Audio Sample Count Calculator
Convert an audio clip’s duration into total samples and uncompressed PCM file size from the sample rate, bit depth and channel count.
How it works
Samples/channel = duration · sample rate · PCM bytes = duration · rate · channels · (bit depth ÷ 8)
Digital audio stores one number (a sample) per channel at each tick of the sample clock, so the sample count is just the duration times the sample rate. Multiplying by the bytes per sample (bit depth ÷ 8) and the channel count gives the raw PCM size — the same math a WAV file uses. Compressed formats like AAC or FLAC end up much smaller, but the uncompressed size sets your disk and buffer budgets.
Worked example
A 3-minute (180 s) stereo CD track at 44,100 Hz, 16-bit: 180 × 44,100 = 7,938,000 samples per channel. Times 2 channels and 2 bytes each is 31,752,000 bytes ≈ 30.3 MiB, at a data rate of 44,100 × 16 × 2 ÷ 1000 = 1,411 kbps.
Frequently asked questions
Why 44,100 samples per second?
The Nyquist theorem says you must sample at more than twice the highest frequency you want to capture. Human hearing tops out near 20 kHz, so 44.1 kHz (chosen for compatibility with early digital video equipment) captures the full audible band with margin for anti-aliasing filters.
How much smaller is a compressed file?
Lossy codecs like AAC or MP3 at 256 kbps are roughly a fifth to a sixth the size of 16-bit CD PCM. Lossless FLAC typically saves 40–60% with no quality change. This calculator shows the uncompressed size, which is what editing software and buffers actually handle.
Does 32-bit float double the size versus 16-bit?
It doubles the size versus 16-bit and stores each sample as a floating-point number, which effectively removes clipping worries during editing because values above 0 dBFS are retained. The trade-off is exactly the larger file this calculator reports.
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This calculator is for educational purposes. Double-check important results before acting on them.