S = t × f × C × b / 8
Start your audio project
Drop an audio file here or choose an option below.
Recording…
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How does this tool work?
Resample audio to 8, 16, 22.05, 32, 44.1, 48, 88.2 or 96 kHz. The operation adapts sample rate without claiming to add detail that is absent from the source.
FFmpeg/WASM decodes the track and generates new samples at the selected rate before encoding PCM WAV. Raising the sample rate of a source that was already bandwidth-limited cannot invent extra high-frequency content; lowering it can remove information above the new usable bandwidth. The initial rate is 44,100 Hz. Only eight discrete rates are accepted; an intermediate value passed by another engine consumer is rejected. Resampling recalculates the signal and writes the new WAV rate rather than merely changing a playback label. Speed and pitch therefore remain unchanged to frame precision. Output is 16-bit PCM even when the source uses another resolution. Raising the rate increases the number of stored samples without recovering information lost in earlier compression. Check your destination’s requirements before choosing a higher rate simply because its number is larger.
How to use this tool
Import the audio
Choose a compatible file.
Choose the sample rate
Select one of the available rates from 8 to 96 kHz.
Resample
Download the generated WAV and check compatibility with the target software.
Change audio sample rate — Calculation reference
- S
- PCM payload in bytes, excluding headers
- t
- Duration in seconds
- f
- Sample rate in Hz
- C
- Channel count, dimensionless
- b
- Bits per sample
Use cases
Standardize a project
Bring several files to the same rate before editing or merging.
Voice and telephony
Prepare audio at a lower rate when a target system requires it.
Audio production
Adapt a source to 44.1 or 48 kHz for a particular workflow.
Examples
One second at two rates
Frame count changes with rate, while target duration remains one second. Filter edges can introduce rounding.
Input
1 s at 44,100 Hz, resampled to 48,000 HzResult
Approximately 48,000 frames for the same secondOne minute of stereo PCM
96,000 ÷ 48,000 = 2, excluding headers. Sample rate in Hz is different from a compressed bitrate in kbit/s.
Input
60 s, 48,000 Hz, 16-bit PCM, 2 channelsResult
11,520,000 PCM bytes; at 96 kHz: 23,040,000Tips and best practices
- Increasing sample rate does not restore detail that is already absent from the source.
- Use the same sample rate across files in one project to avoid repeated conversions.
- Sample rate is different from bitrate: it describes samples per second, not directly the size of a compressed file.
Sources and references
- FFmpeg project, FFmpeg Filters Documentation (opens in a new tab)Page in English
- Microsoft, WAVEFORMATEX structure (mmeapi.h) (opens in a new tab)Page in English
Frequently asked questions
Do 44.1 kHz and 48 kHz change playback speed?
Not when audio is properly resampled: duration and pitch should remain unchanged.
Does 96 kHz automatically improve a 44.1 kHz source?
No. Upsampling cannot recreate information that never existed in the source.
What format is produced?
The sample-rate converter currently outputs PCM WAV.
Can I enter a custom rate such as 50 kHz?
No. Accepted rates are 8,000, 16,000, 22,050, 32,000, 44,100, 48,000, 88,200 and 96,000 Hz. Other values are rejected even if technically plausible.
Why does a 96 kHz WAV use more space than 48 kHz?
At the same duration, channel count and resolution, doubling the rate doubles PCM data. It cannot add detail missing from a lower-rate recording.
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