Converter — Resample 44.1kHz, 48kHz & 96kHz Online: Free Audio Sample Rate Converter tool: resample audio sample rates online between 44.1 kHz, 48 kHz, 88.2 kHz, and 96 kHz using browser sinc interpolation..

Primary Use Audio format conversion, waveform inspection, sound trimming
Input Formats Audio files (MP3, WAV, AAC, OGG)
Output Format Converted audio files and playback waveform data
Processing Mode In-browser Web Audio API & localized processing
Access Tier Free, unlimited, no sign-up needed

How to Convert Audio Sample Rates Online

1

Upload Audio Track

Import your MP3, WAV, FLAC, or OGG recording into the sample rate converter.

2

Select Target Sample Rate

Choose 48 kHz for YouTube and video editing, or 44.1 kHz for CD audio distribution.

3

Process with OfflineAudioContext

The browser engine renders high-order sinc interpolation with anti-aliasing filtering.

4

Download Resampled WAV

Download your clean, drift-free uncompressed WAV file ready for import into Adobe Premiere or Final Cut Pro.

Why Sample Rate Mismatch Causes Audio-Video Drift in Video Editors

One of the most persistent headaches in video production occurs when an editor imports music tracks recorded at $44,100 ext{ Hz}$ into a timeline configured for $48,000 ext{ Hz}$ video standard. Over a 30-minute documentary or podcast interview, dialogue slowly slips out of synchronization by several frames, causing lip-sync mismatch.

This happens because video frames divide evenly into 48,000 samples per second ($48,000 / 24 = 2,000 ext{ samples/frame}$; $48,000 / 30 = 1,600 ext{ samples/frame}$), whereas $44,100$ produces fractional samples ($44,100 / 24 = 1,837.5 ext{ samples/frame}$). Over time, rounding errors accumulate into noticeable drift.

The Mathematics of Sinc Interpolation and Anti-Aliasing

Sample rate conversion is not simply dropping or duplicating samples (which causes severe harmonic distortion and ringing). High-quality resampling reconstructs the continuous analog curve using the Whittaker-Shannon interpolation formula:

$$x(t) = sum_{n=-infty}^{infty} x[n] cdot ext{sinc}left( rac{t - nT}{T} ight) quad ext{where } ext{sinc}(u) = rac{sin(pi u)}{pi u}$$

When downsampling (e.g. from 96 kHz to 44.1 kHz), an aggressive steep low-pass brickwall filter is applied at $22,050 ext{ Hz}$ to prevent ultrasonic energy from folding over into audible frequencies as harsh aliasing noise.

Key Benefits of the Audio Sample Rate Converter

Fix Video Lip-Sync Drift

Convert 44.1 kHz audio to 48 kHz standard to prevent audio-video drift in Premiere Pro and DaVinci Resolve.

High-Order Sinc Interpolation

Uses Web Audio's native multi-stage polyphase sinc filters to preserve audio transparency.

Uncompressed WAV Download

Download studio-ready 16-bit uncompressed WAV files with clean headers.

DAW Compatibility Check

Eliminates 'Sample Rate Mismatch' alerts when importing audio into Ableton Live, Logic, or Pro Tools.

Anti-Aliasing Filter Protection

Automatically suppresses ultrasonic harmonics above target Nyquist limit to prevent mirror distortion.

100% Client-Side Privacy

All resampling algorithms execute directly in your browser without transmitting your files.

Frequently Asked Questions

Why is 48 kHz preferred over 44.1 kHz for video?
48,000 divides cleanly by all standard video frame rates: 24 fps (2000 samples/frame), 25 fps (1920 samples/frame), 30 fps (1600 samples/frame), and 60 fps (800 samples/frame). This prevents cumulative audio drift over long runtimes.
Does upsampling a 44.1 kHz track to 96 kHz improve sound quality?
Upsampling cannot restore ultrasonic frequencies that were never recorded in the original file. However, upsampling is widely done before digital audio signal processing (such as saturation, pitch-shifting, and EQ) to push digital processing anti-aliasing artifacts far above the audible spectrum.
What is aliasing distortion?
When sound frequencies higher than half the sampling rate (the Nyquist frequency) are downsampled without proper filtering, they 'fold over' into the audible range as harsh, discordant harmonic artifacts. Our resampler uses steep anti-aliasing filters to prevent this.
How fast is the conversion process?
Because it utilizes your device's multi-threaded Web Audio hardware pipeline, a 5-minute audio track typically resamples in less than two seconds.
Can I convert stereo and mono audio?
Yes. Both single-channel mono and discrete dual-channel stereo tracks are fully supported with exact channel balance preservation.