Sample rate converter: 48 kHz to 44.1 kHz and back

A sample rate converter resamples audio from one rate to another, most often 48 kHz to 44.1 kHz, without changing its pitch or length. This tool decodes a WAV, MP3, M4A, FLAC or OGG file, resamples it to 22.05, 32, 44.1, 48, 88.2 or 96 kHz in your browser and downloads a 16-bit WAV with "-resampled" added to the file name. Nothing is uploaded.

Drop the file, pick the target rate, click Process and download. The status line shows the loaded file's channel count, rate and length, then the size of the finished WAV in KB. The default target is 44.1 kHz because most people use this page as a 48 kHz to 44.1 kHz converter.

48 kHz to 44.1 kHz: why it is the most common conversion

Two worlds run at two rates. Video, film, broadcast, cameras, phones recording video and most game engines work at 48 kHz. Music distribution grew out of the CD, which is fixed at 16-bit, 44.1 kHz, and most distributors and streaming services still take 44.1 kHz masters as the baseline (many also accept 48 kHz). Whenever audio made alongside video becomes a music release, or a music session feeds a video edit, the file crosses that line.

The ratio is awkward: 44,100 ÷ 48,000 = 147 ÷ 160, so every 160 input samples become 147 output samples and almost none of them line up, which is why a real resampler interpolates between samples instead of dropping or repeating them. The reverse direction, 44.1 kHz to 48 kHz for a video edit or a game project, uses the same method with the ratio flipped: 1 second stays 1 second, now described by 48,000 samples instead of 44,100.

What a converter does not do is rewrite the header and leave the samples alone. A 44.1 kHz file played as if it were 48 kHz runs 8.84% fast and 1.47 semitones (147 cents) sharp, so a 3:30 song finishes in 3:13; a 48 kHz file played at 44.1 kHz runs 8.13% slow and 1.47 semitones flat. That tape-style shift is what the tape / vari-speed calculator works out, and it is the sound of a sampler playing a file at the wrong rate.

Sample rates and what they are for

The sample rate sets the highest frequency a file can hold, at half the rate (the Nyquist limit), and how many samples make up a millisecond. Hearing tops out around 20 kHz, which is why 44.1 and 48 kHz are the delivery rates and the higher ones are production rates.

Samples per ms turns a delay or buffer size into time: 480 samples is 10 ms at 48 kHz but 10.9 ms at 44.1 kHz. The samples to ms calculator does that for any rate.

Does converting sample rate lose quality?

Honest answer: a little, and far less than people fear, provided you do it once. Going down from 48 to 44.1 kHz, the converter removes everything above the new Nyquist limit of 22.05 kHz with a low-pass filter. That band sits above hearing, so the audible loss is not the missing top octave but any ripple or aliasing the filter adds, which a good resampler keeps far below the noise floor of a 16-bit file. This tool renders through the browser's own resampling engine (an OfflineAudioContext at the target rate), not a mastering-grade algorithm: fine for samples, demos and stems, but for a final master prefer the resampler in your DAW or mastering chain.

Going up, from 44.1 to 48 or 96 kHz, adds no information: the file gets bigger and the bandwidth limit of the original stays where it was. Do it when a project or device demands the higher rate, not for quality.

Bit depth is the other half of the story. This converter writes 16-bit PCM without dither, so a 24-bit source is truncated to 16 bits on the way out. For a sample or a reference that is irrelevant; for a quiet, dynamic master it is the difference between a smooth fade and a faintly grainy one, so resample that in the DAW with dither. The WAV bit depth converter handles the depth step on its own. Resampling does not change loudness, so a mix at -14 LUFS before is -14 LUFS after; confirm with the LUFS meter if a platform spec is on the line, and see our guide to mastering for streaming for the rest of the delivery checklist.

Avoid chains of conversions (44.1 to 48 for the edit, back to 44.1 for the single, up again for the trailer); each pass adds its own filter. Keep the original at its native rate and convert from it every time.

Worked examples

A 3:30 stereo stem recorded at 48 kHz, 24-bit on a video shoot is 60.5 MB. Converted here to 44.1 kHz it comes out as a 16-bit WAV of 37.0 MB, still 3:30 long, still the same pitch, with 2,646,000 samples per channel per minute instead of 2,880,000. A 44.1 kHz drum one-shot headed for a 48 kHz game project has each second of 44,100 samples turned into 48,000; a 500 ms sample is still 500 ms. An MP3 or M4A dropped here is decoded, resampled and written as WAV in one step, the same output the audio to WAV converter gives at the original rate; the audio to MP3 converter goes the other way for a small preview.

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